[asterisk-bugs] [JIRA] (ASTERISK-28725) Bridge error on incoming calls on asterisk 16.8.0 and 13.31.0

Asterisk Team (JIRA) noreply at issues.asterisk.org
Tue Feb 4 13:07:25 CST 2020


    [ https://issues.asterisk.org/jira/browse/ASTERISK-28725?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249655#comment-249655 ] 

Asterisk Team commented on ASTERISK-28725:
------------------------------------------

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.

> Bridge error on incoming calls on asterisk 16.8.0 and 13.31.0
> -------------------------------------------------------------
>
>                 Key: ASTERISK-28725
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28725
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/General
>    Affects Versions: 13.30.0, 16.7.0
>         Environment: Debian 9/asterisk 13 Debian 10/asterisk 16
>            Reporter: tootai
>
> As soon as a call is answered we see in logs
> Asterisk 16 log, incoming call from peer:
>     -- Executing [100 at to-MairieZWR:25] Goto("Local/PRESDA9140 at to-MairieZWR-00000000;2", "callEndStatus,s-CHANUNAVAIL,1") in new stack                         
>     -- Goto (callEndStatus,s-CHANUNAVAIL,1)                                                                                                                   
>     -- Executing [s-CHANUNAVAIL at callEndStatus:1] GotoIf("Local/PRESDA9140 at to-MairieZWR-00000000;2", "0?noVM") in new stack                                   
>     -- Executing [s-CHANUNAVAIL at callEndStatus:2] Set("Local/PRESDA9140 at to-MairieZWR-00000000;2", "CHANNEL(language)=fr") in new stack                        
>     -- Executing [s-CHANUNAVAIL at callEndStatus:3] VoiceMail("Local/PRESDA9140 at to-MairieZWR-00000000;2", "100 at MairieZWR") in new stack                         
>     -- Local/PRESDA9140 at to-MairieZWR-00000000;1 answered PJSIP/101743-00000000
> [2020-02-04 19:19:48] ERROR[3768][C-00000001]: stasis_bridges.c:199 bridge_topics_init: Bridge id initialization required                                    
> [2020-02-04 19:19:48] WARNING[3768][C-00000001]: bridge.c:809 bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f: Could not initialize topics     
>     -- <Local/PRESDA9140 at to-MairieZWR-00000000;2> Playing 'vm-intro.gsm' (language 'fr')                                                                     
>   == Spawn extension (callEndStatus, s-CHANUNAVAIL, 3) exited non-zero on 'Local/PRESDA9140 at to-MairieZWR-00000000;2'
> Asterisk 13 log, outgoing call to client:
> [2020-02-04 19:16:25] VERBOSE[17176][C-0000002c] app_dial.c: Called SIP/33388917474/33388917474                                                               
> [2020-02-04 19:16:25] VERBOSE[17176][C-0000002c] app_dial.c: SIP/33388917474-00000033 answered SIP/101742-00000032                                            
> [2020-02-04 19:16:25] ERROR[17176][C-0000002c] stasis_bridges.c: Bridge id initialization required                                                            
> [2020-02-04 19:16:25] WARNING[17176][C-0000002c] bridge.c: Bridge f2d5eff9-5fd8-4b1d-8463-d6fc97583421: Could not initialize topics                           
> [2020-02-04 19:16:25] VERBOSE[17176][C-0000002c] pbx.c: Spawn extension (from-TOBJECT, 33388917474, 106) exited non-zero on 'SIP/101742-00000032'             
> [2020-02-04 19:16:25] VERBOSE[17176][C-0000002c] pbx.c: Executing [h at from-TOBJECT:1] NoOp("SIP/101742-00000032", "Hangup Cause: 16") in new stack             
> [2020-02-04 19:16:25] VERBOSE[17176][C-0000002c] pbx.c: Executing [h at from-TOBJECT:2] NoOp("SIP/101742-00000032", "Dial status : ANSWER") in new stack
> And from the client side which is an asterisk:
>     -- Accepting AUTHENTICATED call from 10.1.58.11:
>     --        > requested format = ulaw,
>     --        > requested prefs = (ulaw|alaw|g722),
>     --        > actual format = ulaw,
>     --        > host prefs = (ulaw|alaw|g722),
>     --        > priority = mine
>     -- Executing [33388917474 at TOOTAiAudio-to-Cabinet_Medical:1] NoOp("IAX2/TOOTAi-1809", "") in new stack
>     -- Executing [33388917474 at TOOTAiAudio-to-Cabinet_Medical:2] GotoIf("IAX2/TOOTAi-1809", "0?dest:") in new stack
>     -- Executing [33388917474 at TOOTAiAudio-to-Cabinet_Medical:3] Answer("IAX2/TOOTAi-1809", "") in new stack
>   == Spawn extension (TOOTAiAudio-to-Cabinet_Medical, 33388917474, 3) exited non-zero on 'IAX2/TOOTAi-1809'
>     -- Hungup 'IAX2/TOOTAi-1809'
> Here you cabn see that as soon as the answer from client is done, the error appears on the other end.



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list