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This is an ***OLD AND OUTDATED*** video of my Cisco 7960 IP phone: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@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)

=====================================================

<ports>

</mgcpPorts>

</ports>

</callManager>

</member>

</members>

</callManagerGroup>

</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>

# 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>

# 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@freepbx ~]# 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

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@freepbx ~]# 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@freepbx ~]# iptables -S

-P INPUT ACCEPT

-P FORWARD ACCEPT

-P OUTPUT ACCEPT

[root@freepbx ~]# 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@freepbx ~]# fwconsole firewall start

Enabling Firewall.

[root@freepbx ~]#

Broadcast message from root@freepbx.sangoma.local (Sun Dec 20 03:37:45 2020):

Firewall service now starting.

# iptables -L

# iptables -S

Please check the following item:

System Firewall: Firewall Active

Power off and then power on the Cisco 7960 IP Phone.

[root@freepbx ~]# 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@sip.sg.didlogic.net

Reliably Transmitting (NAT) to 107.6.123.181:5060:

Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport

Max-Forwards: 70

From: <sip:60751@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@192.168.1.9:5160>

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@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@127.0.0.1' 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@192.168.1.9:5160>;tag=as41ddf4a6

To: <sip:sip.sg.didlogic.net>

Contact: <sip:60751@192.168.1.9:5160>

Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160

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@192.168.1.9:5160>;tag=as41ddf4a6

To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924

Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160

CSeq: 102 OPTIONS

Content-Length: 0

<------------->

--- (7 headers 0 lines) ---

Really destroying SIP dialog '0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160' 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@192.168.1.9:5160>;tag=as004073f3

To: <sip:sip.sg.didlogic.net>

Contact: <sip:60751@192.168.1.9:5160>

Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160

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@192.168.1.9:5160>;tag=as004073f3

To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d

Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160

CSeq: 102 OPTIONS

Content-Length: 0

<------------->

--- (7 headers 0 lines) ---

Really destroying SIP dialog '500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160' Method: OPTIONS

[2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:15893 sip_reregister: -- Re-registration for 60751@sip.sg.didlogic.net

Reliably Transmitting (NAT) to 107.6.123.181:5060:

Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport

Max-Forwards: 70

From: <sip:60751@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@192.168.1.9:5160>

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@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e426

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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

Reliably Transmitting (NAT) to 107.6.123.181:5060:

Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport

Max-Forwards: 70

From: <sip:60751@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@192.168.1.9:5160>

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@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f557

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@127.0.0.1' 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@192.168.1.9:5160>;tag=as786e56f3

To: <sip:sip.sg.didlogic.net>

Contact: <sip:60751@192.168.1.9:5160>

Call-ID: 49595bb227bd6e390ca40ce6508308aa@192.168.1.9:5160

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@192.168.1.9:5160>;tag=as786e56f3

To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e4fb

Call-ID: 49595bb227bd6e390ca40ce6508308aa@192.168.1.9:5160

CSeq: 102 OPTIONS

Content-Length: 0

<------------->

--- (7 headers 0 lines) ---

Really destroying SIP dialog '49595bb227bd6e390ca40ce6508308aa@192.168.1.9:5160' 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@sip.sg.didlogic.net

Reliably Transmitting (NAT) to 107.6.123.181:5060:

Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK030dc571;rport

Max-Forwards: 70

From: <sip:60751@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@192.168.1.9:5160>

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@sip.sg.didlogic.net>;tag=as6df6d977

To: <sip:60751@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2fbb

Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1

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@127.0.0.1' 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@192.168.1.9:5160>;tag=as3a8d9861

To: <sip:sip.sg.didlogic.net>

Contact: <sip:60751@192.168.1.9:5160>

Call-ID: 4f9667021511db2968a0dc9964673ac9@192.168.1.9:5160

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@192.168.1.9:5160>;tag=as3a8d9861

To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a139

Call-ID: 4f9667021511db2968a0dc9964673ac9@192.168.1.9:5160

CSeq: 102 OPTIONS

Content-Length: 0

<------------->

--- (7 headers 0 lines) ---

Really destroying SIP dialog '4f9667021511db2968a0dc9964673ac9@192.168.1.9:5160' 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@192.168.1.9:5160>;tag=as5d241373

To: <sip:sip.sg.didlogic.net>

Contact: <sip:60751@192.168.1.9:5160>

Call-ID: 201fab2b13e08922618f34911ce37ce0@192.168.1.9:5160

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@192.168.1.9:5160>;tag=as5d241373

To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.19e3

Call-ID: 201fab2b13e08922618f34911ce37ce0@192.168.1.9:5160

CSeq: 102 OPTIONS

Content-Length: 0

<------------->

--- (7 headers 0 lines) ---

Really destroying SIP dialog '201fab2b13e08922618f34911ce37ce0@192.168.1.9:5160' 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

Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK5ab98907

From: <sip:1600@192.168.1.9>;tag=000e84c060a60002421e8cdc-0e061182

To: <sip:1600@192.168.1.9>

Call-ID: 000e84c0-60a60002-3cf23940-64dcadcc@192.168.1.130

Max-Forwards: 70

CSeq: 101 REGISTER

User-Agent: Cisco-CP7960G/8.0

Contact: <sip:1600@192.168.1.130:5060;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@192.168.1.130

From: <sip:1600@192.168.1.9>;tag=000e84c060a60002421e8cdc-0e061182

To: <sip:1600@192.168.1.9>;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

Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK10cd9d0e

From: <sip:1600@192.168.1.9>;tag=000e84c060a60002421e8cdc-0e061182

To: <sip:1600@192.168.1.9>

Call-ID: 000e84c0-60a60002-3cf23940-64dcadcc@192.168.1.130

Max-Forwards: 70

CSeq: 102 REGISTER

User-Agent: Cisco-CP7960G/8.0

Contact: <sip:1600@192.168.1.130:5060;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@192.168.1.130

From: <sip:1600@192.168.1.9>;tag=000e84c060a60002421e8cdc-0e061182

To: <sip:1600@192.168.1.9>;tag=z9hG4bK10cd9d0e

CSeq: 102 REGISTER

Date: Thu, 24 Dec 2020 07:26:30 GMT

Contact: <sip:1600@192.168.1.130:51098;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.

REFERENCES

==========

[1] http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html

[2] https://lists.sandelman.ca/pipermail/tcpdump-workers/2020-December/001595.html

[3] https://lkml.org/lkml/2020/12/24/82

[4] http://lkml.iu.edu/hypermail/linux/kernel/2012.3/00081.html

[5] https://marc.info/?l=linux-kernel&m=160879957428376&w=2

[6] https://lwn.net/ml/linux-kernel/1aa82a4137514f65d417a47a57483bc0%40teo-en-ming.com/

[7] https://marc.info/?l=netfilter&m=160879999428484&w=2

[8] http://lists.linuxfromscratch.org/pipermail/lfs-chat/2020-December/029174.html