similar to: monitoring asteriks

Hi all, I''m trying to come up with a lens for the zabbix agent config files. I find the lens language untransparent at best, so I''m struggeling to figure out what''s up. The debugging possibilities are extremely limited. Here''s what I have now: zabbix.aug: ==== (** An adjusted copy of the postfix_main module **) module Zabbix_agent = autoload xfm

Hi, group! I have such strings on my Freebsd 7.0 + puppet 0.24.8 server: ===================CUT=================== define zabbix_agent_conf($zabbixserverip, $zabbixagenthostname, $zabbixagentip, $startagents="") { file { "/var/tmp/zabbix_agentd.conf": owner => root, group => wheel, mode => 444, backup

I''m running a previously working set of modules with the Puppet master version 3.1.0-rc2. I''m getting the Error: stack level too deep Here is a chunk of the debug Debug: Scope(Class[Zabbix]): Retrieving template zabbix/zabbix.conf.php.erb Debug: template[/etc/puppet/environments/production/modules/zabbix/templates/zabbix.conf.php.erb]: Bound template variables for

Hi everybody, what are the current options to get an Asterisk-system high available? Using two servers as active/passive with DRBD, Pacemaker/Corosync works very good, there are no quality issues of the voice quality, even not on high loaded servers and no problems with a lot of small packages. But for this you need two systems for every Asterisk-system, what is not "economic" in any

2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>: > > Do you really want to detect "ChallengeSent"? That should occur also on > legitimate login processes... > Hi , strange thing is that I still have not this asterisk in production and I see many attempts Connection. Now keep in mind that when a connection of authentication is successful the

Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)?

Is anyone using Zabbix to monitoring Xen at the dom0 level? Right now I have the Zabbix agent running in each domU but I would like to get per domU and aggregated statistics from the hyper visor perspective. I see a bunch of ''hard'' ways to collect these stats but first I want to make sure there isn''t some ''magic pill'' for Zabbix / Xen that I might of

Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx

Hi all, class zabbix { file { "/etc/zabbix/": ensure => directory, recurse => true, purge => false, force => false, owner => "root", group => "root", mode => 0755, source => $operatingsystem ? { solaris => "puppet:///zabbix/solaris",

Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one

Help, Any ideas ? I guess I missing something. make[1]: Entering directory `/usr/src/asterisk/utils' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-HEAD-12/27/04-21:28:39\" -DASTERISK_VERSION_NUM =999999 -DINSTALL_PREFIX=\"\"

Greetings, I have one centos server for network monitoring. there are remote devices which are connected through ADSL lines and hence Dynamic IPs Q1. Is there any tool which is capable of handling this type of situation? Q2. Is there a workaround for this problem Regards Rajagopal

Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase

Do you have DISA setup? We're seeing lots of attackers running scripts that send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it wasn't a stupid employee forwarding an inbound call to a 9xxxxxxx number etc). Have a look at SecAst (www.generationd.com) - it detects callers sending too many digits, monitors digit dialing speeds, etc. to help identify and block

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue