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Re: Upgraded Firmware to SIP now UNPROVISIONED? Help :(

ipgflorida
Level 1
Level 1

Hi,

I am new to this form, and I am new to IP telephones. Currently I am using AXON pbx, but I don't have any service currently hooked up to the pbx.

I have a couple (18) CP-7960G phones, and I have successfully upgraded to SIP firmware: P0S3-8-12-00.

Now, when I go to network configuration it says for my callmanager, unprovisioned, it also says unprovisioned for when I go to SIP Configuration and Line 1 settings. My shortname and Display name say unprovisioned as well.

Here are my current STATUS MESSAGES:

W350 unprovisioned proxy_backup

W351 unprovisioned proxy_emergency

W310 1 Error(s) Parsing: SIPDefault.cnf

In my SIP(mac address).cnf file I have the following code:

; phone-specific configuration file sample

line1_name : Jamie

line1_authname : 125

line1_password : jG736@ne

In my SIPDefault.cnf I have the following code:

# SIP Default Generic Configuration File    # Image Version image_version: P0S3-8-12-00;  # Proxy Server proxy1_address: 192.168.1.131; sip_server_name proxy2_address: ""          ; Can be dotted IP or FQDN proxy3_address: ""          ; Can be dotted IP or FQDN proxy4_address: ""          ; Can be dotted IP or FQDN proxy5_address: ""          ; Can be dotted IP or FQDN proxy6_address: ""          ; Can be dotted IP or FQDN  # Proxy Server Port (default - 5060) proxy1_port: 5060 proxy2_port: 5060 proxy3_port: 5060 proxy4_port: 5060 proxy5_port: 5060 proxy6_port: 5060  # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1   # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 120   # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711alaw  # TOS bits in media stream [0-5] (Default - 5) tos_media: 5  # 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 10 sip_invite_retx: 6           ; Default 6 timer_invite_expires: 180      ; Default 180 sec  ####### New Parameters added in Release 2.0 ####### language:  messages_uri : ""  # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan  # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ""          ; Example:  ./   # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: ""               ; SNTP Server IP Address sntp_mode: directedbroadcast     ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: CET               ; Time Zone Phone is in dst_offset: 1               ; Offset from Phone's time when DST is in effect  dst_start_month: April          ; Month in which DST starts dst_start_day: ""          ; Day of month in which DST starts dst_start_day_of_week: Sun     ; Day of week in which DST starts dst_start_week_of_month: 1     ; Week of month in which DST starts dst_start_time: 02          ; Time of day in which DST starts dst_stop_month: Oct          ; Month in which DST stops dst_stop_day: ""          ; Day of month in which DST stops dst_stop_day_of_week: Sunday     ; Day of week in which DST stops dst_stop_week_of_month: 8     ; Week of month in which DST stops 8=last week of month dst_stop_time: 2          ; Time of day in which DST stops dst_auto_adjust: 1          ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 1          ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)  # 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-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0          ; Default 0 (Disable blocking of anonymous calls)  # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101          ; Default 101  # Sync value of the phone used for remote reset  sync: 1                    ; Default 1  ####### New Parameters added in Release 2.1 #######  # Backup Proxy Support proxy_backup: ""          ; Dotted IP of Backup Proxy proxy_backup_port: 5060          ; Backup Proxy port (default is 5060)  # Emergency Proxy Support proxy_emergency: ""           ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060     ; Emergency Proxy port (default is 5060)  # Configurable VAD option enable_vad: 0               ; VAD setting 0-disable (Default), 1-enable  ####### New Parameters added in Release 2.2 ######  # NAT/Firewall Traversal nat_enable: 0                   ; 0-Disabled (default), 1-Enabled nat_address: ""                  ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5063          ; UDP port used for SIP messages (default - 5060) start_media_port: 16384      ; Start RTP range for media (default - 16384) end_media_port: 32766        ; End RTP range for media (default - 32766) nat_received_processing: 0     ; 0-Disabled (default), 1-Enabled  # Outbound Proxy Support outbound_proxy: ""           ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060       ; default is 5060  ####### New Parameter added in Release 3.0 #######  # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable : 0          ; 0-Disabled, 1-Enabled (default)  ####### New Parameters added in Release 3.1 #######  # 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 the phone)  telnet_level: 2               ; 0-Disabled (default), 1-Enabled, 2-Privileged  ####### New Parameters added in Release 4.0 #######  # XML URLs services_url: ""          ; URL for external Phone Services directory_url: ""          ; URL for external Directory location logo_url: ""               ; URL for branding logo to be used on phone display  # HTTP Proxy Support http_proxy_addr: ""          ; Address of HTTP Proxy server http_proxy_port: 80          ; Port of HTTP Proxy Server (80-default)  # Dynamic DNS/TFTP Support dyn_dns_addr_1: ""              ; restricted to dotted IP dyn_dns_addr_2: ""              ; restricted to dotted IP dyn_tftp_addr: ""               ; restricted to dotted IP  # Remote Party ID remote_party_id: 0          ; 0-Disabled (default), 1-Enabled  ####### New Parameters added in Release 4.4 #######  # Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control) call_hold_ringback: 1          ; Default 0 (Call Hold Ringback feature is off)  ####### New Parameters added in Release 6.0 #######  # Dialtone Stutter for MWI  stutter_msg_waiting: 0          ; 0-Disabled (default), 1-Enabled  # RTP Call Statistics (SIP BYE/200 OK message exchange) call_stats: 0               ; 0-Disabled (default), 1-Enabled Can someone please tell me what is wrong with my phone system, and why it is telling me something is wrong?!?! I am using tftpd32 by John to transfer the information to the Cisco devices as I do NOT have Cisco Callmanager.  Your help is greatly appreciated!!!  Thank You  Jamie Greene

3 Replies 3

girardan66
Level 1
Level 1

Hi jamie

I've same pbe on my phone 7940 , do you have found solution ?

regards

Antoine

Attach the following files:  OS79XX.TXT, SIPDefault.conf and SIPmacaddress.cnf

Hi

thanks, I've solved this pbe, very late this morning, but i don't have "let go of the piece"...

I think pbe came from Firmeware name, I have a name as P0S3-08-12-00.loads and .sb2 with good reference in my config files OS79XX.TXT and other, when i've rename file on P0S3-8-12-00.loads and .sb2, with little adaptation config in cisco phone (not sure it was necessary), it was ok.

Best it's do a debug on telnet cisco :

telnet ciscophoneIP

SIP Phone>debug sip-messages

SIP Phone>show register

# run a SIP registration

SIP Phone> register 1 1

SIP Phone>debug sip-state sip-reg-state

Find  in attachment my config files.

Below my TFTP-ROOT with files just necessary :

TFTP-ROOT

MacBook-Air-de-Antoine:TFTP-Root antoine$ ls

OS79XX.TXT        caramba.raw        nyuknyuk.raw        ringer3

P0S3-8-12-00.loads    dalek.raw        ohno.raw        sax.raw

P0S3-8-12-00.sb2    dialplan.xml        piano1.raw        scottie.raw

RINGLIST.DAT        dive.raw        piano2.raw        sheep.raw

SEP00249734AD3B.cnf.xml    doh.pcm            ring1.pcm        synth1.pcm

SIP00249734AD3B.cnf    doorbell.raw        ring2.raw        synth2.pcm

SIPDefault.cnf        jamaica.raw        ring3.pcm        synth3.raw

XMLDefault.cnf.xml    klaxon.raw        ring4.pcm        synth4.raw

areyouthere.raw        lis.raw            ring5.pcm        synth5.raw

asleep.raw        lurch.raw        ring6.pcm        woody.raw

beepbeep.raw        mayihelp.raw        ringer2

MacBook-Air-de-Antoine:TFTP-Root antoine$

regards

Antoine Girard Resolance