Yes, this is frustrating I know. In fact the wiki could be updated to provide this info. Basically if you have the phones out of the box (brand spankin new) then you probly have the SCCP image installed on it by default. Your tftp server root will need a number of files to start if this is the case. Ok with that said, most of this I had to figure out on my own. Cisco's website as we all know is a pain in the a$$ to find any useful info on how to do anything.
Be sure to remove #comments before experimenting. ALSO, DO THIS WITH ONE PHONE AT A TIME. If you have other phones plugged in they WILL automatically try to upgrade :) .. [OS79XX.txt] P0S3-07-3-00 # This is the version we used, S stands for Sip, 7-3 stands for 7.3 .. If you need firmware you'll have to get them off Cisco's site, there was a posting recently stating where to obtain these without a cisco login. # once you have that file in place your SEP (yes, SEP) device will start looking for the file with that extension. It cuts off the file extension, for example in your tftp root you will need: P0S3-07-3-00.sb2 P0S3-07-3-00.loads #Once you have those 3 files your phones should start upgrading, be careful though. It's been known that older versions that come on the phones have bugs and can blow up (crash) if you try to put too large an image on them. # Moving on, once you get that completed your phone should boot and start looking for the following files. Before I post them below, take note on how this all works. First you have a general config file, SIPDefault.cnf .. This contains such things as your proxy address, logo, services, directories, ntp, that kind of stuff. The second is your SIP<MAC ADDRESS>.cnf .. This is per phone, that contains your phone line info, names, etc.. [sipdefault.cnf] # Image Version image_version: "P0S3-07-3-00" # Proxy Server proxy1_address: "192.168.1.17" # Proxy Server Port (default - 5061) #proxy1_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: "120" # 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: "1" ; 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: "8500" #********* Release 2 new config parameters ********** # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "./" # Time Server sntp_mode: "directedbroadcast" sntp_server: "17.254.0.49" time_zone: "CST" 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: "1" ####### New Parameters added in Release 4.0 ####### # XML URLs #services_url: "http://192.168.1.65/menu.pl"; ; URL for external Phone Services directory_url: "http://192.168.1.17/directories.xml"; # URL for external Directory location logo_url: "http://192.168.1.17/netlogic.bmp"; ; URL for branding logo to be used on phone display # put your own logo in the logo_url location; I include the 10-20.com one for reference in building your own # 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: "192.168.1.2" ; restricted to dotted IP dyn_dns_addr_2: "" ; restricted to dotted IP dyn_tftp_addr: "192.168.1.2" ; restricted to dotted IP # The dynamic tftp server should be set to whatever your TFTP server is. This way, it # keeps the tftp server setting even though you might be using DHCP (default behavior # is to use the DHCP server as a tftp server, which is rarely correct.) # Remote Party ID remote_party_id: 1 ; 0-Disabled (default), 1-Enabled # EOF --------------- #WHEW THAT WAS A LOT --------------- # Luckily the next file is a lot shorter, it should be self explanatory [SIP00059BB47680.cnf] image_version: P0S3-07-3-00 line1_name: 107 # Line 1 Registration Authentication line1_authname: "107" # Line 1 Registration Password line1_password: "LADEDA" ## See the pattern? For another line you would put line2_authname: "107" etc.. ####### 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: "Matt S 107" ; Has no effect on SIP messaging # Line 1 Display Name (Display name to use for SIP messaging) line1_displayname: "Matt S" ####### 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: "BLAHBLAH" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none -----Original Message----- From: Jose Cruz (Branders IT) [mailto:[EMAIL PROTECTED] But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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