| Summary: | ASTERISK-28725: Bridge error on incoming calls on asterisk 16.8.0 and 13.31.0 | ||
| Reporter: | tootai (tootai) | Labels: | |
| Date Opened: | 2020-02-04 13:06:48.000-0600 | Date Closed: | 2020-02-05 06:28:14.000-0600 |
| Priority: | Major | Regression? | Yes |
| Status: | Closed/Complete | Components: | Applications/General |
| Versions: | 13.30.0 16.7.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Debian 9/asterisk 13 Debian 10/asterisk 16 | Attachments: | |
| Description: | As soon as a call is answered we see in logs
Asterisk 16 log, incoming call from peer: -- Executing [100@to-MairieZWR:25] Goto("Local/PRESDA9140@to-MairieZWR-00000000;2", "callEndStatus,s-CHANUNAVAIL,1") in new stack -- Goto (callEndStatus,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@callEndStatus:1] GotoIf("Local/PRESDA9140@to-MairieZWR-00000000;2", "0?noVM") in new stack -- Executing [s-CHANUNAVAIL@callEndStatus:2] Set("Local/PRESDA9140@to-MairieZWR-00000000;2", "CHANNEL(language)=fr") in new stack -- Executing [s-CHANUNAVAIL@callEndStatus:3] VoiceMail("Local/PRESDA9140@to-MairieZWR-00000000;2", "100@MairieZWR") in new stack -- Local/PRESDA9140@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@to-MairieZWR-00000000;2> Playing 'vm-intro.gsm' (language 'fr') == Spawn extension (callEndStatus, s-CHANUNAVAIL, 3) exited non-zero on 'Local/PRESDA9140@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@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@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@TOOTAiAudio-to-Cabinet_Medical:1] NoOp("IAX2/TOOTAi-1809", "") in new stack -- Executing [33388917474@TOOTAiAudio-to-Cabinet_Medical:2] GotoIf("IAX2/TOOTAi-1809", "0?dest:") in new stack -- Executing [33388917474@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. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2020-02-04 13:06:49.878-0600 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: George Joseph (gjoseph) 2020-02-04 13:18:57.234-0600 We're looking at this now but can you confirm that this did not happen with 13.30 or 16.7? Can you reproduce the issue with a simple sip to sip call? Are you able to compile from git? If so, can you clone https://gerrit.asterisk.org/asterisk, check out branch 16.8, and build and test again? If you can still reproduce the issue, can you try reverting commit dd82ebecd3 and try to reproduce again? By: Joshua C. Colp (jcolp) 2020-02-04 13:23:06.754-0600 As well, how did you apply this change in the first place - did you download the tarball, extract, and build? By: tootai (tootai) 2020-02-04 16:22:52.527-0600 - Yes, 13.30 and 16.7 are working correctly - The above 16.8 logs are sip to sip call - For a git clone we can do it tomorrow - We applied the 13.31.0-patch and 16.8.0-patch to the previous sources By: Joshua C. Colp (jcolp) 2020-02-04 16:27:06.451-0600 If you do "make clean" and then rebuild using your patched version, does it then work? By: tootai (tootai) 2020-02-05 06:26:11.810-0600 After make clean it's OK on both versions. Thanks for your support | ||