This article will go over how to integrate Queuemetrics with ViciDial
ViciDial integration
ViciDial is an enterprise class, open source call center suite in use by many large call centers around the world.
VICIdial has a full featured predictive dialer. It can also function as an ACD for inbound calls, or closer calls coming from VICIdial outbound frontiers. It is capable of inbound, outbound, and blended call handling.
It can also be easily integrated with QueueMetrics.
A working ViciDial instance, version 2.0.4 or later
It is very important that all servers involved (be they for QueueMetrics or ViciDial or general Asterisk usage) are on the same time zone and time, aligned with sub-second precision by an NTP daemon. If this is not so, the setting may lead to data corruption and inaccurate reports.
In order to translate ViciDial data to QueueMetrics, the following conventions are used:
The campaign_id in ViciDial is seen as the queue in QueueMetrics
The user ID in ViciDial is prepended by “agent/” and translated to the agent code in QueueMetrics (e.g. user 123 appears as agent/123)
The UniqueID for the call appears as Asterisk’s unique id prepended with server_id field (e.g. 1-1170345123.1234)
In this example, we imagine that:
The QueueMetrics server has IP 1.2.3.4
The QueueMetrics database server has IP 1.2.3.5 and the QM database is called “queuemetrics”
The ViciDial server has IP 1.2.3.6
Changes to QueueMetrics database
ViciDial and QueueMetrics work together by sharing the database.
You must log on to the QueueMetrics database and create a user for ViciDial to connect to it. We use a different username from the one QM uses so it is easy to monitor who is doing what.
GRANT ALL PRIVILEGES ON queuemetrics.* TO vicidial@'1.2.3.6' IDENTIFIED BY 'qm';
ViciDial will also need special indexing on the ‘queue_log’ table to work efficiently:
CREATE INDEX vici_time_id on queue_log(time_id);
CREATE INDEX vici_call_id on queue_log(call_id);
Changes to ViciDial
The system configuration can easily be set from the ViciDial Admin / System Settings page:
Enable QueueMetrics logging: set to 1
QueueMetrics server IP: this is the IP for the MySQL DB server, in our example “1.2.3.5”
QueueMetrics DB name: the database name, in our example “queuemetrics”
QueueMetrics DB login: the database login, in our example “vicidial”
QueueMetrics DB password: the database password, in our example “qm”
QueueMetrics URL: the login URL for QM, e.g. “http://1.2.3.4:8080/queuemetrics”
QueueMetrics LogID: leave it to VIC (this in an ID for the server)
QueueMetrics EnterQueue Prepend: This field is used to allow for prepending of one of the vicidial_list data fields in front of the phone number of the customer for customized QueueMetrics reports. Default is NONE to avoid populating anything.
A set of cron jobs is expected to run to keep the logs updated; check that they are present by issuing a ‘crontab -e’:
### fix the vicidial_agent_log once every hour and the full day run at night
33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl
50 0 * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --last-24hours
*/5 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --only-qm-live-call-check
1 1 * * * /usr/share/astguiclient/Vtiger_optimize_all_tables.pl --quiet
Also, you will need to install the PHP XML-RPC library in order to have audio data accessible from the QueueMetrics server:
pear install XML_RPC-1.5.1
Changes to QueueMetrics
Edit the ‘configuration.properties’ file in order to set the following properties:
# This is the default queue log file.
default.queue_log_file=sql:P01
By default, ViciDial logs all data to partition “P01”.
After this, you need to define each ViciDial campaign as a QueueMetrics queue, and set it properly as an inbound or outbound one. After that, you can freely create composite queues to report on all or some activity at once.
The live monitoring asks for an extension to send the call to, this is an extension dialed on the active voicemail server as defined in the system settings. If there is no active voicemail defined then the live monitor will place the call to the extension on the server that the agent is on.
As far as I know, no one uses this anymore, but I wanted to give those of you that are interested the ability to give it a try if you wanted, so here it is
How to – enable and use password encryption in ViciDial
This article is going to go over how to enable and use password encryption in ViciDial. By default passwords are displayed in plain text in the ViciDial Admin GUI, this will show you how to encrypt those.
Step 1: Installing the Bcrypt perl module
Using the CPAN console install the Bcrypt, run the below command
cpan install Crypt::Eksblowfish::Bcrypt
Step 2: Enabling the Password Encryption
By default the Password encryption is disabled in Vicidial and we need to enable it by using a perl script via the Linux command line as show below:
Now navigate to ADMIN > SYSTEM SETTINGS > PASSWORD ENCRYPTION and you’ll see the Password Encryption is now enabled. Now all new users added to the system will automatically be encrypted.
Step 3: Encrypting Plain Text Password
All the users passwords which are created before enabling the Password encryption, will remain as clear plain text ,to encrypt the existing plain text passwords either manually edit them and update or run the below command to convert all the plain text to encrypted text.
/usr/share/astguiclient/ADMIN_bcrypt_convert.pl –clear-plaintext-pass or /usr/share/astguiclient/ADMIN_bcrypt_convert.pl –debugX –update-override –clear-plaintext-pass
How to Reset the Forgotten Password
If you have forget the admin password , you need to update the password under mysql/mariadb with the hashed password, for non-admin users either you can update the password from admin login or follow the below procedure.
Step 1: Generate the Hash Password
Run the below command from SSH console with the password which you want to set of a user for example for admin user 6666 i need to set a password as admin123
/srv/www/htdocs/agc/bp.pl –pass=admin123
The above command will output the HASHED value of admin123 ,copy that proceed to step 2
Step 2: Updating the Mysql user table
once hash password generated run the below mysql command with the password generated in step 1
mysql -p use asterisk; UPDATE vicidial_users set pass_hash=’kfYvywV959fn09rSZML70wHjjxsaYjm’ where user=’6666′;
Now you can login to the vicidial admin or agent portal with the new password.
I hope this helps some of you who need to have tighter security for your systems.
How to – Remove the water drop/bloop sound from ViciDial
Ok, so as most of you probably know already, Asterisk is not going to be using meetMe anymore and they have set it to “End of life” so it will no longer get any updates. So what does this mean for Vicidial which uses MeetMe channels for all its sounds and communications? Well, there is another module called confbridge which can also handle these jobs and actually can do so better and has room for some more advanced features over time I’m sure. Ok so let’s start with the main issue a lot of people are going to have, really because they are scared more than anything, is confbridge needs at least Asterisk 16 to work so our first step is going to be to update Asterisk from 13 to 16. The good news is once you do this, you can remove the water drop sound from the customer side of the calls you make so they no longer know you are calling from a call center and hangup before you even get a chance to say hi. Before getting to step 1, make sure you are at the latest SVN which at the time of this article is 3636 by following this article.
Step 1 – Upgrade to Asterisk 16
ok so lets go ahead and install Asterisk 16 by following the steps from the article below:
We had to fix the Asterisk 16 install over Asterisk 13 by running “make uninstall” and then reinstalling as shown here:
Fixing Asterisk 16 install
Now we need to stop the current Asterisk 13 from running and start back up the new Asterisk 16.
asterisk -r
core restart now
/usr/share/astguiclient/start_asterisk_boot.pl
asterisk -r (make sure the version you see is 16)
Step 2 – Add new confbridge extensions
We need to edit extensions.conf and add some new conferences for confbridge to use so lets start by going into the asterisk directory:
cd /etc/asterisk nano extensions.conf paste the following under the [default] section:
; --------------------------
; ConfBridge Extensions
; --------------------------
; use to send a agent channel into the conference
exten => _9600XXX,1,Answer()
exten => _9600XXX,n,Playback(sip-silence)
exten => _9600XXX,n,ConfBridge(${EXTEN},vici_agent_bridge,vici_customer_user)
exten => _9600XXX,n,Hangup()
;; used to send an customer channel into the conference
exten => _29600XXX,1,Answer()
exten => _29600XXX,n,Playback(sip-silence)
exten => _29600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_agent_user)
exten => _29600XXX,n,Hangup()
;; used by an admin to enter the confernece
exten => _39600XXX,1,Answer()
exten => _39600XXX,n,Playback(sip-silence)
exten => _39600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_admin_user)
exten => _39600XXX,n,Hangup()
;; used to monitor a conference
exten => _49600XXX,1,Answer()
exten => _49600XXX,n,Playback(sip-silence)
exten => _49600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_monitor_user)
exten => _49600XXX,n,Hangup()
;; used to record into a conference
exten => _59600XXX,1,Answer()
exten => _59600XXX,n,Playback(sip-silence)
exten => _59600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_recording_user)
exten => _59600XXX,n,Hangup()
;; used to barge a conference
exten => _69600XXX,1,Answer()
exten => _69600XXX,n,Playback(sip-silence)
exten => _69600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_barge_user)
exten => _69600XXX,n,Hangup()
;; used to trigger DTMF tones in a conference
exten => _79600XXX,1,Answer()
exten => _79600XXX,n,Playback(sip-silence)
exten => _79600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_dtmf_user)
exten => _79600XXX,n,Hangup()
;; used to play an audio file to a conference
exten => _89600XXX,1,Answer()
exten => _89600XXX,n,Playback(sip-silence)
exten => _89600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_audio_user)
exten => _89600XXX,n,Hangup()
;; used to kick all channels from a conference
exten => _99600XXX,1,ConfKick(${EXTEN:1},all)
exten => _99600XXX,2,Hangup()
exten => _55559600XXX,1,ConfKick(${EXTEN:4},all)
exten => _55559600XXX,2,Hangup()
Save and exit
Step 3 – Add additional code for confbridge to work correctly
We have to edit a couple files so first lets do:
nano /etc/asterisk/confbridge.conf and paste this at the bottom: #include confbridge-vicidial.conf
Now create a new file called confbridge-vicidial.conf and add the following lines:
; Bridge Profile for agent conferences
[vici_agent_bridge]
type=bridge
max_members=10
record_conference=no
internal_sample_rate=8000
mixing_interval=40
video_mode=none
sound_join=enter
sound_leave=leave
sound_has_joined=sip-silence
sound_has_left=sip-silence
sound_kicked=sip-silence
sound_muted=sip-silence
sound_unmuted=sip-silence
sound_only_person=confbridge-only-participant
sound_only_one=sip-silence
sound_there_are=sip-silence
sound_other_in_party=sip-silence
sound_begin=sip-silence
sound_wait_for_leader=sip-silence
sound_leader_has_left=sip-silence
sound_get_pin=sip-silence
sound_invalid_pin=sip-silence
sound_locked=sip-silence
sound_locked_now=sip-silence
sound_unlocked_now=sip-silence
sound_error_menu=sip-silence
sound_participants_muted=sip-silence
; User Profile for agent channels
[vici_agent_user]
type=user
admin=no
quiet=no
startmuted=no
marked=yes
dtmf_passthrough=yes
hear_own_join_sound=yes
dsp_drop_silence=yes
; User Profile for admin channels
[vici_admin_user]
type=user
admin=yes
quiet=no
startmuted=no
marked=yes
dtmf_passthrough=yes
dsp_drop_silence=yes
; User Profile for monitoring
[vici_monitor_user]
type=user
admin=no
quiet=no
startmuted=yes
marked=no
dtmf_passthrough=no
dsp_drop_silence=yes
; User Profile for barging
[vici_barge_user]
type=user
admin=no
quiet=no
startmuted=no
marked=no
dtmf_passthrough=yes
dsp_drop_silence=yes
; User Profile for customers channels
[vici_customer_user]
type=user
admin=no
quiet=no
startmuted=no
marked=yes
dtmf_passthrough=yes
hear_own_join_sound=no
dsp_drop_silence=yes
; User Profile for call recording channels
[vici_recording_user]
type=user
admin=no
quiet=yes
startmuted=yes
marked=no
dtmf_passthrough=no
dsp_drop_silence=yes
; User Profile for audio playback channels
[vici_audio_user]
type=user
admin=no
quiet=yes
startmuted=yes
marked=no
dtmf_passthrough=no
dsp_drop_silence=yes
; User Profile for triggering DTMF
[vici_dtmf_user]
type=user
admin=no
quiet=yes
startmuted=yes
marked=no
dtmf_passthrough=yes
dsp_drop_silence=yes
Step 4 – Add ConfBridge Conferences to Database
Go into mysql and add the conferences to the vicidial_confbridges table by pasting the following commands:
(Click Enter for Y) Next enter your server IP and press enter twice to chaneg it in the DB as show below:
Update Server IP
Step 5 – Code changes to ViciDial files
There are some files now that have to be patched in order to include the changes needed for confbrides to work. They are in the “extras/ConfBridge/” directory of the svn/trunk codebase. Lets copy the files over to where they are needed, this will depend on if you are using a single server or a cluster to where the files go. Here is a list of where they go:
Dialers:
/usr/share/astguiclient/ -
- ADMIN_keepalive_ALL.pl.diff
- ADMIN_update_server_ip.pl.diff
- AST_DB_optimize.pl.diff
- AST_reset_mysql_vars.pl.diff
- AST_VDremote_agents.pl.diff
- AST_conf_update_screen.pl
Webservers:
/srv/www/htdocs/agc/ -
- vicidial.php.diff
- vdc_db_query.php.diff
- manager_send.php.diff
/srv/www/htdocs/vicidial/ -
- non_agent_api.php.diff
You can copy and paste the entire code below to get it all done
Step 6 – Add the confbridge keepalive and turn off the conf_update keepalive in crontab
A new screen session has been added called 'AST_conf_update_screen.pl'. This screen session replaces both the AST_conf_update.pl and AST_conf_update_3way.pl scripts. It checks both MeetMe and ConfBridge conferences for unnecessary channels and removes them. For instance if an agent does a Leave 3way and a few minutes later one of the remaining channels hangs up. This script will remove the remaining channel and free the conference for use by Vicidial. This screen session is optional for use with MeetMe but is required to be used with ConfBridge.
Add C to VARactive_keepalives
To enable this screen session you need to add a 'C' to the 'VARactive_keepalives' variable in the '/etc/astguiclient.conf' on your dialers, and comment out the following line from your crontab:
### updater for conference validator
#* * * * * /usr/share/astguiclient/AST_conf_update.pl
You can configure the refresh interval for how often the screen session checks for unnecessary channels by changing "Conf Update Interval" in Admin -> Servers.
Step 7- Make the needed changes in ViciDial GUI
Login to the ViciDial Admin GUI and go to Admin > Servers and click on the server(s) that are dialers and change the conferencing engine to “CONFBRIDGE” as shown below:
Change conferencing Engine to CONFBRIDGE
Step 8 – Add confbridge to manager.conf
nano /etc/asterisk/manager.conf
Go to the bottom and paste:
[confcron]
secret = 1234
read = command,reporting
write = command,reporting
eventfilter=Event: Meetme
eventfilter=Event: Confbridge
Thats a lot to get through but now that its done your customers will no longer hear the “Water drop” or “bloop” sound that they do now and you’re ready for the future of Asterisk now that meetme has been deprecated. Thats it for now, you’re all done. Hopefully this helps those of you who are a little intimidated to try the change and like always, if you have any problems, feel free to visit our Skype live support with almost 200 Vicidial experts from around the world.
How to – Switch ViciDial from meetme to confbridge
Ok, so as most of you probably know already, Asterisk is not going to be using meetMe anymore and they have set it to “End of life” so it will no longer get any updates. So what does this mean for Vicidial which uses MeetMe channels for all its sounds and communications? Well, there is another module called confbridge which can also handle these jobs and actually can do so better and has room for some more advanced features over time I’m sure. Ok so let’s start with the main issue a lot of people are going to have, really because they are scared more than anything, is confbridge needs at least Asterisk 16 to work so our first step is going to be to update Asterisk from 13 to 16. Make sure you are at the latest SVN which at the time of this article is 3636 by following this article.
Step 1 – Upgrade to Asterisk 16
ok so lets go ahead and install Asterisk 16 by following the steps below:
First you will need to follow this article on how to download, patch and compile Asterisk 16 or you can use the compile commands below:
./configure --libdir=/usr/lib64 --with-pjproject-bundled --with-jansson-bundled
rm -rf menuselect.makeopts
make menuselect*
**** Go down one to applications and hit enter, go down until you find the "meetme" app and press space bar, then hit "X" ****
make && make install
make uninstall
make install
We had to fix the Asterisk 16 install over Asterisk 13 by running “make uninstall” and then reinstalling as shown here:
Fixing Asterisk 16 install
Now we need to stop the current Asterisk 13 from running and start back up the new Asterisk 16.
asterisk -r
core restart now
/usr/share/astguiclient/start_asterisk_boot.pl
asterisk -r (make sure the version you see is 16)
Step 2 – Add new confbridge extensions
We need to edit extensions.conf and add some new conferences for confbridge to use so lets start by going into the asterisk directory:
cd /etc/asterisk nano extensions.conf paste the following at the bottom:
; --------------------------
; ConfBridge Extensions
; --------------------------
; use to send a agent channel into the conference
exten => _9600XXX,1,Answer()
exten => _9600XXX,n,Playback(sip-silence)
exten => _9600XXX,n,ConfBridge(${EXTEN},vici_agent_bridge,vici_customer_user)
exten => _9600XXX,n,Hangup()
;; used to send an customer channel into the conference
exten => _29600XXX,1,Answer()
exten => _29600XXX,n,Playback(sip-silence)
exten => _29600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_agent_user)
exten => _29600XXX,n,Hangup()
;; used by an admin to enter the confernece
exten => _39600XXX,1,Answer()
exten => _39600XXX,n,Playback(sip-silence)
exten => _39600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_admin_user)
exten => _39600XXX,n,Hangup()
;; used to monitor a conference
exten => _49600XXX,1,Answer()
exten => _49600XXX,n,Playback(sip-silence)
exten => _49600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_monitor_user)
exten => _49600XXX,n,Hangup()
;; used to record into a conference
exten => _59600XXX,1,Answer()
exten => _59600XXX,n,Playback(sip-silence)
exten => _59600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_recording_user)
exten => _59600XXX,n,Hangup()
;; used to barge a conference
exten => _69600XXX,1,Answer()
exten => _69600XXX,n,Playback(sip-silence)
exten => _69600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_barge_user)
exten => _69600XXX,n,Hangup()
;; used to trigger DTMF tones in a conference
exten => _79600XXX,1,Answer()
exten => _79600XXX,n,Playback(sip-silence)
exten => _79600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_dtmf_user)
exten => _79600XXX,n,Hangup()
;; used to play an audio file to a conference
exten => _89600XXX,1,Answer()
exten => _89600XXX,n,Playback(sip-silence)
exten => _89600XXX,n,ConfBridge(${EXTEN:1},vici_agent_bridge,vici_audio_user)
exten => _89600XXX,n,Hangup()
;; used to kick all channels from a conference
exten => _99600XXX,1,ConfKick(${EXTEN:1},all)
exten => _99600XXX,2,Hangup()
exten => _55559600XXX,1,ConfKick(${EXTEN:4},all)
exten => _55559600XXX,2,Hangup()
Save and exit
Step 3 – Add additional code for confbridge to work correctly
We have to edit a couple files so first lets do:
nano /etc/asterisk/confbridge.conf and paste this at the bottom: #include confbridge-vicidial.conf
Now create a new file called confbridge-vicidial.conf and add the following lines:
; Bridge Profile for agent conferences
[vici_agent_bridge]
type=bridge
max_members=10
record_conference=no
internal_sample_rate=8000
mixing_interval=40
video_mode=none
sound_join=enter
sound_leave=leave
sound_has_joined=sip-silence
sound_has_left=sip-silence
sound_kicked=sip-silence
sound_muted=sip-silence
sound_unmuted=sip-silence
sound_only_person=confbridge-only-participant
sound_only_one=sip-silence
sound_there_are=sip-silence
sound_other_in_party=sip-silence
sound_begin=sip-silence
sound_wait_for_leader=sip-silence
sound_leader_has_left=sip-silence
sound_get_pin=sip-silence
sound_invalid_pin=sip-silence
sound_locked=sip-silence
sound_locked_now=sip-silence
sound_unlocked_now=sip-silence
sound_error_menu=sip-silence
sound_participants_muted=sip-silence
; User Profile for agent channels
[vici_agent_user]
type=user
admin=no
quiet=no
startmuted=no
marked=yes
dtmf_passthrough=yes
hear_own_join_sound=yes
dsp_drop_silence=yes
; User Profile for admin channels
[vici_admin_user]
type=user
admin=yes
quiet=no
startmuted=no
marked=yes
dtmf_passthrough=yes
dsp_drop_silence=yes
; User Profile for monitoring
[vici_monitor_user]
type=user
admin=no
quiet=no
startmuted=yes
marked=no
dtmf_passthrough=no
dsp_drop_silence=yes
; User Profile for barging
[vici_barge_user]
type=user
admin=no
quiet=no
startmuted=no
marked=no
dtmf_passthrough=yes
dsp_drop_silence=yes
; User Profile for customers channels
[vici_customer_user]
type=user
admin=no
quiet=no
startmuted=no
marked=yes
dtmf_passthrough=yes
hear_own_join_sound=no
dsp_drop_silence=yes
; User Profile for call recording channels
[vici_recording_user]
type=user
admin=no
quiet=yes
startmuted=yes
marked=no
dtmf_passthrough=no
dsp_drop_silence=yes
; User Profile for audio playback channels
[vici_audio_user]
type=user
admin=no
quiet=yes
startmuted=yes
marked=no
dtmf_passthrough=no
dsp_drop_silence=yes
; User Profile for triggering DTMF
[vici_dtmf_user]
type=user
admin=no
quiet=yes
startmuted=yes
marked=no
dtmf_passthrough=yes
dsp_drop_silence=yes
Step 4 – Add ConfBridge Conferences to Database
Go into mysql and add the conferences to the vicidial_confbridges table by pasting the following commands:
Update the IP to your server IP by running the following command:
/usr/share/astguiclient/ADMIN_update_server_ip.pl –old-server_ip=10.10.10.17 (Click Enter for Y) Next enter your server IP and press enter twice to chaneg it in the DB as show below:
Update Server IP
Step 5 – Code changes to ViciDial files
There are some files now that have to be patched in order to include the changes needed for confbrides to work. They are in the “extras/ConfBridge/” directory of the svn/trunk codebase. Lets copy the files over to where they are needed, this will depend on if you are using a single server or a cluster to where the files go. Here is a list of where they go:
Dialers:
/usr/share/astguiclient/ -
- ADMIN_keepalive_ALL.pl.diff
- ADMIN_update_server_ip.pl.diff
- AST_DB_optimize.pl.diff
- AST_reset_mysql_vars.pl.diff
- AST_VDremote_agents.pl.diff
- AST_conf_update_screen.pl
Webservers:
/srv/www/htdocs/agc/ -
- vicidial.php.diff
- vdc_db_query.php.diff
- manager_send.php.diff
/srv/www/htdocs/vicidial/ -
- non_agent_api.php.diff
You can copy and paste the entire code below to get it all done
Step 6 – Add the confbridge keepalive and turn off the conf_update keepalive in crontab
A new screen session has been added called 'AST_conf_update_screen.pl'. This screen session replaces both the AST_conf_update.pl and AST_conf_update_3way.pl scripts. It checks both MeetMe and ConfBridge conferences for unnecessary channels and removes them. For instance if an agent does a Leave 3way and a few minutes later one of the remaining channels hangs up. This script will remove the remaining channel and free the conference for use by Vicidial. This screen session is optional for use with MeetMe but is required to be used with ConfBridge.
Add C to VARactive_keepalives
To enable this screen session you need to add a 'C' to the 'VARactive_keepalives' variable in the '/etc/astguiclient.conf' on your dialers, and comment out the following line from your crontab:
### updater for conference validator
#* * * * * /usr/share/astguiclient/AST_conf_update.pl
You can configure the refresh interval for how often the screen session checks for unnecessary channels by changing "Conf Update Interval" in Admin -> Servers.
Step 7- Make the needed changes in ViciDial GUI
Login to the ViciDial Admin GUI and go to Admin > Servers and click on the server(s) that are dialers and change the conferencing engine to “CONFBRIDGE” as shown below:
Change conferencing Engine to CONFBRIDGE
Thats a lot to get through but now that its done your customers will no longer hear the “Water drop” or “bloop” sound that they do now and you’re ready for the future of Asterisk now that meetme has been deprecated. Thats it for now, you’re all done. Hopefully this helps those of you who are a little intimidated to try the change and like always, if you have any problems, feel free to visit our Skype live support with almost 200 Vicidial experts from around the world.
This short post is just going to briefly go over the more common commands you need in order to manage firewalld and make sure you keep your servers safe. This firewall works in conjunction with the dynamic portal for ViciDial as well as the built in whitelist initially named ViciWhite in the IP list area in Admin.
Dynamic portal
Systemctl and Firewalld
Enable firewalld
This makes sure that firewalld will be started automatically with the server.
systemctl enable firewalld
Start firewalld
After the firewalld service is enabled, you’ll need to start it manually the first time. This is how you would manually start firewalld if it were not already running.
systemctl start firewalld
Stop firewalld
When troubleshooting rules and connection issues, you may need to stop the fireawlld service momentarily. You can stop the service with the following command.
systemctl stop firewalld
Restart firewalld
If for some reason, you need to restart the service, you can do that with the systemctl restart command.
systemctl restart firewalld
Firewalld status
Checking the status of the service gives us the most meaningful and informative output. Here you can see whether the service is enabled, running, failed, or anything else.
systemctl status firewalld
In this example output, you can see that the service is enabled, active, and running on the server. If it were not running or in a failed state, this would be displayed.
[root@alma ~]# systemctl status firewalld ● firewalld.service - firewalld - dynamic firewall daemon Loaded: loaded (/usr/lib/systemd/system/firewalld.service; enabled; vendor preset: enabled) Active: active (running) since Tue 2019-01-22 22:50:32 EST; 1h 0min ago Main PID: 808 (firewalld) CGroup: /system.slice/firewalld.service └─808 /usr/bin/python -Es /usr/sbin/firewalld --nofork --nopid
Managing Firewalld and Configuring Rules
Now that we have firewalld running, we can get down to set the configuration. We can open ports, allow services, whitelist IPs for access, and more. In all of these examples, we include the –permanent flag. This is important to make sure a rule is saved even after you restart firewalld, or reboot the server. Once you’re done adding new rules, you need to reload the firewall to make the new rules active.
Add a Port for TCP or UDP
You do have to specify TCP or UDP and to open a port for both. You will need to add rules for each protocol.
Using a slight variation on the above structure, you can remove a currently open port, effectively closing off that port.
firewall-cmd --permanent --remove-port=444/tcp
Add a Service
These services assume the default ports configured within the /etc/services configuration file; if you wish to use a service on a non-standard port, you will have to open the specific port, as in the example above.
You can also allow a range of IPs using what is called CIDR notation. CIDR is outside the scope of this article but is a shorthand that can be used for noting ranges of IP addresses.
As the firewall-cmd tool is mostly used for opening or allowing access, rich rules are needed to block an IP. Rich rules are similar in form to the way iptables rules are written.
Whitelist an IP Address for a Specific Port (More Rich Rules)
We have to reach back to iptables and create another rich rule; however, we are using the accept statement at the end to allow the IP access, rather than reject its access.
firewall-cmd --permanent --add-rich-rule='rule family="ipv4" source address="192.168.1.100" port protocol="tcp" port="3306" accept'
Removing a Rich Rule
To remove a rich rule, use the option —remove-rich-rule, but you have to fully specify which rule is being removed, so it is best to copy and paste the full rule, rather than try to type it all out from memory.
firewall-cmd --permanent --remove-rich-rule='rule family="ipv4" source address="192.168.1.100" port protocol="tcp" port="3306" accept'
Saving Firewall Rules
After you have completed all the additions and subtraction of rules, you need to reload the firewall rules to make them active. To do this, you again use the firewall-cmd tool but using the option –reload.
firewall-cmd --reload
Viewing Firewall Rules
After reloading the rules, you can confirm if the new rules are in place correctly with the following.
firewall-cmd --list-all
Here is an example output from the –list-all option, you can see that this server has a number of ports, and services open in the firewall along with a rich rule (that forwards one port to another).
Hopefully this will help a lot of you that end up just not using a firewall at all because it intimidates you not knowing how to use it correctly. Well, I’ve just eliminated that excuse, so now I want to see more of you securing your servers and dialer systems. Here is a few articles to get you started in the right direction.
That’s it for this article, hopefully you guys take this serious because hackers, especially ransomware thieves are targeting dialer servers in particular for their schemes, such as using the VoIP to call their victims to either trick them into downloading files or threatening them over the phone with blackmail or other means. BazarCall is one of the more well known tools thats being used by the ransomware group called Ryuk.
In this article, I am going to go over some of the more often asked questions and how to fir them. If you still don’t understand, feel free to join our live chat. Also included will be some pretty useful ways to use Linux’ built in tools for your benefit.
1) Lets start easy with “htop”. This command, “htop” will display your current drain on system resources and show you where you stand. Take a look at the picture below:
htop
2) Creating a filter to only call certain status every 90 days only
This next one can be very useful for those of you that need to drop certain dispositions from being called for a set time period, for this example, we’re gonna say 90 days for any lead marked with NI(not interested). You need to create a “filter” and paste the following mysql query which can be altered to fit your needs: “status IN (‘NI’) and modify_date < NOW() – INTERVAL 90 DAY OR status IN (‘N’,’NA’ ,’B’,’AB’,’DROP’,’PDROP’,’NEW’,’NP’,’NANQUE’,’ADCT’)”. The first status “NI” can be set to anything you want, like ‘ADCT’ to go back and see if those temporary disconnected numbers are back or whatever dispo you’d like to go back to. You can also change the 90 to however many days you want to wait before dialing those again and finally, you can change from the right side of the query where it says “status IN (‘N’,’NA’ ,’B’,’AB’,’DROP’,’PDROP’,’NEW’,’NP’,’NANQUE’,’ADCT’)” and either add to it or remove from it, etc. Here is a small example of the one used in this article:
90 day filter
3) Is your audio store not accepting new files? Does it keep telling you improper format? Not a problem, we need to run a few commands in the Linux cli to get this fixed up. Copy and paste the code below:
cd /usr/share/astguiclient/
sed -i 's/wgetbin -q/wgetbin --no-check-certificate -q/g' ADMIN_audio_store_sync.pl
/usr/share/astguiclient/ADMIN_audio_store_sync.pl --debugX
chmod -R 777 audio folder in /var/www/html/audio_folder (somehting like ndt7h4rr8fynf3y8er)
chown -R apache:apache /var/www/html/audio_folder (somehting like ndt7h4rr8fynf3y8er)
4) Need to change your recording links from http to https? Try the following query in mysql:
UPDATE recording_log
SET location = REPLACE(location, 'http://127.0.0.1/', 'https://127.0.0.1/')
WHERE location LIKE '%http://127.0.0.1/%';
you can also run this with archive_log instead of recording_log and the Ip's can be switched out with FQDN's or domain names.
5) Here is a simple way to change the passwords for all users as well as a couple variants to get more specific:
update vicidial_users
set pass="newpass"; ###update all users password
update vicidial_users
set pass="newpass"
where user between 7000 AND 7015; ##update users that are only between 7000 and 7015
update vicidial_users
set pass="newpass"
where user_level between 1 AND 8; ##change password for all users between levels 1 and 8, dont change level 9 passwords
That’s it for today, I’ll add some more tomorrow and the days to follow so stay tuned.
Thanks, Chris aka carpenox
9/5/2022 – Adding a few more
6) Speed up call handling for agent only or dial servers only. (no web/DB)
If you are using servers where agents only log in and handle calls, but no calls are placed out from it, then you can use the new delay options on those servers to enhance efficient operations. The –autodial-delay=X option in the ADMIN_keepalive_ALL.pl script will allow you to set the delay to 100 milliseconds for these agent-only servers(the default is 2500ms). Lowering the delay for agent-only servers makes the auto-dial FILL process more responsive to the changes in the agent state on the agent-only servers which will enhance how your cluster operates. You can also use the –adfill-delay=X CLI option for the ADMIN_keepalive_ALL.pl script on the server that is running the FILL process and lower it as well if you have a larger cluster. The default of that process is also 2500ms, but you can lower it down to 500ms if needed.
7) Are all your calls showing DISPO?
This is usually because your php time doesnt match the system. You can change it by editing /etc/php.ini for CentOS/Alma/Rocky or /etc/php7/php.ini for Leap. Just change it to match and you’ll be good to go.
8) How can I move a lead based on how many times its been called or how old the lead is?
The answer is yes, you can use the script named dispo_move_list.php, that you can find in your /usr/src/astguiclient/trunk/extras folder on your web server. It has a few different choices you can use that you can see below:
# Definable Fields: (other fields should be left as they are)
# - log_to_file - (0,1) if set to 1, will create a log file in the agc directory
# - sale_status - (SALE---XSALE) a triple-dash "---" delimited list of the statuses that are to be moved
# - exclude_status - (Y,N) if set to Y, will trigger for all statuses EXCEPT for those listed in sale_status, default is N
# - talk_time_trigger - (0,1,2,3,...) if set to number greater than 0, will only trigger for talk_time at or above set number, default is 0
# - called_count_trigger - (1,2,3,...) if set to number greater than 0, will only trigger for called_count at or above set number, default is 0
# - list_id_trigger - (101,...) if set to number greater than 99, will only trigger for list_id equal to the set number(NOTE: list_id must be sent), default is disabled
# - list_id - (101,...) if you want to use list_id_trigger then this must be set: "list_id=--A--list_id--B--", default is disabled
# - lead_age - (1,2,3,...) if set to number greater than 0, will only trigger for a lead entry_date this number of days old or older, default is 0
# - new_list_id - (999,etc...) the list_id that you want the matching status leads to be moved to
# - reset_dialed - (Y,N) if set to Y, will reset the called_since_last_reset flag on the lead
# - populate_sp_old_list - (Y,N) if set to Y, will populate the security_phrase field of the lead with the old list_id
# - populate_comm_old_date - (Y,N) if set to Y, will populate the comments field of the lead with the date and time when the lead was last called
# Multiple sets of statuses:
# - sale_status_1, new_list_id_1, reset_dialed_1, exclude_status_1, called_count_trigger_1 - adding an underscore and number(1-99) will allow for another set of statuses to check for and what to do with them
# - multi_trigger - (talk-age...) if set to 1 or more of "talk,age,list,count,status"(separated by '-') it will check for only one of included triggers to be met for the lead to be moved, (does not work with multiple sets)
9) Can I record my agents outside of ViciDial or once calls are transferred outside the system?
Yes, you can. You have to use an agi script called agi-NVA_recording.agi which was made for this purpose. Here are some triggers for it below:
# ; 1. logging output (NONE|STDERR|FILE|BOTH)
# ; 2. the ViciDial user ID, if empty it defaults to accountcode(usually phone extension) or vicidial_live_agents user who launched the call
# ; 3. log this call in user_call_log (Y|N) default N
# ; 4. log this call in call_log (Y|N) default N, ONLY NEEDED FOR INBOUND AND INTERSYSTEM CALLS!!!
# ; 5. audio record this call (Y|N) default N
# ; 6. double-log this call in call_log (Y|N) default N, ONLY NEEDED FOR INBOUND CALLMENU FORWARDED CALLS!!!
# ; 7. play the recording ID of this call before recording starts
# ; 8. include the recording ID in the filename
# ; 9. search vicidial_list for phone number dialed (Y|N) default N, assumes 10 digit phone numbers
# ; 10. if 9 is Y, this is search method (ALLLISTS|PHONE) default ALLLISTS, search all lists, use phone setting, CURRENTLY DOES NOTHING
# ; 11. error out and end call if phone number is not found (Y|N) default N
# ; 12. run the phone entry's NVA Call URL (Y|N) default N
# ; 13. if 9 is Y, and phone number is not found, insert into phone's NVA List ID (Y|N) default N
# ; 14. if 13 is Y, override phone's NVA List ID with this list ID when lead is inserted
# ; 15. if 13 is Y, override phone's NVA Phone Code with this phone code when lead is inserted
# ; 16. if 13 is Y, override phone's NVA Status with this status when lead is inserted
Here as an example to give you an idea how it looks:
# ;custom dialplan entry example: (similar to the defaultlog Call Menu)
#exten => _X.,1,AGI(agi-NVA_recording.agi,BOTH------Y---N---Y---N---N---N)
#exten => _X.,n,Goto(default,${EXTEN},1)
#exten => _X.,n,Hangup
10) Why don’t calls to Canada show the caller ID? (They display those weird V435345874353457353 numbers)
It’s because Canadian telco’s accept any CID that is sent and Vicidial sends this UID first which is ignored by American telco’s and it was a hard solution for me to find, so here it is for you guys. It uses a dialplan that was given to me by The dialplan God – Striker24/7.
In this article I am going to over how to create an inbound call menu or IVR (Interactive Voice Response) for ViciDial which will help reduce your drop percent to help you legally within the 3% drop rate for the USA or 5% drop rate for Canada, by way of the FCC guidelines.
Step 1 – Create the call menu
Login to the admin GUI for ViciDial and click “inbound” then “add a new call menu” as shown below., then name your call menu. (For this example, I named it IVR)
Add a new call menu
Step 2 – Setup your basic IVR options
Now we are going to setup the top portion on the call menu/IVR. Assuming you have already created your audio recording for your IVR and added it to the audio store, select that file at the “Menu Prompt” option. You can also fill out your timeout prompt, invalid prompt and other options as shown below:
IVR Options
Step 3 – Create your IVR options
Now we will select a few common IVR options to add to your call menu. For this example I have given options to 1) leave a voicemail message, 2) be transferred to a live agent, 3) Be added to the Do Not call list or #) to play the options again. See below:
IVR Selections
Step 4 – Route unanswered Inbound calls to IVR
The final step is to add this IVR/Call menu to your inbound calls when your agents aren’t available. For this you need to go to your inbound ingroup and change your “call time” and no agent queueing/after hours options to reflect the picture below:
Set the call menu for the inbound calls
This articles assumes you know how to already setup the other parts needed for this such as creating audio recordings, uploading them to the audio store and setting up inbound groups. If you have any questions feel free to visit our skype live chat. Hopefully this helps some of you that have been asking me about this.
How to – Setup Email for ViciDial reports and voicemail
This article will go over how to setup your email configuration on your server in order to get reports from ViciDial or to receive voicemail to email notifications and recordings. Gmail recently changed the way their security is handled so some changes need to be made in order for your emails to go through. You can alter the config below for any email service you may use.
Step 1 – Edit your postfix config file
Ok so first you need to edit your postfix configuration by editing main.cf and adding a few lines to the bottom of it.
nano /etc/postfix/main.cf
### Now paste the following lines to the bottom: ###
relayhost = [smtp.gmail.com]:587
smtp_sasl_auth_enable = yes
smtp_sasl_password_maps = lmdb:/etc/postfix/sasl_passwd
smtp_sasl_security_options = noanonymous
smtp_sasl_mechanism_filter = plain
smtp_use_tls = yes
Step 2 – Create a Gmail APP Password
Go to Gmail and log in to your account.
Now, in the upper-right corner, click the Home Account icon. In the open wizard, click “Manage my Accounts.”
You will be redirected to Gmail Settings. In the left tab, click Security, and then click ‘2-step verification’ in the ‘Signing into Google’ section.
The 2-step verification window will appear on the screen, click ‘Get started.’
In the next step, Google will ask you to use your phone as your second sign-in step. Click ‘Try it Now’ to move ahead. You will be asked to log in to Gmail again, and a notification will appear on your phone screen to confirm the sign-in, click Yes.
After giving access from your phone, Google will ask you to add a backup option, enter your phone number, and click Send.
A code will be sent to your phone to confirm the login, enter the code in the opened window and click ‘Next.’
2-step verification has been enabled. Now, you can easily create an App Password to login to third-party apps.
Creating App Passwords
To create the App Password after enabling 2-step verification, go to ‘Manage your Account’ from the home tab.
Now, in managing accounts, click the Security tab in the left pane, and then scroll down to ‘Signing to Google.’
Here, you’ll see the App Passwords option, click over it. You will be asked to sign in again to your Gmail account.
After signing in, select the type of app and device for which you want to generate an app password and click Generate.
The App Password will be created; you can copy it to the clipboard, and click ‘Done.’
That’s how you can create App Password for your Gmail account. Now, you can use this Password to login to Gmail with your ViciDial server.
Step 3 – Edit your sasl_password file
Now you need to edit your sasl_password file as shown below:
nano /etc/postfix/sasl_passwd
### Alter the line below with your username and password ###
[smtp.gmail.com]:587 youremail@gmail.com:password
That’s all there is to it, now set your email account on your voicemail boxes or automated reports in ViciDial and you should be good to go.
This article is going to go over the method to best setup CID groups and the different ways to configure each option available for them including statefill, statelookup and creating auto rotators. This will assume you know how to add DID’s to the system which is fairly easy. So the first thing you need to do is create the CID group as defined below:
Step 1 – Create CID group
Within the ViciDial admin gui go to the Admin section then to CID groups
CID Groups
Once you are here, just click “Add a CID Group”:
Add a CID group
The first one we will create is for state lookup method, define “AREACODE” as the group type like shown below:
CID group – State lookup
hit submit and you’ll see it looking like the below picture
CID Group added
Step 2 – Add the DID’s to the CID Group
Next, we will be using the admin utilities to add your DIDs to the CID Group. Go to the “Reports” page and scroll down to the bottom to click on “Admin utilities” then click on “Admin bulk Tools”
Click Admin Utilities
click Admin Bulk Tools
Now scroll down to “CID Groups and AC-CID Bulk Add” and add your DID’s, selecting “state lookup” will assign each number with its area code to the proper state its from, selecting “statefill” will automatically add every area code for each state you have a DID for with separate entries for each area code which is a really nice option to ensure local presence in some fashion even if you don’t own DID’s from every area code.
Add DID to the CID group
When you choose statefill youll see the same number submitted many times, this is normal
StateFill Option
After doing this you’ll notice it filled it every area code for this example I did it for Florida, and even though I only entered 5 area codes, it filled in these 5 numbers for every area code in Florida:
Area Code Statefill
Step 3 – Assign CID group to your campaign
Now you just have to assign the newly created CID group to the detail view of your campaign as shown below, make sure you set “Custom CallerID:” to Y
Assinging CID Group to campaign
Now I will show you how to create an auto rotating CID group which will rotate all the DID’s added to this group based on intervals you can choose. Complete step one again but choose “None” for the “CID Group Type” as shown below”
Create CID Group for Auto Rotator
Once your hit submit, choose how often you want the DID’s to change, a good starting point is every 5 minutes or 35 calls as shown below:
Set intervals for auto rotator
Go back and complete step 2 again using the “state lookup” method and assign the CID Group to the campaign, now your DID’s will auto rotate, I hope you have enjoyed this tutorial and like always, if you have any questions, feel free to join our Live Support on Skype.
How to – Add conferences for add on servers to a cluster
This article will show you how to add additional servers to your cluster and make sure all the conferences are also added. Its just a few simple steps needed to have it done for you.
Step 1 – Adding the second server(or 3rd, 4th, 5th, whatever)
There is a SQL script already created that does all the hard work for you, just follow the commands below:
mysql -A asterisk
\. /usr/src/astguiclient/trunk/extras/second_server_install.sql
This should be the output you see if done correctly
Step 2 – Updating the IP from 10.10.10.16 to your new server IP
For this part, there is a perl script that will update the conferences and vicidial conferences as well as the new server that was added to your GUI as “TESTast”. Just copy and paste the following line into your Linux CLI:
Then make sure you add your IP to the third question it asks you as shown below:
Changing the server IP
Step 3 – Run the install.pl script
Now we need to run the install.pl script to connect the second server to DB server
cd /usr/src/astguiclient/trunk
perl install.pl
Make sure when you get to the DB server question you input your DB server IP(where the blue arrow is)
Change DB server IP
Step 4 – Change the name of the server in the ViciDial GUI
Make sure to change the name of your server and update the Asterisk version and trunks
Change TESTast to your new server name
Change your asterisk version and trunks
Thats all there is to it, you should now have your second server added to your cluster. IF you have any questions feel free to comment here or stop by our Skype Live support
