| Summary: | ASTERISK-24773: Hung call in "sip show channels" not listed in "show channels" | ||
| Reporter: | jessup ross (jessup.r) | Labels: | |
| Date Opened: | 2015-02-09 14:59:16.000-0600 | Date Closed: | 2015-02-09 20:22:34.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | 1.8.15.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | We have a system with 415 active SIP dialogs but only 42 active channels
23 active calls. Clearly we don't have that many calls in setup/tear-down. These hung calls can't be hung up and they are rapidly eating our bandwidth and RTP ports (1000). rtptimeout=60 is currently commented out, does this affect our issue or only with user=peer? The only way we have found to clean them is an asterisk restart and we can't do that in-production. Is this a patch or mis-configuration issue? | ||
| Comments: | By: Matt Jordan (mjordan) 2015-02-09 20:22:27.008-0600 Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.8 branch has ended. For continued maintenance support please move to the 11 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. After testing with the latest Asterisk 11 release, if you find this problem has not been resolved, please open a new issue against Asterisk 11. | ||