Summary: | ASTERISK-28336: After ARI continue, hangup() application does not create SoftHangupRequest event | ||
Reporter: | sungtae kim (pchero) | Labels: | pjsip |
Date Opened: | 2019-03-13 17:09:34 | Date Closed: | |
Priority: | Minor | Regression? | |
Status: | Open/New | Components: | 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 ? |