similar to: Asterisk executable suddenly about 40KB

Hi guys Thanks for the pointers - I'll look into the possible compromise scenario though I've got no idea how I'll counter it -if- I manage to detect it...! I've disabled prelinking (thanks Tony!) and I'll see if that helps. Interesting thing I've now discovered (had this failure again at the head office this morning) is the "growth" in the file's size is

>I would also start by putting an audit rule on the binary. Something like this: >auditctl -w /usr/sbin/asterisk -p war -k asterisk-bin >then you can get a report on who modified it and when by using: >ausearch -f /usr/sbin/asterisk >Its a start, but eventually you might need to monitor even keystrokes with pam_tty_audit.so to understand who is doing this:

Hi all I have a strange issue with 1.8.11.0 on a production Asterisk machine at our head office, and the same issue with a production machine at a branch office. Every now and then, on the head office machine, ODBC CEL and CDR logging will stop working. On examination in the CLI, Asterisk behaves as if the config files for ODBC in the /etc directory are just gone. Repeated tests have then

Hi, do you have NAT between Asterisk and agent phones? S pozdravem Tomáš Holý Hi Tomas Thanks for replying. Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw

Hi all Asterisk 13 instance - I've got a situation in an agent queue that an agent will be talking to one person, then suddenly the same agent will be talking to another person who was talking to another agent. The calls do not switch around between the two agents, the "losing" agent will just suddenly have silence in his handset and the other agent will now be talking to

Hi Marie Thanks! I was just worried about thread safety if I had to use a global variable, e. g. it might be set to a value by one call (since I'm using the same global for every incoming call to transfer the accountcode gotten from my HTTP endpoint to the same macro, and there can be several calls simultaneously all inserting HTTP-sourced values at more or less the same instant) and then

Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'.

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

Hi Joshua HTTP is used on in our setup on 127.0.0.1/mxml?<command> to send commands to the server, such as http://127.0.0.1/mxml?action=login&username=myuser&secret=thesecret to log in and then http://127.0.0.1/mxml?ActionID=123&Action=BlindTransfer&Channel=Channel&Context=local&Exten=123&Priority=1 etc. to control transfers, for example. ARI is not being

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Hi all I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 - 32GB RAM. When I approach about 320 channels, I -sometimes- get thousands of these messages suddenly streamed in the CLI / Asterisk log: WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length queuing to Local/xxxxxxxxxx at local-0002dbea;2 WARNING[71993][C-00022dcc] channel.c: Exceptionally

Hi, We''re using thin stored configs heavily to manage things like Munin and Nagios and we''re currently wondering what to do when nodes go away, or change considerably. If we get rid of a server then all it''s stored config is still present in the database, as well as it''s nagios and munin configurations. We created a script that iterates through the database

Here's another data point regarding the GDM problem I'm having. Once again, I get this on the console: There was an error loading the theme Default Couldn't recognize the image file format for file '/usr/share/gdm/themes/Default/background.png' Yesterday I needed to edit some graphics I had done for a job. I use gimp. Now gimp tells me: Image type 'png' is not

Hi all I need to pass a parameter in a thread-safe manner to the Queue pickup macro. This is to know when (and who) picked up an incoming call to a queue and log that to my back-office system with a CURL to a HTTP endpoint. However, the Queue application does not appear to allow passing of parameters to the called queue pickup macro. E. g. non-working code is: [queuetest] timeout = 60 retry =

Folks: I''ve started to play around with the comprehensive integrated module set from example42 (http://www.example42.com/). Following the instructions, I set up one node as a nagios server, with ''include nagios''. I set up another node to be monitored via nagios, with ''include nagios::target". I set $monitor=true for the base node, which is inherited by

Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13. Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script. This works, and reliably calls the script:

Hi all I'm trying to limit the maximum concurrent calls on my Asterisk to try and mitigate another problem I posted about earlier. I've edited /etc/asterisk/asterisk.conf And uncommented this line, and put a value of 60 in there: maxcalls = 60 in an effort to limit my Asterisk to 60 simultaneous calls. I did a core reload in the CLI after doing that.

What are the pros and cons of disabling prelinking in CentOS 4? From what I understand, prelinking is ment to be performance improvement (faster loading time for binaries). I can see some potential benefit on workstations where many different processes are started frequently. But how big is that benefit in real life? Would I slow down things noticably if there is no prelinking? Is

Hi all, I am new to puppet and also newcomer in this group. I heart good things about puppet and I decided to give it a try 2 weeks now. After reading various documentation, I began using the complete configuration collection of puppet modules (thanks to David). But I have a problem with the munin module. It works fine, except that it seems to not export virtual ressources. Let me give more detail

Hello list. I''ve the following parent class: class munin::client { ... munin::client::plugin { "postfix_mailqueue": name => "postfix_mailqueue", enable => "false" } } And the following inherited class: class munin::client::plugins::postfix inherits munin::client { munin::client::plugin {