| Summary: | ASTERISK-14731: [patch] sip session timer: Does not work if initial INVITE min-se timer is too small | ||||
| Reporter: | Johann Steinwendtner (steinwej) | Labels: | |||
| Date Opened: | 2009-09-23 10:55:23 | Date Closed: | 2015-01-19 20:10:02.000-0600 | ||
| Priority: | Minor | Regression? | No | ||
| Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
| Versions: | Frequency of Occurrence | ||||
| Related Issues: |
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| Environment: | Attachments: | ( 0) full_sip_session-timer.gz ( 1) session_timer_422.patch | |||
| Description: | Asterisk sip with session timer enabled. sip.conf: session-timers=accept session-expires=600 session-minse=180 Patton box connected to asterisk. Patton sends INVITE with session timer 90 asterisk responds with 422 session interval too small patton reinvites with the proposed session timer. asterisk send 200 ok, nothing happens. no tones or anything. When patton sends BYE, asterisk sends ACK But sip channels remains, audio ports are not released voip-1*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message 91.128.104.50 (None) 302e3db1464e650 0x0 (nothing) No Rx: OPTIONS 91.128.104.50 test_user 9e2ec18f1622d61 0x8 (alaw) No Rx: BYE 2 active SIP dialogs voip-1*CLI> voip-1*CLI> core show channels Channel Location State Application(Data) SIP/test_user-b7 01229922640@from_sip Down (None) 1 active channel 0 active calls 0 calls processed voip-1*CLI> | ||||
| Comments: | By: Johann Steinwendtner (steinwej) 2009-09-23 13:03:25 The problem was not there in 1.6.0.10. See patch. By: Leif Madsen (lmadsen) 2009-09-24 09:07:43 I'm not 100% sure if 15890 is related, but I'm marking it as such for now as they appear similar. By: Paul Belanger (pabelanger) 2010-06-01 10:50:45 Is this still an issue with the 1.6.2 branch (see below). --- Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.6.0 and 1.6.1 branches has ended. For continued maintenance support please move to the 1.6.2 branch. More information on this change can be found in the release announcement: http://www.asterisk.org/node/49924 By: Johann Steinwendtner (steinwej) 2010-06-07 13:28:28 Yes. Tested with 1.6.2.8 By: Corey Farrell (coreyfarrell) 2014-11-09 05:29:45.669-0600 I believe this is related to ASTERISK-23373. The channel/RTP leaks should be resolved by this. Is the REINVITE issue resolved? Note 1.6 is out of support, so you need to be running 1.8.26.1, 11.8.1, 12.1.1 or higher. By: Matt Jordan (mjordan) 2015-01-19 20:09:58.312-0600 I'm going to go with Corey on this. If it turns out I was wrong, please feel free to comment here (whoever) and we'll reopen the issue. | ||||