[tcpdump-workers] Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones

2020-12-24 Thread Turritopsis Dohrnii Teo En Ming via tcpdump-workers
--- Begin Message ---
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This is an ***OLD AND OUTDATED*** video of my Cisco 7960 IP phone: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@freepbx 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)
=



 
 
 
 
 2000
 
 2427
 2428
 
 
 
 
 
 
 
P0S3-8-12-00
P0S3-8-12-00
SIP45.8-4-2S
SIP45.8-4-2S
SIP70.8-0-3S









# 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_

[tcpdump-workers] Addendum to Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones

2020-12-24 Thread Turritopsis Dohrnii Teo En Ming via tcpdump-workers
--- Begin Message ---
Subject: Addendum to 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: 25 Dec 2020 Friday Singapore Time

Type of Publication: Plain Text

Document version: 20201225.01

=

Please refer to Teo En Ming's earlier guide.

Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 
7960 IP Phones
Link: 
http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html


In the above mentioned guide, under Section 12: Caveats, it was 
mentioned as follows:


"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."

Now, I am pleased to announce that, this final issue has been resolved 
on Christmas Day 2020.


The steps taken to resolve this final issue are as follows:

Login to Teo En Ming's FreePBX GUI at https://192.168.1.9.

Click Applications > Extensions.

Click the Pencil icon to edit the PJSIP extension 1600.

Click Advanced tab.

For Rewrite Contact, you MUST change the setting from Yes to No.

Click Voicemail tab.

Change Enabled to Yes.

For Require From Same Extension, click Yes.

Click Submit.

You MUST click Apply Config for the changes to take effect!

Voila! All issues have been solved.

Mr. Turritopsis Dohrnii Teo En Ming, 42 years old as of 25 December 2020 
Friday, 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.






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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

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