[Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
Walid Azab
wazab at star-communications.net
Tue Jul 26 09:30:22 MST 2005
I went from 5.3 to 6.3 then from 6.3 t 7.5 directly. However, I have the
warning message (Protocol Application Invalid)!!!!
Please any help.
Walid
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Watkins,
Bradley
Sent: Tuesday, July 26, 2005 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
I believe you have to upgrade to 5.3 in order to go from unsigned to signed
executables. Once you're at 5.3, you can go directly to 7.5. I did this
recently with a couple of 7960s I had in the lab and it worked perfectly.
Regards,
- Brad
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Walid Azab
Sent: Tuesday, July 26, 2005 10:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
Hi,
I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go
up to 7.5
However in my first attempt to go from V.5.1 to 6.0 this is hat happens:
- The phone reboots
- The phone then reads the file OS79XX.TXT from the TFP server. In the file
I added the version "P0S3-06-0-00"
- It starts upgrading firmware
- Then I get the following message: (Upgrade Failed - Unauthorized)
Any ideas? Please find below my conf files.
SIP.CONF
[300]
username=300
type=friend
secret=cisco
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
mailbox=300 at default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="" <300>
SIP000CCE351C07.cnf
# SIP Configuration Generic File (start)
# Line 1 Settings
line1_name: "300" ; Line 1 Extension\User ID
line1_displayname: "300" ; Line 1 Display Name
line1_authname: "300" ; Line 1 Registration Authentication
line1_password: "cisco" ; Line 1 Registration Password
# Line 2 Settings
line2_name: "" ; Line 2 Extension\User ID
line2_displayname: "" ; Line 2 Display Name
line2_authname: "UNPROVISIONED" ; Line 2 Registration Authentication
line2_password: "UNPROVISIONED" ; Line 2 Registration Password
# Line 3 Settings
line3_name: "" ; Line 3 Extension\User ID
line3_displayname: "" ; Line 3 Display Name
line3_authname: "UNPROVISIONED" ; Line 3 Registration Authentication
line3_password: "UNPROVISIONED" ; Line 3 Registration Password
# Line 4 Settings
line4_name: "" ; Line 4 Extension\User ID
line4_displayname: "" ; Line 4 Display Name
line4_authname: "UNPROVISIONED" ; Line 4 Registration Authentication
line4_password: "UNPROVISIONED" ; Line 4 Registration Password
# Line 5 Settings
line5_name: "" ; Line 5 Extension\User ID
line5_displayname: "" ; Line 5 Display Name
line5_authname: "UNPROVISIONED" ; Line 5 Registration Authentication
line5_password: "UNPROVISIONED" ; Line 5 Registration Password
# Line 6 Settings
line6_name: "" ; Line 6 Extension\User ID
line6_displayname: "" ; Line 6 Display Name
line6_authname: "UNPROVISIONED" ; Line 6 Registration Authentication
line6_password: "UNPROVISIONED" ; Line 6 Registration Password
# NAT/Firewall Traversal
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32766"
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging
# Time Zone phone will reside in
time_zone: EST
# Phone prompt/password for telnet/console session
phone_prompt: "Cisco7960" ; Telnet/Console
Prompt
phone_password: "abc" ; Telnet/Console Password
# SIP Configuration Generic File (stop)
SIPDefault.cnf
# Image Version
image_version: "P0S3-06-0-00"
# Proxy Server
proxy1_address: "10.150.200.165"
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
# Emergency Proxy info
proxy_emergency: "10.150.200.165"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "10.150.200.165"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"
# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "none"
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"
# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"
# Out of band DTMF Settings (none-disable, avt-avt enable (default),
avt_always - always avt )
dtmf_outofband: "avt"
# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db
up, 5-6dB up)
dtmf_db_level: "3"
# SIP Timers
timer_t1: "500" ; Default 500 msec
timer_t2: "4000" ; Default 4 sec
sip_retx: "10" ; Default 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "*97"
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "unicast"
sntp_server: "10.150.200.165"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: "1"
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with
no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is
off)
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control,
3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as
anonymous)
# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user
control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous
calls)
# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control,
3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100
# XML file that specifies the dialplan desired
dial_template: "dialplan"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "0"
# URL for external Phone Services
services_url: "http://10.150.200.165/cisco/directory/services.php"
# URL for external Directory location
directory_url: "http://10.150.200.165/cisco/directory/directory.php"
# URL for branding logo
logo_url: "http://10.150.200.165/cisco/aah.bmp"
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
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