|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|
|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 188.8.131.52 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.
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
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