vos3000 illegal calls query to block IP address
All call records are saved in e_cdr_files
you can query to get ip addresses of illegal calls source.
mysql> select DISTINCT callerip from e_cdr_20180818 where endreason = ‘-9’;
| callerip |
| 18.104.22.168 |
| 22.214.171.124 |
| 126.96.36.199 |
| 188.8.131.52 |
| 184.108.40.206 |
| 220.127.116.11 |
| 18.104.22.168 |
| 22.214.171.124 |
| 126.96.36.199 |
| 188.8.131.52 |
10 rows in set (0.00 sec)
if someone can write a script to get IP address from current date CDR for every 5 min and update the iptables automatically.
How to install vos3000 184.108.40.206 on CentOS 6.X
Follow the steps
#chkconfig iptables off
#chkconfig ip6tables off
Continue reading How to install vos3000 220.127.116.11 on CentOS 6.X
Protect your Asterisk PBX server from Black listed IP address
VoIPBL is a distributed VoIP blacklist that is aimed to protects against VoIP Fraud and minimizing abuse for network that have publicly accessible PBX’s.
For more details http://www.voipbl.org/
For Asterisk PBX you need to install Fail2ban. This is the only required dependency needed to run VoIP Blacklist on your server.
Continue reading Protect your Asterisk PBX server from Black listed IP address
NAT information saved in Elastix by FreePBX
Continue reading NAT information saved in Elastix by FreePBX
Extension information saved in Elastix
[root@pbx asterisk]# cat /etc/asterisk/sip_additional.conf
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
Elastix IVR Recording PCM Encoded, 16 Bits, at 8000Hz
You can record your IVR prompts by two ways one is by your Telephone Extension and another is with Sound Recorder software.
1- Go to Elastix PBX setting > Systems Recordings
Give your Extension no an press Go.
Now go to your Extension and press *99 and record .
Continue reading Elastix IVR Recording PCM Encoded, 16 Bits, at 8000Hz
How to configure Cisco Phone 7911 with Elastix
Cisco Phone 7911 looks for term06.default.loads file to upgrade in to SIP firmware.
Download the SIP firmware from cisco or google it and get the firmware files
Configure your TFTP server and upgrade SIP firmware.
For SIP configuration this phone support XML configuration file . SEPMACADD.cnf.xml
Continue reading How to configure Cisco Phone 7911 with Elastix
How to configure Cisco Phone 7960 or 7940 with Elastix
Cisco Phone 7960 or 7940 comes default with SCCP firmware you need to change the firmware to SIP .
After the SIP firmware update you need to configure these phone to work with Elastix .
Cisco Phone 7960 or 7940 does not support XML configuration file it support cnf configuration file.
If my Phone MAC address is 0014A9713F98 then my configuration file name should be SIP0014A9713F98.cnf and place it in to Elastix server /tftpboot/ folder
I am giving simple example to configure one Extension . You can change according to your requirement.
Continue reading How to configure Cisco Phone 7960 or 7940 with Elastix
How to Configure Cisco Phone 7912G with Elastix
Cisco Phone 7912G comes with SCCP firmware and you have to convert it to SIP .
The easy way to convert this phone in to SIP .
1- Download the SIP firmware from Cisco or google it and get it.
SIP Firmware Version: 8.0001
2- Extract the file and upload to Elastix server /tftpboot/ folder
CP7912080001SIP060412A.sbin and gkdefault.cfg
Continue reading How to Configure Cisco Phone 7912G with Elastix
How to troubleshoot CallerID issues on a FXO
The Digium® X100M FXO module and X400M FXO module are daughter cards that allow Digium® analog cards to terminate analog telephone lines (POTS).
This document is intended to be a brief description of toublehsooting steps that Digium’s customers can take to resolve Caller ID issues on their Asterisk Server.
At least one analog line.
A regular analog phone with Caller ID display feature
Fully configured Asterisk server with the latest version of DAHDI and Asterisk
A Caller ID issue could be caused by several factors. This section explores the possibility that the issue could be located at the telco or by noise on the lines.
Connect an analog phone to the demarcation point. In telephony, the demarcation point is the point at which the telephone company network ends and connects with the wiring at the customer premises.
Place a call — using another line or mobile phone — from the PSTN to the analog phone connected at the demarcation point and check if the phone displays Caller ID. If the Caller ID information is not being shown by the phone, please contact your telco and ask them how to enable this feature.
Otherwise, please disconnect the phone and plug it to the line that is directly connected or closest to your Asterisk server and repeat the test call.
Continue reading How to troubleshoot CallerID issues on a FXO