| Summary: | ASTERISK-29273: Incorrect off-hold on ReINVITE via Replaces | ||||
| Reporter: | Igor Olhovskiy (IhorOlkhovskyi) | Labels: | |||
| Date Opened: | 2021-02-01 07:45:44.000-0600 | Date Closed: | 2021-02-01 08:32:04.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Bridges/bridge_holding Bridges/bridge_softmix | ||
| Versions: | 13.38.0 16.16.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | CentOS 7 | Attachments: | ( 0) ast16_failed_offhold.pcap | ||
| Description: | When the call is off-hold via ReINVITE with Replaces, media is not being established back correctly.
SIP Client A - Asterisk - SIP Client B SIP Client A calls Asterisk and Asterisk bridges it to SIP Client B. SIP Client B puts Client A on hold (a=sendonly), Client A listens to MoH. All ok. Then client B makes internal transfers (it's PBX) and reINVITE's Asterisk with Out-of dialog ReINVITE with Replaces header. Client A still hearing Music On Hold, but what Client A said, is being heard by Client B. But what Client B is said, cannot be heard by Side A. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2021-02-01 07:45:46.359-0600 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Igor Olhovskiy (IhorOlkhovskyi) 2021-02-01 07:48:01.397-0600 pcap of scenario with RTP is attached. All network is plan and routable, there is no NAT or something like this. By: George Joseph (gjoseph) 2021-02-01 08:21:20.595-0600 Igor, It looks like this is related to ASTERISK-29253. Can you try unloading the bridge_simple module and see if the calls work correctly? It's not a fix but it'll give us more info about the cause. Thanks. By: Igor Olhovskiy (IhorOlkhovskyi) 2021-02-01 08:26:11.973-0600 Yes, with unloading `bridge_simple` it works as expected. Call cuts after 2nd ReINVITE with replace, but not sure it's Asterisk issue. By: George Joseph (gjoseph) 2021-02-01 08:31:00.475-0600 OK, Thanks for checking. I'm going to close this issue and link it to the other one. | ||||