I have a server that was running fine on Asterisk 16. I wanted to do some work on the source code so I switched over to the current master branch and since that recompile I have been having this issue:
This is printed to the console whenever Asterisk tries to make any type of in-dialog request to the PJSIP endpoint (WebRTC if that’s relevant) such as re-invites, when SendText is called from the dialplan or for the confbridge enhanced messaging events.
I’ve confirmed that the client works fine with when connected to another similar Asterisk server. I have also ruled out anything strange in the dialplan by stripping it right back to a single line.
The really strange thing is I have tried both Asterisk 18 and Asterisk 16, the same version I originally had everything working on and I get the same result.
I am building Asterisk with bundled pjproject and jansson
I would suggest getting a debug log[1] as they may provide better detail into it. Is the client directly connected to Asterisk? What is the configuration?
I believe this is the relevant excerpt from the debug log (I can post a full one if you think I’m missing something)
[May 16 08:05:57] DEBUG[21590][C-00000005] chan_pjsip.c: Sending MESSAGE from ‘1’ to ‘:PJSIP/dmc-live-call-00000004’: {
“type”: “ConfbridgeWelcome”,
“timestamp”: “2023-05-16T08:05:57.545+0000”,
“bridge”: {
“id”: “620788a5-fee5-4792-9f25-3fb2aeacc47d”,
“name”: “1”,
“creationtime”: “2023-05-16T08:05:54.381+0000”,
“video_mode”: “sfu”
},
“channels”: [
{
“id”: “yo26-hv02-dmc-coredev-ast01.dmcip.net-1684224353.16”,
“name”: “PJSIP/dmc-live-call-00000004”,
“state”: “Up”,
“caller”: {
“name”: “Live Call User”,
“number”: “abcd123”
},
“creationtime”: “2023-05-16T08:05:53.878+0000”,
“language”: “en”,
“admin”: false,
“muted”: false
}
]
}
[May 16 08:05:57] DEBUG[9356] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target ‘217.64.114.51’
[May 16 08:05:57] DEBUG[9356] res_pjsip/pjsip_resolver.c: Transport type for target ‘217.64.114.51’ is ‘(null)’
[May 16 08:05:57] DEBUG[9356] res_pjsip/pjsip_resolver.c: Target ‘217.64.114.51’ is an IP address, skipping resolution
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: The state change pertains to the endpoint ‘dmc-live-call(PJSIP/dmc-live-call-00000004)’
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: The inv session does NOT have an invite_tsx
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: The UAC MESSAGE transaction involved in this state change is 0x2cf1c28
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: The current transaction state is Terminated
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: The transaction state change event is TRANSPORT_ERROR
[May 16 08:05:57] DEBUG[9356] res_pjsip_session.c: The current inv state is CONFIRMED
Does this indicate that Asterisk believes the websocket connection is down?
The client connects directly to the Asterisk websocket
Yes, the websocket connection is still active. The call stays online until I hang it up and this console message is not printed until the call is completed
WebSocket connection from ‘217.64.114.51:51614’ closed
For whatever reason a transport level error is occurring when sending the request. Why that is is not clear from the logging and would probably require someone being able to reproduce the issue and digging into it. There haven’t been any other reports of such an issue at this time.