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Summary:ASTERISK-28336: After ARI continue, hangup() application does not create SoftHangupRequest event
Reporter:sungtae kim (pchero)Labels:pjsip
Date Opened:2019-03-13 17:09:34Date Closed:
Priority:MinorRegression?
Status:Open/NewComponents:PBX/General
Versions:16.2.1 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:If the channel hits the Hangup() application in the Dialplan after ARI continue action, it doesn't create SoftHangupRequest AMI event and ChannelHangupRequest ARI event.

Console logs
{noformat}
Asterisk Ready.
*CLI>  Creating Stasis app 'test'
 == WebSocket connection from '192.168.30.1:38314' for protocol '' accepted using version '13'
 == Manager 'asterisk' logged on from 192.168.30.1
0xc2bff50 - Created CDR for channel PJSIP/sipp-uac-00000000
0xc2bff50 - Transitioning CDR for PJSIP/sipp-uac-00000000 from state NONE to Single
Dial Begin message for (none), PJSIP/sipp-uac-00000000: 1552514782.00055901
0xc2bff50 - Processing Dial Begin message for channel (none), peer PJSIP/sipp-uac-00000000
0xc2bff50 - Updated Party A PJSIP/sipp-uac-00000000 snapshot
0xc2bff50 - Transitioning CDR for PJSIP/sipp-uac-00000000 from state Single to Dial
   -- Called sipp-uac/sip:localhost:5061
   -- PJSIP/sipp-uac-00000000 is ringing
Dial End message for (none), PJSIP/sipp-uac-00000000: 1552514782.00288398
   -- PJSIP/sipp-uac-00000000 is ringing
Dial End message for (none), PJSIP/sipp-uac-00000000: 1552514782.00306677
   -- PJSIP/sipp-uac-00000000 answered
      > Launching Stasis(test) on PJSIP/sipp-uac-00000000
0xc2bff50 - Set answered time to 1552514782.328749
Dial End message for (none), PJSIP/sipp-uac-00000000: 1552514782.00321811
0xc2bff50 - Processing Dial End message for channel (none), peer PJSIP/sipp-uac-00000000
0xc2bff50 - Transitioning CDR for PJSIP/sipp-uac-00000000 from state Dial to DialedPending
      > 0xc9ccf50 -- Strict RTP learning after remote address set to: 127.0.0.1:6000
   -- Executing [s@sipp-uac:2] Answer("PJSIP/sipp-uac-00000000", "") in new stack
0xc2bff50 - Transitioning CDR for PJSIP/sipp-uac-00000000 from state DialedPending to Single
   -- Executing [s@sipp-uac:3] Hangup("PJSIP/sipp-uac-00000000", "") in new stack
 == Spawn extension (sipp-uac, s, 3) exited non-zero on 'PJSIP/sipp-uac-00000000'
0xc2bff50 - Transitioning CDR for PJSIP/sipp-uac-00000000 from state Single to Finalized
0xc2bff50 - Beginning finalize/dispatch for PJSIP/sipp-uac-00000000
0xc2bff50 - Dispatching CDR for Party A PJSIP/sipp-uac-00000000, Party B <none>
{noformat}

AMI events
{noformat}
Event: DialEnd
Privilege: call,all
DestChannel: PJSIP/sipp-uac-00000000
DestChannelState: 6
DestChannelStateDesc: Up
DestCallerIDNum: <unknown>
DestCallerIDName: <unknown>
DestConnectedLineNum: <unknown>
DestConnectedLineName: <unknown>
DestLanguage: en
DestAccountCode:
DestContext: sipp-uac
DestExten: s
DestPriority: 1
DestUniqueid: test_call
DestLinkedid: test_call
DialStatus: ANSWER

Event: DeviceStateChange
Privilege: call,all
Device: PJSIP/sipp-uac
State: INUSE

Event: QueueMemberStatus
Privilege: agent,all
Queue: test
MemberName: PJSIP/sipp-uac
Interface: PJSIP/sipp-uac
StateInterface: PJSIP/sipp-uac
Membership: static
Penalty: 0
CallsTaken: 0
LastCall: 0
LastPause: 0
InCall: 0
Status: 2
Paused: 0
PausedReason:
Ringinuse: 1
Wrapuptime: 0

Event: VarSet
Privilege: dialplan,all
Channel: PJSIP/sipp-uac-00000000
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Language: en
AccountCode:
Context: sipp-uac
Exten: s
Priority: 1
Uniqueid: test_call
Linkedid: test_call
Variable: STASISSTATUS
Value:

Event: TestEvent
Privilege: reporting,all
Type: StateChange
State: CallIDChange
AppFile: channel_internal_api.c
AppFunction: ast_channel_callid_set
AppLine: 811
State: CallIDChange
Channel: PJSIP/sipp-uac-00000000
CallID: [C-00000001]
PriorCallID:

Event: Newexten
Privilege: dialplan,all
Channel: PJSIP/sipp-uac-00000000
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Language: en
AccountCode:
Context: sipp-uac
Exten: s
Priority: 2
Uniqueid: test_call
Linkedid: test_call
Extension: s
Application: Answer
AppData:

Event: Newexten
Privilege: dialplan,all
Channel: PJSIP/sipp-uac-00000000
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Language: en
AccountCode:
Context: sipp-uac
Exten: s
Priority: 3
Uniqueid: test_call
Linkedid: test_call
Extension: s
Application: Hangup
AppData:

Event: VarSet
Privilege: dialplan,all
Channel: PJSIP/sipp-uac-00000000
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Language: en
AccountCode:
Context: sipp-uac
Exten: s
Priority: 3
Uniqueid: test_call
Linkedid: test_call
Variable: STASISSTATUS
Value: SUCCESS

Event: Hangup
Privilege: call,all
Channel: PJSIP/sipp-uac-00000000
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Language: en
AccountCode:
Context: sipp-uac
Exten: s
Priority: 3
Uniqueid: test_call
Linkedid: test_call
Cause: 16
Cause-txt: Normal Clearing

Event: DeviceStateChange
Privilege: call,all
Device: PJSIP/sipp-uac
State: NOT_INUSE

Event: QueueMemberStatus
Privilege: agent,all
Queue: test
MemberName: PJSIP/sipp-uac
Interface: PJSIP/sipp-uac
StateInterface: PJSIP/sipp-uac
Membership: static
Penalty: 0
CallsTaken: 0
LastCall: 0
LastPause: 0
InCall: 0
Status: 1
Paused: 0
PausedReason:
Ringinuse: 1
Wrapuptime: 0

Event: TestEvent
Privilege: reporting,all
Type: StateChange
State: SESSION_DESTROYING
AppFile: res_pjsip_session.c
AppFunction: session_destructor
AppLine: 2153
Endpoint: sipp-uac
AOR: <none>
Contact: <none>

Event: TestEvent
Privilege: reporting,all
Type: StateChange
State: SESSION_DESTROYED
AppFile: res_pjsip_session.c
AppFunction: session_destructor
AppLine: 2186
Endpoint: sipp-uac
{noformat}
Comments:By: Asterisk Team (asteriskteam) 2019-03-13 17:09:35.251-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
By: sungtae kim (pchero) 2019-03-13 17:14:06.122-0500

I'm working on this, will submit the patch soon. :)
By: Benjamin Keith Ford (bford) 2019-03-14 09:56:24.823-0500

I'll assign the issue to you. Thanks for your support!
By: Abhay Gupta (agupta) 2019-05-10 03:27:30.846-0500

I have a question , as soon as we do a continue from ARI and it goes to dialplan we get a STASIS END . Do you mean you want to send a ChannelHangupRequest after stasis has ended ?