| Summary: | ASTERISK-27984: PJSIP: Stuck channel when using conference on an LTE connection -> Wifi | ||
| Reporter: | ADTopkek (ADTopkek) | Labels: | pjsip |
| Date Opened: | 2018-07-24 12:35:55 | Date Closed: | 2018-07-24 13:08:55 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | pjproject/pjsip |
| Versions: | 13.21.1 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Sangoma 7 (Centos 7) updated as far as it can be as of the date of this post. | Attachments: | |
| Description: | - I was on a conference call using the default SIP client on android phones.
- My extension is using PJSIP with 3 contacts - I had connected to the conference call by dialing an internal number to enter the conference and I was on the LTE network - I was walking and went into an LTE deadzone. My phone showed "Signal lost". - It switched to WIFI - After the conference ended my extension showed busy but was not on DND - I checked "pjsip show channels" and got this: Channel: <ChannelId........................................> <State.....> <Time.....> Exten: <DialedExten.............> CLCID: <ConnectedLineCID.......> ========================================================================================== Channel: PJSIP/1170-0000612b/ConfBridge Up 00:33:34 Exten: STARTMEETME CLCID: "" <> Channel: PJSIP/1170-0000630f/ConfBridge Up 00:10:38 Exten: STARTMEETME CLCID: "" <> My extension has a stuck channel in the conference bridge. I assume its because it completely lost signal on the LTE so no end signals were sent. I called a 2nd time to replicate it. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-07-24 12:35:56.293-0500 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: Joshua C. Colp (jcolp) 2018-07-24 12:40:25.573-0500 Multiple methods are configurable in pjsip.conf to determine if a channel has died without Asterisk being told. You can configure it to use SIP signaling to send a request and if no response is received then the channel will be hung up[1]. You can also configure it to hang up the channel if no media flows for a period of time[2]. [1] https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L668 [2] https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L776 By: ADTopkek (ADTopkek) 2018-07-24 13:06:41.282-0500 Interesting. Perhaps this is a Freepbx bug then. I will pass it onto them thanks! By: Asterisk Team (asteriskteam) 2018-07-24 13:06:41.514-0500 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: ADTopkek (ADTopkek) 2018-07-24 13:07:48.936-0500 Probably a Freepbx thing. | ||