Finally Cisco 7960 SUCCESS!

Discussion in '3CX Phone System - General' started by tekknow, Oct 5, 2008.

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  1. tekknow

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    Here is the working config that I used to get my Cisco 7960 to register from behind a NAT to a 3CX system on a public IP

    Please note the trick seemed in my case to be in

    # NAT/Firewall Traversal
    nat_enable: "1"
    nat_address: ""
    voip_control_port: "5061"
    start_media_port: "16384"
    end_media_port: "32766"
    nat_received_processing: "1"

    The nat_received_processing can not be changed in the menus, and I suspect this was the stickler. O

    File Name SIPDefault.cnf

    # Image Version
    image_version: "P0S3-08-2-00"

    # Proxy Server
    proxy1_address: "sip.mydmain.com"

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


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

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

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

    # NAT/Firewall Traversal
    nat_enable: "1"
    nat_address: ""
    voip_control_port: "5061"
    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: "60"

    # 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: "999"

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

    # Time Server
    sntp_mode: "unicast"
    sntp_server: "148.234.7.30"
    time_zone: "CST"
    dst_offset: "1"
    dst_start_month: "Mar"
    dst_start_day: ""
    dst_start_day_of_week: "Sun"
    dst_start_week_of_month: "2"
    dst_start_time: "02"
    dst_stop_month: "Nov"
    dst_stop_day: ""
    dst_stop_day_of_week: "Sunday"
    dst_stop_week_of_month: "1"
    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: ""

    # URL for external Directory location
    directory_url: ""

    # URL for branding logo
    logo_url: ""

    # Remote Party ID
    remote_party_id: 1 ; 0-Disabled (default), 1-Enabled






    FIle name SIPXXXXXXXXXXXX.cnf (DEPENDENT ON MAC ADDRESS)
    # Cisco SIP Configuration

    phone_label: "WHatever you want to call your phone"
    line1_name: "200"
    line1_shortname: "200"
    line1_displayname: "200"
    line1_password: "XXXXXXX"
    line2_name: "201"
    line2_shortname: "201"
    line2_displayname: "201"
    line2_password: "XXXXXXX"
    line3_name: "202"
    line3_shortname: "202"
    line3_displayname: "202"
    line3_password: "XXXXXXX"
    line4_name: "203"
    line4_shortname: "203"
    line4_displayname: "203"
    line4_password: "XXXXXXX"
    line5_name: "UNPROVISIONED"
    line5_shortname: "UNPROVISIONED"
    line5_displayname: "UNPROVISIONED"
    line5_password: "UNPROVISIONED"
    line6_name: "UNPROVISIONED"
    line6_shortname: "UNPROVISIONED"
    line6_displayname: "UNPROVISIONED"
    line6_password: "UNPROVISIONED"
    line1_authname: "200"
    line2_authname: "201"
    line3_authname: "202"
    line4_authname: "203"
    line5_authname: "UNPROVISIONED"
    line6_authname: "UNPROVISIONED"
     
  2. smprr2

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    hi,
    i just upgrade firmware for cisco 7960 to work with SIP, but phone couldn't register with my SIP server, what i can't understand is how can cisco phone load it's configs, the firmware version we setup was P0S3-08-3-00, please somebody let us know what we have to, do we have to set up a TFTP server, how we gonna find config files to load phone
    help me

    thank you
     
  3. tuxiteer

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    This post is a few months old, but if you are still looking for a response... here one is :)

    The Cisco phones use TFTP (defined in your DHCP) to download a series of files to check for firmwares and configs. The default file names for my Cisco 7960 phones are:

    SIPDefault.cnf
    OS79XX.TXT
    P0S3-8-12-00.loads
    P0S3-8-12-00.sb2
    P003-8-12-00.bin
    P003-8-12-00.sbn
    SIPPHONEMACADDRESS.cnf
    XMLDefault.cnf.xml

    You would update the SIPDefault.cnf, XMLDefault.cnf.xml, and SIPPHONEMACADDRESS.cnf for your environment. You can also place ringtones in the same TFTP "root" folder to consume on your phones.

    If a Cisco phone goes to the menu screen (where you can dial, browse settings, etc) it may not have been able to load the .cnf files... it is just using it's last known config. A little trick I do to see if it is working is to change the title on the top of the config file with "Phone Name v2".
     
  4. odf

    odf

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    Hello,
    Know if this can work with a 7945 and 7975?
     
  5. KerryG

    KerryG Active Member

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    http://www.voipstore.com/configuring-cisco-7975-ip-phones-for-sip/
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  6. odf

    odf

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    Got something for 7945, or this procedure will work?
     
  7. connectica

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    CP-7960G worked perfectly as a remote extension of 3CX. Thanks for your help.
     
  8. miso

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    CP-7960G does not call blind transfer or transfer. I used the below config file. The phone is connected to the same LAN as the PBX Please help
     
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