ftocc

stupid phone provision on 7960G.....Can't cope anymore!!! lol

Rexz
Posts: 1
Member Since:
2008-08-14

Hi,
I think I have finaly come to a stand still in thinking what to do next. I have trixbox CE 2.2 and a cisco 7960G. I have spent soooo long trying to get these two to work with absolutely no luck! I have fully upgraded the 7960G to version 8.9 of SIP but still no luck.
I was able to iron out all the errors in the SIPDefault.cnf but now I am stuck with "W362 No Valid Line Names" and the phone unprovisioned. I have looked all over these forums, trying numerous ways as people have suggested, trying to get this to work but with no luck. Looking at the logs of the tftp and the telnet screen, the phone downloads the SIP0016C7B74967.cnf file from the tftp server but seems to do nothing with the contents. No errors are thrown or anything?
Here are a few logs that may help in someone helping me =D

SIPDefault.cnf

# Image Version
image_version: "P0S3-08-9-00"

# Proxy Server
proxy1_address: "192.168.2.200"
proxy2_address: "192.168.2.200"
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""

# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:"5060"
proxy3_port:"5060"
proxy4_port:"5060"
proxy5_port:"5060"
proxy6_port:"5060"

# Emergency Proxy info
proxy_emergency: "192.168.2.200"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "192.168.2.200"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"

# NAT/Firewall Traversal
nat_enable: "0"
nat_address: "192.168.2.200"
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "1" 

# 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: "g711ulaw"

# 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: "1" ; 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"

#********* Release 2 new config parameters **********

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./sip/"

# Time Server
sntp_mode: "unicast"
sntp_server: "192.168.2.200"
time_zone: "GMT"
dst_offset: "1"
dst_start_month: "March"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "4"
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: "0"

#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"

# URL for external Phone Services
services_url: "http://192.168.2.200/cisco/services/index_cisco.php"

# URL for external Directory location
directory_url: "http://192.168.2.200/cisco/services/PhoneDirectory.php"

# URL for branding logo
logo_url: ""

SIP0016C7B74967.cnf

# SIP Configuration Generic File 
 
# Line 1 appearance
line1_name: 200 

# Line 1 Registration Authentication 
line1_authname: "200"

# Line 1 Registration Password
line1_password: "200"



####### New Parameters added in Release 2.0 #######

# All user_parameters have been removed

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: ""	; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "200"

# Line 2 Display Name (Display name to use for SIP messaging)



####### New Parameters added in Release 3.0 ######

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt:   "SIP Phone"      ; Limited to 15 characters (Default - SIP Phone) 

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco2" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none 

Telnet

SIP Phone> erase protflash
[12:07:14:108209] Parse error: string too big (9,12)
[12:07:14:108209] Default error: Name: [DSPLoadID] Value: [4.0(2.0)[A0]] rc:1
[12:07:14:108210] %W350 unprovisioned proxy_backup
[12:07:14:108211] %W351 unprovisioned proxy_emergency
[12:07:14:108211] %W362 No Valid Line Names Provisioned
[12:07:14:108221] Protocol Platform Area has changed
[12:07:14:108222] Programming Flash Config ROM
[12:07:14:108224] cprCancelTimer - NULL pointer passed in.
[12:07:14:108224] LINE 0/0: sip_platform_unregistration_timer_stop: Error: cprCancelTimer returned error.
SIP Phone> [12:07:30:109892] TFTP: Request file:SIP0016C7B74967.cnf from: <192.168.2.100>
[12:07:30:109894] TFTP: File received successfully!
[12:07:30:109894] %W350 unprovisioned proxy_backup
[12:07:30:109895] %W351 unprovisioned proxy_emergency
[12:07:30:109895] %W362 No Valid Line Names Provisioned
[12:07:31:109906] upgrade_check(P0S3-08-9-00)
[12:07:31:109972] %W350 unprovisioned proxy_backup
[12:07:31:109972] %W351 unprovisioned proxy_emergency
[12:07:31:109973] %W362 No Valid Line Names Provisioned


jingxi02
Posts: 105
Member Since:
2007-05-19
It's what is telling you

The error is indecating your line1_name is able to provision. In your SIP0016C7B74967.cnf file, line1_name: 200 should be line1_name: "200". Missing quotation MARKS.



pcbytc
Posts: 61
Member Since:
2007-12-04
Also - NO NAT!

I run lots of 7960's, the thing that always bugged me is after you setup the SIP extension in TB, you then need to go and immediately edit the extension you just made and change NAT to 0 (or no) or the phone vegitates and never registers.

Be nice if NAT=0 was the default, Kerry...

Timmy



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