Summary: | ASTERISK-17101: SIP crash on transfer | ||
Reporter: | Cédric Chantepie (cchantep) | Labels: | |
Date Opened: | 2010-12-13 12:52:53.000-0600 | Date Closed: | 2011-02-25 13:01:34.000-0600 |
Priority: | Critical | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) backtrace.txt ( 1) full.4 ( 2) transferts-bug-lock.txt | |
Description: | When trying to transfer a call from an Aastra SIP phone (6755i) -- A<-->B(Aastra)<-->C -- nonetheless call failed, but then Asterisk no longer manager any SIP function. Main Asterisk process still works, "sip show users" still displays users, but phone are no longer registered, hard/soft phone cannot perform any new registeration, no call can be done. Even if I try to send a REGISTER UDP message through netcat from Asterisk machine, I don't see anything in a Asterisk console (with sip debug enabled). It seems that a major part a SIP handling is then broken. Moreover I can't reload sip module from Asterisk console. I need to kill it before restarting service. ****** ADDITIONAL INFORMATION ****** CentOS 5.5 / kernel 2.6.18-194.26.1.el5 Asterisk 1.8 installer from yum (digium+asterisk repositories) Only SIP channels (no ISDN or other things ...) PostgreSQL Realtime (with rtcachefriends for presence support) | ||
Comments: | By: Kun Liu (knkcn) 2010-12-13 21:48:04.000-0600 It should be sip deadlock issue as mine(issue ASTERISK-1795204). It is fixed from 1.8.1-rc1. I sugguest you to run 1.8.1 version. By: Leif Madsen (lmadsen) 2010-12-16 10:31:32.000-0600 Please test 1.8.1.1 or later (1.8.2-rc1 is now available, along with the branch directly). By: Cédric Chantepie (cchantep) 2010-12-17 07:53:33.000-0600 Even with 1.8.2-rc2. It seems related with Realtime dialplan. Edgecore SIP phone also triggers that problem. By: Jonathan Thurman (jthurman) 2010-12-21 13:51:54.000-0600 This sounds like a duplicate of 18403, or at least related. By: Leif Madsen (lmadsen) 2011-01-19 09:51:22.000-0600 If this is a duplicate of ASTERISK-17046 then revision 301790 or later in the 1.8 branch should resolve this. Please test after that revision and report back. By: Bernard Merindol (bernard merindol) 2011-01-24 08:02:12.000-0600 I have tested on my labs version 1.8.3-rc1 on this isue with 3 aastra phone. The bug is same. When I use 3 aastra phones, A call B, B tansfert (attended) A to C, after hangup B the sip part is hang on asterisk. By: Leif Madsen (lmadsen) 2011-01-24 08:09:19.000-0600 OK, then someone needs to perform the test and provide a SIP trace of the transfer from the Asterisk console along with SIP history, and the dialplan in use. By: Bernard Merindol (bernard merindol) 2011-01-24 08:29:15.000-0600 I send my full: A: Poste 1000 B: Poste 1001 C: Poste 1002 A Call B, B transfert to C , when C pickup B speak with C and finish transfert. I have debug and verbose at 10. sip set debug on sip set history on By: Leif Madsen (lmadsen) 2011-01-24 14:57:44.000-0600 Backtrace and 'core show locks' need to be provided as well. https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace By: Bernard Merindol (bernard merindol) 2011-01-25 01:36:26.000-0600 I have send the result of core show locks and backtrace. For backtrace Asterisk is not dead (don't have core) I have get back trace with this command. gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch asterisk 30388 Where 30388 is the processID of asterisk, I hope is good. By: Bernard Merindol (bernard merindol) 2011-01-28 01:21:52.000-0600 No News ? You need more feedback ? By: Leif Madsen (lmadsen) 2011-02-07 14:41:24.000-0600 num = <value optimized out> ch = <value optimized out> That typically means asterisk wasn't compiled with DONT_OPTIMIZE in the Compiler Flags section of menuselect. Please install Asterisk with that option enabled and submit the backtrace as previous. By: Andrew Latham (lathama) 2011-02-07 14:43:59.000-0600 https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace just add 1. make menuselect.makeopts 2. menuselect/menuselect --enable DONT_OPTIMIZE menuselect.makeopts 3. menuselect/menuselect --enable DEBUG_THREADS menuselect.makeopts to your normal build process before you run make... By: Alec Davis (alecdavis) 2011-02-23 16:37:33.000-0600 please try the patch bug18837.diff3.txt on ASTERISK-17431 This may solve the deadlock issue. By: Alec Davis (alecdavis) 2011-02-25 13:01:32.000-0600 Fix in 1.8 branch and trunk as of r308945 See issue ASTERISK-17431 for details |