Summary: | ASTERISK-18652: Asterisk Doesn't Release RTP ports as it should. | ||
Reporter: | Sarikov Gabriel (gabriel) | Labels: | |
Date Opened: | 2011-10-03 02:16:38 | Date Closed: | 2011-11-16 11:23:18.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Resources/General |
Versions: | 1.8.6.0 1.8.7.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | ubuntu natty 11.04 32bit | Attachments: | ( 0) ports.txt ( 1) ports2.txt ( 2) rtp.conf |
Description: | Asterisk Doesn't Release RTP ports as it should. | ||
Comments: | By: Sarikov Gabriel (gabriel) 2011-10-03 02:19:29.093-0500 only about 22 calls and about 900+ ports open. By: Sarikov Gabriel (gabriel) 2011-10-03 02:21:57.787-0500 one hour later 18 calls and more the before ports open By: Sarikov Gabriel (gabriel) 2011-10-03 02:27:55.038-0500 after all ports are used we got this message: [Oct 2 07:33:58] ERROR[26747] res_rtp_asterisk.c: Oh dear... we couldn't allocate a port for RTP instance '0xb3000bd0' [Oct 2 07:33:58] ERROR[26747] chan_sip.c: Got SDP but have no RTP session allocated. By: Sarikov Gabriel (gabriel) 2011-10-03 02:32:17.633-0500 on sip.conf we have : rtptimeout=60 By: Sarikov Gabriel (gabriel) 2011-10-03 03:20:05.859-0500 added this to sip.conf: session-timers=originate session-expires=900 session-minse=90 session-refresher=uas maybe it will solve this problem. By: Sarikov Gabriel (gabriel) 2011-10-03 09:30:00.692-0500 didn't solve the problem entirely still stuck sips. examples: * SIP Call Curr. trans. direction: Incoming Call-ID: 248248-352662xxx-633263@xxx.xxx.xxx Owner channel ID: <none> Our Codec Capability: 0x8 (alaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: 0xc (ulaw|alaw) Joint Codec Capability: 0x8 (alaw) Format: 0x0 (nothing) T.38 support No Video support No MaxCallBR: 384 kbps Theoretical Address: xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open Force rport: Yes Audio IP: xxx.xxx.xxx.xxx(local) Our Tag: as15306d4c Their Tag: 700315239615004xxxx SIP User agent: Peername: johndoe Original uri: sip:3503xxxx@xxx.xxx.xxx.xxx:5060 Caller-ID: 3503xxxx Need Destroy: No Last Message: Rx: BYE Promiscuous Redir: No Route: sip:3503xxxx@xxx.xxx.xxx.xxx:5060;user=phone DTMF Mode: rfc2833 SIP Options: timer Session-Timer: Inactive * SIP Call Curr. trans. direction: Incoming Call-ID: 1208549848_714xxxx@xxx.xxx.xxx.xxx Owner channel ID: <none> Our Codec Capability: 0x8 (alaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: 0x8 (alaw) Joint Codec Capability: 0x8 (alaw) Format: 0x0 (nothing) T.38 support No Video support No MaxCallBR: 384 kbps Theoretical Address: xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open Force rport: Yes Audio IP: 10.1.2.40 (local) Our Tag: as658d91c5 Their Tag: gK0958374c SIP User agent: Peername: johndoe2 Original uri: sip:044208806xxxx@xxx.xxx.xxx.xxx:5060 Caller-ID: 044208806xxxx Need Destroy: No Last Message: Rx: BYE Promiscuous Redir: No Route: sip:044208806xxxx@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options: 100rel timer Session-Timer: Inactive By: Leif Madsen (lmadsen) 2011-10-03 12:51:27.659-0500 Sounds like you're going to need to provide a backtrace from the running process and 'core show locks' output from the console. I'll post additional messages shortly with information about how to provide that information. By: Leif Madsen (lmadsen) 2011-10-03 12:51:42.122-0500 Thank you for your bug report. In order to move your issue forward, we require a backtrace[1] from the core file produced after the crash. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then: make install After enabling, reproduce the crash, and then execute the backtrace[1] instructions. When complete, attach that file to this issue report. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace By: Leif Madsen (lmadsen) 2011-10-03 12:52:22.570-0500 Debugging deadlocks: Please select DEBUG_THREADS and DONT_OPTIMIZE in the Compiler Flags section of menuselect. Recompile and install Asterisk (i.e. make install). This will then give you the console command "core show locks." When the symptoms of the deadlock present themselves again, please provide output of the deadlock via: # asterisk -rx "core show locks" | tee /tmp/core-show-locks.txt # gdb -se "asterisk" <pid of asterisk> | tee /tmp/backtrace.txt gdb> bt gdb> bt full gdb> thread apply all bt Then attach the core-show-locks.txt and backtrace.txt files to this issue. Thanks! By: Sarikov Gabriel (gabriel) 2011-10-04 03:28:47.246-0500 the problem is rtptimeout doesnt work at all on our previous version 1.6.2(rc2) it always cleared the "zombie channels" after 60 seconds. i cant do a trace on this machine its production server. By: Sarikov Gabriel (gabriel) 2011-10-04 03:32:08.960-0500 i think its the same issue as : ASTERISK-18559 By: Sarikov Gabriel (gabriel) 2011-10-04 08:02:59.536-0500 sip show channels: Peer User/ANR Call ID Format Hold Last Message Expiry Peer xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Tx: ACK B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: INVITE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE C xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE C xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Tx: ACK A xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE A xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK B xxx.xxx.xxx.xxx censored censored 0x0 (nothing) No Rx: BYE B xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: INFO C xxx.xxx.xxx.xxx censored censored 0x8 (alaw) No Rx: ACK C 52 active SIP dialogs all the channels with Rx: BYE are stuck channels. By: Sarikov Gabriel (gabriel) 2011-10-06 03:59:29.452-0500 restarted asterisk at 5am today(scheduld) after 5 hours at 10am more then 330 channels aren't released . even not one rtptimeout message in log. rtptimeout not working at all. By: Leif Madsen (lmadsen) 2011-11-01 09:12:39.560-0500 You still haven't provided the backtrace or 'core show locks' output as requested on October 3rd. There will be nothing we can do to move this issue forward without the requested information. By: Sarikov Gabriel (gabriel) 2011-11-16 07:11:58.218-0600 this issue was resolved on asterisk 1.8.7.1. no more ports staying open. By: Richard Mudgett (rmudgett) 2011-11-16 11:23:18.476-0600 Reporter stated that the issue is resolved. |