User logon / logoff does not work

Hello

After upgrading FreePBX from 2.10 to 2.11 user logoff stopped working:

  • user logins to phone. If he calls to *65 he hears correct extension
  • if someone calls to the extension, Asterisk tells it is unavailable
  • logging out says user logged out successfully but if user calls to *65 again from the phone it tells the users is still logged in

I checked FreePBX and asterisk configuration:

  • Device and user mode enabled on web interface
  • Dynamically Generate Hints set to true (but if I change it, apply button does not appear)
  • DYNAMICHINTS set to 1 in freepbx_settings MySQL table
  • DYNAMICHINTS=TRUE in /etc/amportal.conf
  • execincludes=yes set in /etc/asterisk/asterisk.conf

I use custom FreePBX installation and it worked fine before.
Any ideas?

Thanks.

Update:
user successfully logged in to a phone which is configured in Adhoc mode. They user logs off:

[2013-12-02 15:46:05] VERBOSE[10943] res_agi.c: -- AGI Script Executing Application: (UserEvent) Options: (UserDeviceRemoved,Data: 519\,5191) [2013-12-02 15:46:05] VERBOSE[10943] res_agi.c: -- AGI Script user_login_out.agi completed, returning 0 [2013-12-02 15:46:05] VERBOSE[10943] pbx.c: -- Executing [[email protected]:4] Playback("SIP/5191-00000432", "agent-loggedoff") in new stack [2013-12-02 15:46:05] VERBOSE[10943] file.c: -- Playing 'agent-loggedoff.ulaw' (language 'uk') [2013-12-02 15:46:05] NOTICE[10943] channel.c: Dropping incompatible voice frame on SIP/5191-00000432 of format alaw since our native format has changed to 0x4 (ulaw) [2013-12-02 15:46:07] VERBOSE[10943] pbx.c: -- Executing [*[email protected]:2] Hangup("SIP/5191-00000432", "") in new stack [2013-12-02 15:46:07] VERBOSE[10943] pbx.c: == Spawn extension (from-internal, *12, 2) exited non-zero on 'SIP/5191-00000432' [2013-12-02 15:46:07] VERBOSE[10943] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/5191-00000432", "") in new stack [2013-12-02 15:46:07] VERBOSE[10943] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5191-00000432'

but if user tries to login again to the phone:

[2013-12-02 15:46:43] VERBOSE[10953] pbx.c: -- Executing [[email protected]:7] Read("SIP/5191-00000433", "AMPUSER,please-enter-your-extension-then-press-pound,,,4") in new stack [2013-12-02 15:46:43] VERBOSE[10953] file.c: -- Playing 'please-enter-your-extension-then-press-pound.slin' (language 'uk') [2013-12-02 15:46:47] VERBOSE[10953] app_read.c: -- User entered '519' [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:8] GotoIf("SIP/5191-00000433", "0?s-MAXATTEMPTS,1") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/5191-00000433", "0?s-NOUSER,1") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:10] Set("SIP/5191-00000433", "AMPUSERPASS=1111") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:11] GotoIf("SIP/5191-00000433", "0?s-NOPASSWORD,1") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:12] Set("SIP/5191-00000433", "DEVICEUSER=519") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:13] GotoIf("SIP/5191-00000433", "1?s-ALREADYLOGGEDON,1") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Goto (macro-user-logon,s-ALREADYLOGGEDON,1) [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:1] NoOp("SIP/5191-00000433", "This device has already been logged into by this user") in new stack [2013-12-02 15:46:47] VERBOSE[10953] pbx.c: -- Executing [[email protected]:2] Playback("SIP/5191-00000433", "vm-goodbye") in new stack

Any ideas what it may be?

Let me know if you need more info.

It looks like this bug:
http://www.freepbx.org/forum/freepbx/beta-program-issues/devicesanduser-login-logout-not-working-correctly

Did anyone see this? It looks like a bug.
Does it worth to report about the issue to http://issues.freepbx.org/?
Or maybe there is other way to contact developers?

A bug report would need to be opened. Without a bug report no developer will see or resolve it.

Thanks for the answer.
I created an issue: http://issues.freepbx.org/browse/FREEPBX-6998