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Ast11: How To See Call Progress Like In Ast <= 1.8

Hi,



I just did a test install of Ast 11, and have trouble getting the same logging information that Ast 1.x provided. I’m looking specifically for the logging around call progress / dialplan actions.



In ASt 11 I’ve done the same thing that I did before: core set verbose 60



I also tried overwriting the logger.conf with the distribution one from Ast
11, and setting option “logger set level verbose on” (never did that on older versions, but was wondering if that would make a difference).



Still no joy, Googling around for an answer I did see a changelog with an example of the Call Identifier that shows a detailed logline (of level verbose, something I don’t get in 11). ButI’ve been unable to find an answer.



Any hints/tips, I must be overlooking something basic..



TX!

Bas

JAMI Interface Not Working As Expected

I have encountered a problem on an Ast 1.8 install where the AMI interface works fine (I can login and issues command to AMI directly using telnet), but when I use the JAMI/JAMA interface it fails. Using curl (HTTP) I can login fine to the JAMI, but as soon as I issue a second command it fails (despite saving session cookies and extending the httptimeout on the Asterisk side).

The two JAMI commands (login, then ping) are issued within 3 seconds of one another, so this should succeed (and it does on some other test systems). I read about this bug in Ast 1.6 but it should have been fixed by 1.8. See here for more details (http://noojee.com.au/forum/noojee-click/bug-reports/permission-denied-with-ajam/)

This happens to be a FreePBX system…not sure if that makes a difference.?

Ideas? Thanks!

AST-2012-011: Remote Crash Vulnerability In Voice Mail Application

If a single voicemail account is manipulated by two parties simultaneously, a condition can occur where memory is freed twice causing a crash.

Management of the memory in question has been reworked so that double frees and out of bounds array access do not occur. Upgrade to the latest release.

Affected Versions

  • Product Release Series
  • Asterisk Open Source 1.8.x 1.8.11 and newer
  • Asterisk Open Source 10.x 10.3 and newer
  • Certified Asterisk 1.8.11-certx All versions
  • Asterisk Digiumphones 10.x.x-digiumphones All versions

Corrected In

  • Product Release
  • Asterisk Open Source 1.8.13.1, 10.5.2
  • Certified Asterisk 1.8.11-cert4
  • Asterisk Digiumphones 10.5.2-digiumphones

In ooEndCall call state is – OO_CALL_CLEAR (incoming, ooh323c_1)

We have an Ast 1.6 installation which is connected to an Avaya using ooh323. Something is causing the log to fill with “In ooEndCall call state is – OO_CALL_CLEAR (incoming, ooh323c_1)” messages every 100ms. This causes the log to grow to 300MB in just 5 minutes, which eventually overloads the box.

Looking through the ooh323 log below, I suspect this stems from the “Error:Failed to enqueue ReleaseComplete message to outbound queue.(incoming, ooh323c_1)” message – but we don’t don’t see enough H323 installations to dig deeper. Can someone offer some suggested causes and resolutions?

Thanks!

audiohook errors

Hi,

I´m facing some issues on asterisk 1.8.10. I can see this on the console:

[May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex: Operation not permitted
[May 28 15:46:19] ERROR[28099]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 688 (audio_audiohook_write_list): Error obtaining mutex: State not recoverable
[May 28 15:46:19] ERROR[28099]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): mutex ‘&(audiohook)->lock’ freed more times than we’ve locked!

any ideas???

best regards

parsing issue

I get an error when I execute this code
exten => rejected,n,Hangup($[-1*${Z}])

May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
‘-’ or ‘!’ or ‘(‘ or ‘‘; Input:
-1*

The variable “Z” has a negative number, which is the code that I need
to use in the hangup.
Any idea how can I do this? There is no ABS() function in Asterisk. I
already filed a request for it but it turns up that it will cost me
money. How can I remove the sign from a number?
Philip