| Summary: | ASTERISK-26557: Client can't reconnect to a conference with announcements if the client is suddenly killed | ||
| Reporter: | Pablo Saavedra (psaavedra) | Labels: | |
| Date Opened: | 2016-11-05 14:51:06 | Date Closed: | 2016-11-05 14:54:33 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | . I did not set the category correctly. |
| Versions: | 13.11.2 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | GNU/Linux Debian Jessie Client LinphoneAndroid/3.1.1 (belle-sip/1.4.2) | Attachments: | |
| Description: | Observed in conferences with announcements. If the client (Linphone for Android, for example) suddenly fails for some reason (can be reproduce doing a power-off of the device during the call, for example) it is not possible the reconnection to the same conference room with the same client anymore.
I've observed that confbridge doesn't realize about the peer has been disconnected because you still see the channel of the peer in the Asterisk's stats (sip show channelstats) as active. One way to recover from that status is to kick-off manually the related channel to the particular client. A relevant thing here is that this only happens in conferences rooms with announcements. The rooms/users without announcements are safe about this. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2016-11-05 14:51:07.499-0500 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]. By: Pablo Saavedra (psaavedra) 2016-11-05 14:51:43.876-0500 Issue looks pretty similar to this one ASTERISK-22454. {quote} I have had this issue on production systems, it was a new feature we added, so we disabled it and we are no longer using the announcement feature. {quote} By: Joshua C. Colp (jcolp) 2016-11-05 14:54:33.640-0500 This is not a bug in Asterisk. There are a few options in both chan_sip and chan_pjsip which can be used to detect that a channel has gone away. One of them is an RTP timeout that will terminate the channel if RTP is not received within a period of time and the other is session timers which sends a SIP request to the other side and if no response is received terminates the channel. Without either of these there is no way for Asterisk to know that the channel has gone away. By: Pablo Saavedra (psaavedra) 2016-11-05 14:57:18.686-0500 The issue looks pretty linked to {{announce_user_count}} announce. Disabling it the client can rejoin into the room even the if the previous channel is still active and registered in the room. {panel:title=Diff|borderStyle=dashed|borderColor=#ccc|titleBGColor=#F7D6C1|bgColor=#FFFFCE} #!diff diff --git a/confbridge.conf b/confbridge.conf index e773065..0426eed 100644 --- a/confbridge.conf +++ b/confbridge.conf @@ -144,7 +144,7 @@ admin=no music_on_hold_when_empty=yes music_on_hold_class=default quiet=no -announce_user_count=yes +announce_user_count=no announce_user_count_all=no announce_only_user=no dsp_drop_silence=yes {panel} By: Asterisk Team (asteriskteam) 2016-11-05 14:57:18.823-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Joshua C. Colp (jcolp) 2016-11-05 15:05:09.764-0500 If you believe there is an actual problem regardless you'll need to attach console output including SIP traffic. (sip set debug on or pjsip set logger on depending on SIP channel driver) By: Asterisk Team (asteriskteam) 2016-11-21 12:00:01.530-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||