Re: [asterisk-users] Concurrent call monitoring
I wrote my own shell scripts to collect core show calls value from
asterisk and then push the filtered value to an opensource monitoring tool.
That worked perfectly well.
#!/usr/bin/perl -w
use strict;
open(LINE, 'asterisk -rx core show channels|');
my ($chans, $calls, $line)=(0,0,undef);
while ($line = LINE)
{
$calls = $1 if ($line =~ /^(\d+) active call/);
}
close(LINE);
printf $calls;
On Tue, Oct 25, 2011 at 6:40 PM, Danny Nicholas da...@debsinc.com wrote:
The Simplest method of seeing the number of concurrent calls is service
asterisk status. If I understand question two, asterisk -rx core show
channels verbose is probably your best bet.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, October 25, 2011 8:29 AM
To: Asterisk Users
Subject: [asterisk-users] Concurrent call monitoring
Hi
What are people using to monitor the concurrent number of calls at any
given
time?
Also, is there any good way of monitoring concurrent inbound and outbound
calls so that we can see the 2 different numbers?
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OPTIONS to query endpoint capability
I have been sending OPTIONS requests both programatically (my own code), manually via SIP VERIFY PEER x and automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice the Asterisk generated OPTIONS does not specify any Accept header (ie Accept=application/sdp). I was thinking maybe that is why I don't get any SDP coming back. My own code generated OPTIONS includes the Accept header and still I see no SDP. Is Anyone able to query codec capability for any endpoints? I would like to know how you do so. Below is excerpt from the automatic OPTIONS query I see in the sip logs from setting verify=true. No Accept header. Does anyone believe that to be the problem? Notice the response has content length=0 and no SDP. Any ideas appreciated OPTIONS sip:991@192.168.1.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169 Max-Forwards: 70 From: asterisk sip:asterisk@192.168.1.2;tag=as1fd2a50c To: sip:991@192.168.1.4:5060 Contact: sip:asterisk@192.168.1.2:5060 Call-ID: 010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Mon, 24 Oct 2011 19:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP read from UDP:192.168.1.4:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7f05f169 From: asterisk sip:asterisk@192.168.1.2;tag=as1fd2a50c To: sip:991@192.168.1.4:5060;tag=003d3418e2fce011b081701a0413e3f3 Call-ID: 010fdb653903a2022b99ed1d40c0b8db@192.168.1.2:5060 CSeq: 102 OPTIONS Contact: sip:991@192.168.1.4:5060 Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Server: SIPPER for PhonerLite Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting for Asterisk Call Center
Dear wcselby; Thanks a lot for your reply. For below script, I have some questions if you can help me: 1) I am looking to have reports for the call center, so I need to determine how many calls in the queue, and how many agents logged and when the agent logged in and when logged out ... etc. But I do not see this in the fields created for the CDR, only what I see those fields related to the call it self. So how I can get this? 2) What is required from me to be done to have CDR for the call center? From where to be enabled? In that case, from where I can determine the fields that I have to add it for the cdr table in the database to be able to have cdr for the call center events? 3) Generally speaking, any events will be added in the cdr (those related for calls or call center or any other thing), how can I know the fields that I have to add it to be able to store in the database? Is it the only solution is to look for the Master file to see what the cdr is logging, and based on it I have to create the database? Appreciate your kindly help. Regards Bilal Dear Tark; The asterisk version I am running is 1.8 and I can select mysql from menuselect when I am compiling. But when I googled for cdr-mysql, I discovered that I have to login for mysql and create the database and run a script to create this and give the grants. All what I found in google is related to other asterisk versions, while mine is 1.8, so the problem is how to know the required script to create the database and give the right grants to be used for CDR that suite the version I am running? From where I can get this? The following script will generate an asterisk database with a table named CDR that will work with asterisk 1.8. Be sure to change 'PASSWORD' with whatever password you want to use. SET SQL_MODE=NO_AUTO_VALUE_ON_ZERO; CREATE DATABASE `asterisk` DEFAULT CHARACTER SET latin1 COLLATE latin1_swedish_ci; USE `asterisk`; CREATE TABLE IF NOT EXISTS `cdr` ( `recid` mediumint(8) unsigned NOT NULL auto_increment COMMENT 'Record ID', `calldate` datetime NOT NULL default '-00-00 00:00:00', `clid` varchar(80) NOT NULL default '', `src` varchar(80) NOT NULL default '', `dst` varchar(80) NOT NULL default '', `dcontext` varchar(80) NOT NULL default '', `channel` varchar(80) NOT NULL default '', `dstchannel` varchar(80) NOT NULL default '', `lastapp` varchar(80) NOT NULL default '', `lastdata` varchar(80) NOT NULL default '', `duration` int(11) NOT NULL default '0', `billsec` int(11) NOT NULL default '0', `disposition` varchar(45) NOT NULL default '', `amaflags` int(11) NOT NULL default '0', `accountcode` varchar(20) NOT NULL default '', `uniqueid` varchar(32) NOT NULL default '', `userfield` varchar(255) NOT NULL default '', PRIMARY KEY (`recid`), KEY `calldate` (`calldate`), KEY `dst` (`dst`), KEY `accountcode` (`accountcode`), KEY `src` (`src`), KEY `disposition` (`disposition`), KEY `uniqueid` (`uniqueid`) ) ENGINE=InnoDB DEFAULT CHARSET=latin1 AUTO_INCREMENT=1 ; CREATE USER 'asterisk'@'localhost' IDENTIFIED BY 'PASSWORD'; GRANT FILE ON * . * TO 'asterisk'@'localhost' IDENTIFIED BY 'PASSWORD' WITH MAX_QUERIES_PER_HOUR 0 MAX_CONNECTIONS_PER_HOUR 0 MAX_UPDATES_PER_HOUR 0 MAX_USER_CONNECTIONS 0 ; GRANT INSERT ON `asterisk`.`cdr` TO 'asterisk'@'localhost'; If you're going to be running the mysql database on the same server as the asterisk box, the following cdr_mysql.conf should also work for 1.8: [global] hostname=localhost dbname=asterisk table=cdr password=PASSWORD user=asterisk port=3306 sock=/var/lib/mysql/mysql.sock userfield=1 loguniqueid=yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behavior over Zap chennels
Hi, I was wondering if anyone you got asterisk to work with Fax via google voice ? If so, can you please send me extension.conf and sip.conf, jabber.conf and gtalk.conf settings used. I would prefer faxing with Fax for Asterisk (FFA) via .call file. I see post where people got it work with Google Voice. Guide is greatly appreciated. -Charles Fax Free http://sendfreefax.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
