Called is disconnected after 1-2 seconds

[scorpio1441]

I wonder why my call is getting disconnected almost immediately, even though audiofork keeps receiving frames....

Connected to Asterisk 18.15.1 currently running on ithelpdesk1 (pid = 26106)
-- Executing [@mycontext:1] NoOp("SIP/voipms-00000000", "") in new stack
-- Executing [@mycontext:2] Verbose("SIP/voipms-00000000", "starting audio fork") in new stack
starting audio fork
-- Executing [@mycontext:3] AudioFork("SIP/voipms-00000000", "ws://localhost:8080/") in new stack
== <SIP/voipms-00000000> [AudioFork] (both) Setting Direction
== <SIP/voipms-00000000> [AudioFork] Setting reconnection attempts to 5
== <SIP/voipms-00000000> [AudioFork] Setting reconnection timeout to 5
== <SIP/voipms-00000000> [AudioFork] (both) Setting wsserver: ws://localhost:8080/
== <SIP/voipms-00000000> [AudioFork] (both) Completed Setup
== <SIP/voipms-00000000> [AudioFork] (both) Added AudioHook Spy
-- Executing [@mycontext:4] Verbose("SIP/voipms-00000000", "audio fork was started continuing call..") in new stack
audio fork was started continuing call..
-- Executing [@mycontext:5] Playback("SIP/voipms-00000000", "hello-world") in new stack
== <SIP/voipms-00000000> [AudioFork] (both) Keeping Call-ID Association
== <SIP/voipms-00000000> [AudioFork] (both) Connecting websocket server at: ws://localhost:8080/
== <SIP/voipms-00000000> [AudioFork] (both) Creating WS without TLS
== <SIP/voipms-00000000> [AudioFork] (both) Begin AudioFork Recording SIP/voipms-00000000
-- <SIP/voipms-00000000> Playing 'hello-world.gsm' (language 'en')
-- Executing [@mycontext:6] Hangup("SIP/voipms-00000000", "") in new stack
== Spawn extension (mycontext, ****, 6) exited non-zero on 'SIP/voipms-00000000'
== <SIP/voipms-00000000> [AudioFork] (both) AST_AUDIOHOOK_STATUS_RUNNING = 0
== WebSocket connection to '[::1]:8080' closed
== <SIP/voipms-00000000> [AudioFork] (both) Post Process
== <SIP/voipms-00000000> [AudioFork] (both) End AudioFork Recording to: ws://localhost:8080/

got connection
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..
received frame..

[nadirhamid]

Hello @rknetwork,

Thanks for creating this issue.

Can you please also include your dialplan so I can have a look at that ?

Thanks

[Yukari-Tryhard]

Here is my dialplan:

same => n,Progress() same => n,AudioFork(ws://192.168.1.220:8080?ucid=${UCID}, D(in,out)) same => n,Dial(PJSIP/${EXTEN}@drachtio_out,,b(addHeaders^addHeaderXRetHdr^1(${XReturnHeader}))) same => n,HangUp()

with this dialplan. my call is still continue but the audio fork stop sending new data to the websocket server.

[Yukari-Tryhard]

@nadirhamid does this repo still active?

[nadirhamid]

Here is my dialplan:

same => n,Progress() same => n,AudioFork(ws://192.168.1.220:8080?ucid=${UCID}, D(in,out)) same => n,Dial(PJSIP/${EXTEN}@drachtio_out,,b(addHeaders^addHeaderXRetHdr^1(${XReturnHeader}))) same => n,HangUp()

with this dialplan. my call is still continue but the audio fork stop sending new data to the websocket server.

@Yukari-Tryhard seems like you're setting the direction option incorrectly.

You can only specify one of the following: in, out or both.

No support for multiple options with a comma delimited list like what you have in your example.

Try to change the audiofork call to this:
same => n,AudioFork(ws://192.168.1.220:8080?ucid=${UCID}, D(both))

This should fix the issue.

If you have any other questions or need more info let me know.