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Message-ID: <1aa82a4137514f65d417a47a57483bc0@teo-en-ming.com>
Date: Thu, 24 Dec 2020 16:42:32 +0800
From: Turritopsis Dohrnii Teo En Ming <ceo@...-en-ming.com>
To: linux-kernel@...r.kernel.org
Cc: ceo@...-en-ming-corp.com
Subject: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960
IP Phones
Subject: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco
7960 IP Phones
Author: Mr. Turritopsis Dohrnii Teo En Ming (TARGETED INDIVIDUAL)
Country: Singapore
Date: 24 December 2020 Thursday Singapore Time
Type of Publication: Plain Text
Document version: 20201224.01
======================================================================================================================================================================
System Information
==================
My Asterisk version: 16.13.0
My FreePBX version: 15.0.16.81
On 7 December 2020, I was able to get Bria softphone to work with my
Asterisk PBX server successfully (PJSIP extension).
On 19 December 2020, I bought a refurbished Cisco CP-7960G IP hardphone
for SGD$30 in Singapore.
TFTP works. My DHCP server in my pfSense firewall applaince is able to
assign my Cisco 7960 IP phone with an IP address with DHCP option 66
(TFTP server). My Cisco 7960 IP phone is able to connect to my TFTP
server on my Asterisk PBX appliance and download firmware and
configuration files successfully.
On 24 December 2020 Thursday Christmas Eve, I have finally managed to
get my Cisco 7960 IP phone to register on my Asterisk PBX server
***successfully***.
This is an ***OLD AND OUTDATED*** video of my Cisco 7960 IP phone:
https://www.youtube.com/watch?v=ip_F08jmmio
I will publish new and updated Youtube video of my Cisco 7960 IP phone
***in the future***.
BEGINNING OF THIS GUIDE
=======================
Reference Guide: Configure Asterisk with Cisco IP Phones
Link: http://docshare02.docshare.tips/files/6706/67061980.pdf
SECTION 1: INSTALLING TFTP SERVER ON ASTERISK PBX APPLIANCE
===========================================================
Putty/ssh into Teo En Ming's Asterisk VoIP IP PBX SIP Server at
192.168.1.9.
# yum install tftp-server
Package tftp-server-5.2-23.8.sng7.x86_64 already installed and latest
version
# chkconfig xinetd on
# chkconfig tftp on
# systemctl start tftp.service
# ps -ef | grep tftp
root 3424 1 0 11:17 ? 00:00:00 /usr/sbin/in.tftpd -s
/tftpboot
SECTION 2: DOWNLOADING CISCO 7960 IP PHONE SIP FIRMWARE
=======================================================
# cd /tftpboot
# wget
http://www.firewall.cx/downloads/cisco-tools-a-applications/cisco-ip-phone-a-ata-firmware-downloads/107-7940-a-7960-ip-phone-sccp-a-sip/file.html
# mv file.html file.zip
# unzip file.zip
# cd 7940_7960/
# cd SIP/
# tar -xf P0S3-8-12-00.tar
# rm P0S3-8-12-00.tar
# mv * /tftpboot/
# cd /tftpboot/
[root@...epbx tftpboot]# ls
7940_7960 file.zip OS79XX.TXT P003-8-12-00.bin P003-8-12-00.sbn
P0S3-8-12-00.loads P0S3-8-12-00.sb2
SECTION 3: CREATING CISCO 7960 IP PHONE CONFIGURATION FILES
===========================================================
# nano OS79XX.TXT (Create configuration file)
=============================================
P003-8-12-00
# nano XMLDefault.cnf.xml (Create configuration file)
=====================================================
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model="IP Phone 7940">P0S3-8-12-00</loadInformation8>
<loadInformation7 model="IP Phone 7960">P0S3-8-12-00</loadInformation7>
<loadInformation435 model="Cisco 7945">SIP45.8-4-2S</loadInformation435>
<loadInformation436 model=”Cisco 7965”>SIP45.8-4-2S</loadInformation436>
<loadInformation30006 model="IP Phone
7970">SIP70.8-0-3S</loadInformation30006>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
# nano SIPDefault.cnf (Create configuration file)
=================================================
image_version: "P0S3-8-12-00"
proxy1_address: "192.168.1.9"
# proxy2_address: "xxx.xxx.xxx.xxx"
# proxy3_address: "xxx.xxx.xxx.xxx"
# proxy4_address: "xxx.xxx.xxx.xxx"
# Proxy Server Port
proxy1_port:"5060"
# proxy2_port:"5060"
# proxy3_port:"5060"
# proxy4_port:"5060"
proxy_emergency: ""
proxy_emergency_port: "5060"
proxy_backup: ""
proxy_backup_port: "5060"
outbound_proxy: ""
outbound_proxy_port: "5060"
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "1"
dyn_dns_addr_1: ""
dyn_dns_addr_2: ""
dyn_tftp_addr: "192.168.1.9"
tftp_cfg_dir: "./"
proxy_register: "1"
timer_register_expires: "120"
preferred_codec: "none"
tos_media: "5"
enable_vad: "0"
dial_template: "dialplan"
network_media_type: "auto"
autocomplete: "1"
telnet_level: "2"
cnf_join_enable: "1"
semi_attended_transfer: "0"
call_waiting: "1"
anonymous_call_block: "0"
callerid_blocking: "0"
dnd_control: "0"
dtmf_inband: "1"
dtmf_outofband: "avt"
dtmf_db_level: "3"
dtmf_avt_payload: "101"
timer_t1: "500"
timer_t2: "4000"
sip_retx: "10"
sip_invite_retx: "6"
timer_invite_expires: "180"
sntp_mode: "directedbroadcast"
sntp_server: "time-a-g.nist.gov"
time_zone: "8"
time_format_24hr: "0"
dst_offset: "0"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "2"
dst_stop_month: "Nov"
dst_stop_day: "1"
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: ""
dst_stop_time: "2"
dst_auto_adjust: "1"
messages_uri: "*99"
services_url: "http://example.domain.ext/services/menu.xml"
directory_url: "http://example.domain.ext/services/directory.php"
logo_url: "http://example.domain.ext/imagename.bmp"
http_proxy_addr: ""
http_proxy_port: ""
remote_party_id: 0
# nano dialplan.xml (Create configuration file)
===============================================
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>
# nano RINGLIST.DAT (Create configuration file)
===============================================
FlintPhone FlintPhone.raw
HarpSynth HarpSynth.raw
Jamaica Jamaica.raw
Klaxons Klaxons.raw
KotoEffect KotoEffect.raw
MusicBox MusicBox.raw
Ohno Ohno.raw
Piano 1 Piano1.raw
Piano 2 Piano2.raw
# nano SEP000E84C060A6.cnf.xml (Create configuration file)
==========================================================
<device>
<deviceProtocol>SIP</deviceProtocol>
<loadInformation model="IP Phone 7960">P0S3-8-12-00</loadInformation>
</device>
# nano SIP000E84C060A6.cnf (Create configuration file)
=======================================================
proxy1_address: "192.168.1.9"
proxy2_address: "xxx.xxx.xxx.xxx"
proxy3_address: "xxx.xxx.xxx.xxx"
proxy4_address: "xxx.xxx.xxx.xxx"
line1_name: "1600"
line1_shortname: "TEO EN MING"
line1_displayname: "TURRITOPSIS DOHRNII TEO EN MING"
line1_authname: "1600"
line1_password: "IP Phone Extension Password"
line2_name: ""
line2_shortname: ""
line2_displayname: ""
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
line3_name: ""
line3_shortname: ""
line3_displayname: ""
line3_authname: "UNPROVISIONED"
line3_password: "UNPROVISIONED"
line4_name: ""
line4_shortname:
line4_displayname: ""
line4_authname: "UNPROVISIONED"
line4_password: "UNPROVISIONED"
line5_name: ""
line5_shortname: ""
line5_displayname: ""
line5_authname: "UNPROVISIONED"
line5_password: "UNPROVISIONED"
line6_name: ""
line6_shortname: ""
line6_displayname: ""
line6_authname: "UNPROVISIONED"
line6_password: "UNPROVISIONED"
proxy_emergency: ""
proxy_emergency_port: "5060"
proxy_backup: ""
proxy_backup_port: "5060"
outbound_proxy: ""
outbound_proxy_port: "5060"
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "0"
phone_label: "TEO EN MING CORP "
time_zone: 8
logo_url: "http://domain.ext/imagefile.bmp"
telnet_level: "2"
phone_prompt: "Cisco7960"
phone_password: "password"
enable_vad: "0"
network_media_type: "auto"
user_info: phone
SECTION 4: GETTING TFTP SERVER TO START AUTOMATICALLY UPON EVERY BOOT
=====================================================================
# systemctl enable tftp
# systemctl start tftp
[root@...epbx ~]# ps -ef | grep tftp
root 4097 1 0 02:24 ? 00:00:00 /usr/sbin/in.tftpd -s
/tftpboot
SECTION 5: CONFIGURING PFSENSE FIREWALL'S DHCP SERVER WITH DHCP OPTION
66 (TFTP SERVER)
=======================================================================================
Reference Guide: Chapter: Getting Started With Your Cisco SIP IP Phone
(Version 3.0)
Link:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7960g_7940g/sip/3_0/english/administration/guide/ver3_0/install.html
[QUOTE]
Configuring Network Parameters
==============================
Note This section describes how to configure the basic network
parameters that are required for the phone to operate on the network.
For a complete list of the network parameters that you can configure,
see the "Modifying the Phone's Network Settings" section.
The network parameters include those parameters that must be configured
on a phone for the phone to operate in an IP network. You can configure
the required network parameters via DHCP or manually configure them
after you have connected the phone to a power supply.
The following parameters must be defined for your phone to establish
network connectivity:
•Phone's IP address
•Subnet mask
•Default gateway for the subnet (use "0.0.0.0" if not required)
•Domain name
•DNS server IP address (use "0.0.0.0" if not required)
•TFTP server IP address
When configuring the network parameters of an IP phone, adhere to the
following guidelines:
•Use 0.0.0.0 for unused IP addresses.
•You can use 0.0.0.0 for the subnet mask only if the default gateway is
also 0.0.0.0.
•The TFTP server must have a nonzero IP address.
•The default gateway must be on the same subnet as the phone.
•The default gateway can be 0.0.0.0 only if the TFTP or DNS server is on
the same subnet as the phone.
Note By default, DHCP is enabled on your phone. Before you can manually
configure the network parameters, you must disable DHCP after connecting
your phone to a power supply.
Configuring Network Parameters via a DHCP Server
================================================
If you are using DHCP to configure the network parameters, configure the
following DHCP options on your DHCP server before you connect your Cisco
SIP IP phone:
•dhcp option #50 (IP address)
•dhcp option #1 (IP subnet mask)
•dhcp option #3 (Default IP gateway)
•dhcp option #15 (Domain name)
•dhcp option #6 (DNS server IP address)
•dhcp option #66 (TFTP server IP address)
[/QUOTE]
Reference Guide: pfSense Firewall DHCPv4 Server
Link: https://docs.netgate.com/pfsense/en/latest/services/dhcp/ipv4.html
Login to Teo En Ming's pfSense firewall appliance.
Click Services > DHCP Server.
At TFTP, click Display Advanced.
TFTP Server: 192.168.1.9
Please remember to click Save.
Then click Restart Service. You must restart the DHCP server for the
changes to take effect.
SECTION 6: PERFORM FACTORY RESET ON CISCO 7960 IP PHONE
=======================================================
Because I bought an used Cisco 7960 IP phone, I must factory reset it to
clear all previous owner's settings.
Reference Guide: Reset 7900 Series IP Phones if Password is Set
Link:
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-ip-phone-7900-series/15254-7960-default.html
[QUOTE]
Reset the 7940 and 7960 IP Phones to the Factory Default
=========================================================
In order to perform a factory reset of a phone if the password is set,
complete these steps:
Unplug the power cable from the phone, and then plug in the cable again.
The phone begins its power up cycle.
Immediately press and hold # and while the Headset, Mute, and Speaker
buttons begin to flash in sequence, release #.
The Headset, Mute, and Speaker buttons flash in sequence in order to
indicate that the phone waits for you to enter the key sequence for the
reset.
Press 123456789*0# within 60 seconds after the Headset, Mute, and
Speaker buttons begin to flash.
If you repeat a key within the sequence, for example, if you press
1223456789*0#, the sequence is still accepted and the phone resets.
If you do not complete this key sequence or do not press any keys, after
60 seconds, the Headset, Mute, and Speaker buttons no longer flash, and
the phone continues with its normal startup process. The phone does not
reset.
If you enter an invalid key sequence, the buttons no longer flash, and
the phone continues with its normal startup process. The phone does not
reset.
If you enter this key sequence correctly, the phone displays this
prompt:
Keep network cfg? 1 = yes 2 = no
In order to maintain the current network configuration settings for the
phone when the phone resets, press 1. In order to reset the network
configuration settings when the phone resets, press 2.
If you press another key or do not respond to this prompt within 60
seconds, the phone continues with its normal startup process and does
not reset. Otherwise, the phone goes through the factory reset process.
[/QUOTE]
SECTION 7: TEMPORARILY DISABLE FREEPBX FIREWALL TO TEST TFTP CONNECTION
=======================================================================
Reference Guide: Firewall Command Line
Link: https://wiki.freepbx.org/display/FPG/Firewall+Command+Line
# fwconsole firewall disable
[root@...epbx ~]# iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source destination
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
[root@...epbx ~]# iptables -S
-P INPUT ACCEPT
-P FORWARD ACCEPT
-P OUTPUT ACCEPT
[root@...epbx ~]# tail -f /var/log/messages
Dec 20 03:24:42 freepbx systemd: Started Tftp Server.
Dec 20 03:25:25 freepbx xinetd[939]: START: tftp pid=8213
from=192.168.1.130
Dec 20 03:25:25 freepbx in.tftpd[8214]: RRQ from 192.168.1.130 filename
CTLSEP000E84C060A6.tlv
Dec 20 03:25:25 freepbx in.tftpd[8214]: Client 192.168.1.130 File not
found CTLSEP000E84C060A6.tlv
Dec 20 03:25:25 freepbx in.tftpd[8215]: RRQ from 192.168.1.130 filename
SEP000E84C060A6.cnf.xml
Dec 20 03:25:25 freepbx in.tftpd[8215]: Client 192.168.1.130 finished
SEP000E84C060A6.cnf.xml
Dec 20 03:25:25 freepbx in.tftpd[8216]: RRQ from 192.168.1.130 filename
P0S3-8-12-00.loads
Dec 20 03:25:25 freepbx in.tftpd[8216]: Client 192.168.1.130 finished
P0S3-8-12-00.loads
Dec 20 03:25:49 freepbx in.tftpd[8217]: RRQ from 192.168.1.130 filename
SIPDefault.cnf
Dec 20 03:25:49 freepbx in.tftpd[8217]: Client 192.168.1.130 finished
SIPDefault.cnf
Dec 20 03:25:49 freepbx in.tftpd[8218]: RRQ from 192.168.1.130 filename
./SIP000E84C060A6.cnf
Dec 20 03:25:49 freepbx in.tftpd[8218]: Client 192.168.1.130 finished
./SIP000E84C060A6.cnf
Dec 20 03:25:51 freepbx in.tftpd[8219]: RRQ from 192.168.1.130 filename
RINGLIST.DAT
Dec 20 03:25:51 freepbx in.tftpd[8220]: RRQ from 192.168.1.130 filename
dialplan.xml
Dec 20 03:25:51 freepbx in.tftpd[8219]: Client 192.168.1.130 finished
RINGLIST.DAT
Dec 20 03:25:51 freepbx in.tftpd[8220]: Client 192.168.1.130 finished
dialplan.xml
SECTION 8: ENABLE AND START THE FREEPBX FIREWALL FOR PRODUCTION USE
===================================================================
[root@...epbx ~]# fwconsole firewall start
Enabling Firewall.
[root@...epbx ~]#
Broadcast message from root@...epbx.sangoma.local (Sun Dec 20 03:37:45
2020):
Firewall service now starting.
# iptables -L
# iptables -S
Login to FreePBX Dashboard.
Please check the following item:
System Firewall: Firewall Active
Power off and then power on the Cisco 7960 IP Phone.
[root@...epbx ~]# tail -f /var/log/messages
Dec 20 03:42:27 freepbx xinetd[939]: START: tftp pid=10209
from=192.168.1.130
Dec 20 03:42:27 freepbx in.tftpd[10210]: RRQ from 192.168.1.130 filename
CTLSEP000E84C060A6.tlv
Dec 20 03:42:27 freepbx in.tftpd[10210]: Client 192.168.1.130 File not
found CTLSEP000E84C060A6.tlv
Dec 20 03:42:28 freepbx in.tftpd[10211]: RRQ from 192.168.1.130 filename
SEP000E84C060A6.cnf.xml
Dec 20 03:42:28 freepbx in.tftpd[10211]: Client 192.168.1.130 finished
SEP000E84C060A6.cnf.xml
Dec 20 03:42:28 freepbx in.tftpd[10212]: RRQ from 192.168.1.130 filename
P0S3-8-12-00.loads
Dec 20 03:42:28 freepbx in.tftpd[10212]: Client 192.168.1.130 finished
P0S3-8-12-00.loads
Dec 20 03:42:51 freepbx in.tftpd[10286]: RRQ from 192.168.1.130 filename
SIPDefault.cnf
Dec 20 03:42:51 freepbx in.tftpd[10286]: Client 192.168.1.130 finished
SIPDefault.cnf
Dec 20 03:42:52 freepbx in.tftpd[10287]: RRQ from 192.168.1.130 filename
./SIP000E84C060A6.cnf
Dec 20 03:42:52 freepbx in.tftpd[10287]: Client 192.168.1.130 finished
./SIP000E84C060A6.cnf
Dec 20 03:42:53 freepbx in.tftpd[10288]: RRQ from 192.168.1.130 filename
RINGLIST.DAT
Dec 20 03:42:53 freepbx in.tftpd[10288]: Client 192.168.1.130 finished
RINGLIST.DAT
Dec 20 03:42:53 freepbx in.tftpd[10289]: RRQ from 192.168.1.130 filename
dialplan.xml
Dec 20 03:42:53 freepbx in.tftpd[10289]: Client 192.168.1.130 finished
dialplan.xml
SECTION 9: ASTERISK CLI TROUBLESHOOTING COMMANDS
================================================
# asterisk -vvvr
sip set debug on
sip show peers
Name/username Host Dyn
Forcerport Comedia ACL Port Status Description
didlogic/60751 107.6.123.181
Yes Yes 5060 OK (7 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0
offline]
SECTION 10: ASTERISK PBX SERVER DEBUGGING OUTPUT
================================================
# asterisk -vvvr
freepbx*CLI> sip set debug on
freepbx*CLI>
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for 60751@....sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
Max-Forwards: 70
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To: <sip:60751@....sg.didlogic.net>
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 165 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
cnonce="2b1b6d13", nc=00000003
Expires: 120
Contact: <sip:6531590313@....168.1.9:5160>
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To:
<sip:60751@....sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 165 REGISTER
Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC IP>:26462"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
handle_response_register: Outbound Registration: Expiry for
sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog
'005dbc8238e06ac421ef613a3b55e134@....0.0.1' Method: REGISTER
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
Max-Forwards: 70
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as41ddf4a6
To: <sip:sip.sg.didlogic.net>
Contact: <sip:60751@....168.1.9:5160>
Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@....168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as41ddf4a6
To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924
Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@....168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'0b0605df4f4ca7b03402c9fd5a869606@....168.1.9:5160' Method: OPTIONS
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK4ff08179;rport
Max-Forwards: 70
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as004073f3
To: <sip:sip.sg.didlogic.net>
Contact: <sip:60751@....168.1.9:5160>
Call-ID: 500c8eb32071e3fc462be80f243d38fc@....168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:08:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK4ff08179;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as004073f3
To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d
Call-ID: 500c8eb32071e3fc462be80f243d38fc@....168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'500c8eb32071e3fc462be80f243d38fc@....168.1.9:5160' Method: OPTIONS
[2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for 60751@....sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport
Max-Forwards: 70
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To: <sip:60751@....sg.didlogic.net>
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 166 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="074f1f037639144de751dc9231c191c9", qop=auth,
cnonce="6eb58a86", nc=00000004
Expires: 120
Contact: <sip:6531590313@....168.1.9:5160>
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To:
<sip:60751@....sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e426
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 166 REGISTER
WWW-Authenticate: Digest realm="sip.sg.didlogic.net",
nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=", qop="auth"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name
sip.sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport
Max-Forwards: 70
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To: <sip:60751@....sg.didlogic.net>
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 167 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
response="c1184dab1dd50dad14cba70933b6bbaa", qop=auth,
cnonce="4945f552", nc=00000001
Expires: 120
Contact: <sip:6531590313@....168.1.9:5160>
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To:
<sip:60751@....sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f557
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 167 REGISTER
Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC IP>:26462"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
[2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:24961
handle_response_register: Outbound Registration: Expiry for
sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog
'005dbc8238e06ac421ef613a3b55e134@....0.0.1' Method: REGISTER
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK0f99cec8;rport
Max-Forwards: 70
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as786e56f3
To: <sip:sip.sg.didlogic.net>
Contact: <sip:60751@....168.1.9:5160>
Call-ID: 49595bb227bd6e390ca40ce6508308aa@....168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:09:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK0f99cec8;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as786e56f3
To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e4fb
Call-ID: 49595bb227bd6e390ca40ce6508308aa@....168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'49595bb227bd6e390ca40ce6508308aa@....168.1.9:5160' Method: OPTIONS
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
[2020-12-20 07:09:52] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for 60751@....sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK030dc571;rport
Max-Forwards: 70
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To: <sip:60751@....sg.didlogic.net>
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 168 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
response="b6e96ab578798296e812139c383ebbac", qop=auth,
cnonce="6313e774", nc=00000002
Expires: 120
Contact: <sip:6531590313@....168.1.9:5160>
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK030dc571;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: <sip:60751@....sg.didlogic.net>;tag=as6df6d977
To:
<sip:60751@....sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2fbb
Call-ID: 005dbc8238e06ac421ef613a3b55e134@....0.0.1
CSeq: 168 REGISTER
Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC IP>:26462"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
[2020-12-20 07:09:52] NOTICE[2366]: chan_sip.c:24961
handle_response_register: Outbound Registration: Expiry for
sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog
'005dbc8238e06ac421ef613a3b55e134@....0.0.1' Method: REGISTER
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK25d81863;rport
Max-Forwards: 70
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as3a8d9861
To: <sip:sip.sg.didlogic.net>
Contact: <sip:60751@....168.1.9:5160>
Call-ID: 4f9667021511db2968a0dc9964673ac9@....168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:10:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK25d81863;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as3a8d9861
To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a139
Call-ID: 4f9667021511db2968a0dc9964673ac9@....168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'4f9667021511db2968a0dc9964673ac9@....168.1.9:5160' Method: OPTIONS
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
<--- SIP read from UDP:107.6.123.181:5060 --->
<------------->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK443d3196;rport
Max-Forwards: 70
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as5d241373
To: <sip:sip.sg.didlogic.net>
Contact: <sip:60751@....168.1.9:5160>
Call-ID: 201fab2b13e08922618f34911ce37ce0@....168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:11:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK443d3196;rport=26462;received=<CORPORATE
OFFICE PUBLIC IP>
From: "Unknown" <sip:60751@....168.1.9:5160>;tag=as5d241373
To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.19e3
Call-ID: 201fab2b13e08922618f34911ce37ce0@....168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'201fab2b13e08922618f34911ce37ce0@....168.1.9:5160' Method: OPTIONS
<--- SIP read from UDP:107.6.123.181:5060 --->
freepbx*CLI>
SECTION 11: FINAL WORKING CONFIGURATION
=======================================
I changed pjsip extension password to 8 characters, numeric only. My
Cisco 7960 IP phone still can't register on Asterisk PBX server.
Now, login to Teo En Ming's Asterisk VoIP IP PBX SIP Server at
192.168.1.9.
Click Applications > Extensions
Click the pencil icon to edit extension 1600.
Click the Advanced tab.
You MUST make sure "Force rport" is set to No. OTHERWISE, YOUR CISCO
79XX IP PHONE WILL NOT REGISTER ON YOUR ASTERISK PBX SERVER.
Click Submit.
Click Apply Config. VERY IMPORTANT! MUST CLICK THIS BUTTON!
My Cisco 7960 IP phone now ***SUCCESSFULLY*** registers on my Asterisk
PBX server, using PJSIP extension 1600. There is no more "X" next to
line 1 button on the phone.
Yes, my Cisco 7960 IP phone now ***SUCCESSFULLY*** registers on my
Asterisk PBX server, using PJSIP extension 1600. Repeat for emphasis.
I can now make outgoing phone calls on my Cisco 7960 IP phone.
SECTION 12: CAVEATS
===================
I can now make outgoing phone calls on my Cisco 7960 IP phone.
However, if Voicemail in my extension configuration is set to Disabled,
and when I try to dial my DID number, it says "Number not valid".
Only when I change Voicemail to Enabled, and when I try to dial my DID
number, it says:
"The person at extension 1600 is unavailable. Please leave your message
after the tone, when done hang up or press the # key."
It seems that incoming calls will not be routed to extension 1600.
I will need to do more troubleshooting on this at a future date.
SECTION 13: TIPS
================
When you make modifications to Asterisk configuration files, you need to
issue the following commands to make changes take effect.
# asterisk -vvvr
freepbx*CLI> core reload
SECTION 14: TROUBLESHOOTING
===========================
Run tcpdump on your Asterisk PBX server.
# tcpdump -i eth0 -vvv | grep -v ssh
tcpdump output:
07:26:28.535667 IP (tos 0x10, ttl 128, id 0, offset 0, flags [none],
proto UDP (17), length 337)
pfSense.teo-en-ming-corp.com.bootps > 255.255.255.255.bootpc: [udp
sum ok] BOOTP/DHCP, Reply, length 309, xid 0x84c060a6, Flags [Broadcast]
(0x8000)
Your-IP 192.168.1.130
Client-Ethernet-Address 00:0e:84:c0:60:a6 (oui Unknown)
Vendor-rfc1048 Extensions
Magic Cookie 0x63825363
DHCP-Message Option 53, length 1: ACK
Server-ID Option 54, length 4: pfSense.teo-en-ming-corp.com
Lease-Time Option 51, length 4: 5636
Subnet-Mask Option 1, length 4: 255.255.255.0
TFTP Option 66, length 11: "192.168.1.9"
Domain-Name-Server Option 6, length 4:
pfSense.teo-en-ming-corp.com
Default-Gateway Option 3, length 4:
pfSense.teo-en-ming-corp.com
Domain-Name Option 15, length 20: "teo-en-ming-corp.com"
END Option 255, length 0
07:26:30.803900 IP (tos 0x60, ttl 64, id 1960, offset 0, flags [none],
proto UDP (17), length 542)
192.168.1.130.51098 > freepbx.sangoma.local.sip: [no cksum] SIP,
length: 514
REGISTER sip:192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK5ab98907
From:
<sip:1600@....168.1.9>;tag=000e84c060a60002421e8cdc-0e061182
To: <sip:1600@....168.1.9>
Call-ID: 000e84c0-60a60002-3cf23940-64dcadcc@....168.1.130
Max-Forwards: 70
CSeq: 101 REGISTER
User-Agent: Cisco-CP7960G/8.0
Contact:
<sip:1600@....168.1.130:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000e84c060a6>";+u.sip!model.ccm.cisco.com="7"
Content-Length: 0
Expires: 120
07:26:30.804791 IP (tos 0x60, ttl 64, id 33765, offset 0, flags [DF],
proto UDP (17), length 535)
freepbx.sangoma.local.sip > 192.168.1.130.sip: [bad udp cksum 0x85f0
-> 0xc369!] SIP, length: 507
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.130:5060;received=192.168.1.130;branch=z9hG4bK5ab98907
Call-ID: 000e84c0-60a60002-3cf23940-64dcadcc@....168.1.130
From:
<sip:1600@....168.1.9>;tag=000e84c060a60002421e8cdc-0e061182
To: <sip:1600@....168.1.9>;tag=z9hG4bK5ab98907
CSeq: 101 REGISTER
WWW-Authenticate: Digest
realm="asterisk",nonce="1608794790/f4dc9767e5f40180ee8cb4cd78bd9d3e",opaque="4cf02bb020c6e7ae",algorithm=md5,qop="auth"
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
07:26:30.900961 IP (tos 0x60, ttl 64, id 1961, offset 0, flags [none],
proto UDP (17), length 795)
192.168.1.130.51098 > freepbx.sangoma.local.sip: [no cksum] SIP,
length: 767
REGISTER sip:192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK10cd9d0e
From:
<sip:1600@....168.1.9>;tag=000e84c060a60002421e8cdc-0e061182
To: <sip:1600@....168.1.9>
Call-ID: 000e84c0-60a60002-3cf23940-64dcadcc@....168.1.130
Max-Forwards: 70
CSeq: 102 REGISTER
User-Agent: Cisco-CP7960G/8.0
Contact:
<sip:1600@....168.1.130:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000e84c060a6>";+u.sip!model.ccm.cisco.com="7"
Authorization: Digest
username="1600",realm="asterisk",uri="sip:192.168.1.9",response="40ae56341397370a90d6720f19562523",nonce="1608794790/f4dc9767e5f40180ee8cb4cd78bd9d3e",opaque="4cf02bb020c6e7ae",cnonce="61cdcfdc",qop=auth,nc=00000001,algorithm=md5
Content-Length: 0
Expires: 120
07:26:30.902283 IP (tos 0x60, ttl 64, id 33830, offset 0, flags [DF],
proto UDP (17), length 494)
freepbx.sangoma.local.sip > 192.168.1.130.sip: [bad udp cksum 0x85c7
-> 0x7d06!] SIP, length: 466
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.130:5060;received=192.168.1.130;branch=z9hG4bK10cd9d0e
Call-ID: 000e84c0-60a60002-3cf23940-64dcadcc@....168.1.130
From:
<sip:1600@....168.1.9>;tag=000e84c060a60002421e8cdc-0e061182
To: <sip:1600@....168.1.9>;tag=z9hG4bK10cd9d0e
CSeq: 102 REGISTER
Date: Thu, 24 Dec 2020 07:26:30 GMT
Contact: <sip:1600@....168.1.130:51098;user=phone>;expires=119
Expires: 120
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
SECTION 15 : ONLINE DISCUSSIONS AND YOUTUBE VIDEOS FOR READING AND
WATCHING LATER
=================================================================================
Post: Cisco 7960 SIP Registration Problem
Link:
https://community.freepbx.org/t/cisco-7960-sip-registration-problem/19110
Post: Updated FreePBX install instructions for Chan-SCCP-B
Link:
https://community.freepbx.org/t/updated-freepbx-install-instructions-for-chan-sccp-b/37077
Post: How to get FreePBX to work with Cisco 7960
Link:
https://community.freepbx.org/t/how-to-get-freepbx-to-work-with-cisco-7960/20195
Post: Cisco 7940 registration problem RESOLVED
Link:
https://community.freepbx.org/t/cisco-7940-registration-problem-resolved/30285
Post: Help with registering Cisco 7940s in FreePBX
Link:
https://community.freepbx.org/t/help-with-registering-cisco-7940s-in-freepbx/30555/4
Post: HELP! I can’t get my Cisco CP-7960G IP hardphone to register on my
Asterisk VoIP IP PBX SIP Server with FreePBX GUI
Link:
https://community.freepbx.org/t/help-i-cant-get-my-cisco-cp-7960g-ip-hardphone-to-register-on-my-asterisk-voip-ip-pbx-sip-server-with-freepbx-gui/72162
Post: Can not get Cisco Phones to Register
Link:
https://community.freepbx.org/t/can-not-get-cisco-phones-to-register/48784/21
Youtube: Cisco 7942g IP Phone Configuration on FreePBX In-Depth(Without
Endpoint Manager)
Link: https://www.youtube.com/watch?v=gk6w8O3fZlc&feature=youtu.be
Mr. Turritopsis Dohrnii Teo En Ming, 42 years old as of 24 December 2020
Thursday Christmas Eve, is a TARGETED INDIVIDUAL (TI) living in
Singapore. He is an IT Consultant with a System Integrator (SI)/computer
firm in Singapore. He is an IT enthusiast.
--
-----BEGIN EMAIL SIGNATURE-----
The Gospel for all Targeted Individuals (TIs):
[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers
Link:
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html
********************************************************************************************
Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug
2019) and Australia (25 Dec 2019 to 9 Jan 2020):
[1] https://tdtemcerts.wordpress.com/
[2] https://tdtemcerts.blogspot.sg/
[3] https://www.scribd.com/user/270125049/Teo-En-Ming
-----END EMAIL SIGNATURE-----
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