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


Member Since:
2008-08-14