Cisco 7940 Registration Failing

Discussion in '3CX Phone System - General' started by sunstate, Sep 9, 2010.

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  1. sunstate

    sunstate Member

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    The phone is successfully grabbing the config files from the TFTP server and is coming up with all the correct info, but it won't register. 3CX says that the authentication failed because the credentials don't match. But you can see from the phone config and the extension config that they do. Any ideas?

    Code:
    [CM102001]: Authentication failed for SipReq:  REGISTER 172.24.32.5 tid=5a397fa5 cseq=REGISTER contact=3CX@172.24.32.199:5061 / 122 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
    Code:
    # SIP Configuration Generic File
    
    # Image Version
    image_version: "P0S3-07-5-00"
    
    # Line 1 appearance 
    line1_displayname: "3CX Test" 
    line1_shortname:"150" 
    line1_name: 3CX Test 
    line1_authname: "150" 
    line1_password: "9876543210" 
    
    # Line 2 appearance 
    line2_displayname: "" 
    line2_shortname: "" 
    line2_name: UNPROVISIONED 
    line2_authname: "UNPROVISIONED" 
    line2_password: "UNPROVISIONED" 
    
    ####### New Parameters added in Release 2.0 #######
    # Phone Label (Text desired to be displayed in upper right corner)
    phone_label: "3CX Test"; Has no effect on SIP messaging
    
    # Line 1 Display Name (Display name to use for SIP messaging)
    line1_displayname: "3CX Test"
    
    # Line 2 Display Name (Display name to use for SIP messaging)
    line2_displayname: ""
    
    ####### 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: "cisco"; Limited to 31 characters (Default - cisco)
    
    # User classification used when Registering [ none (default), phone, ip ]
    user_info: phone
    [​IMG]
     
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  2. leejor

    leejor Well-Known Member

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    There have been a number of past posts on that particular model. Going by past responses, they don't tend to be looks upon favourably. One of the suggestions, I've seen, to fix registration problems, is to be sure that NAT is turned off.

    What firmware version are you running, apparently some have severe bugs?
     
  3. sunstate

    sunstate Member

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    NAT is disabled. Here is the SIPDefault.cnf

    Code:
    # Image Version
    image_version: "P0S3-07-5-00"
    
    # Proxy Server
    proxy1_address: "172.24.32.5"
    
    # Proxy Server Port (default - 5060)
    proxy1_port:"5060"
    
    # Emergency Proxy info
    proxy_emergency: "172.24.32.5" # IP address here alternatively
    proxy_emergency_port: "5060"
    
    # Backup Proxy info
    proxy_backup: "172.24.32.5"
    proxy_backup_port: "5060"
    
    # Outbound Proxy info
    outbound_proxy: ""
    outbound_proxy_port: "5060"
    
    # NAT/Firewall Traversal
    nat_enable: "0"
    nat_address: "172.24.32.5"
    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: "180"
    
    # 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: "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" ~np~# 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: "ntp2.usno.navy.mil" # IP address here alternatively
    time_zone: "MST"
    dst_offset: "0"
    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: "0"
    
    # 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: "0"
    
    # URL for external Phone Services
    services_url: "" # IP address here alternatively
    
    # URL for external Directory location
    directory_url: "" # IP address here alternatively
    
    # URL for branding logo
    logo_url: "" # IP address here alternatively
    
    # Remote Party ID
    remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
    
    I've read 10 or 15 posts regarding these phones, but none of them had this same problem. Everyone else had problems getting the phones flashed or getting it to pull the configs. Right now it's flashed with P0S3-07-5-00. That's all I could get my hands on. I'm just going to buy a smartnet contract for it so I can download the latest (P0S3-08-9-00). Hopefully that helps.
     
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  4. sunstate

    sunstate Member

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    Ok, it's working now. I've got it running the latest SIP firmware which is actually 8.12 (P0S3-8-12-00), not 8.9. I believe the problem was in the line1 parameters. I had "3CX Test" set for line1_name. Once I changed that to 150, it started working. Below are my working config files in case any one else needs them. Everything seems to be working perfectly (MWI, DTMF, transfer, conference, directory, CFwd, Phone Status, etc).

    SIPxxxxxxxxxxxx.cnf
    Code:
    # SIP Configuration Generic File
    
    # Image Version
    #image_version: "P0S3-8-12-00"
    
    # Line 1 appearance
    line1_displayname: "150"
    line1_authname: "150"
    line1_shortname: "3CX Test"
    line1_name: "150"
    line1_password: "150"
    
    # Line 2 appearance 
    #line2_displayname: "" 
    #line2_shortname: "" 
    #line2_name: UNPROVISIONED 
    #line2_authname: "UNPROVISIONED" 
    #line2_password: "UNPROVISIONED" 
    
    ####### New Parameters added in Release 2.0 #######
    # Phone Label (Text desired to be displayed in upper right corner)
    #phone_label: "3CX Test"; Has no effect on SIP messaging
    
    # Line 1 Display Name (Display name to use for SIP messaging)
    #line1_displayname: "3CX Test"
    
    # Line 2 Display Name (Display name to use for SIP messaging)
    #line2_displayname: ""
    
    ####### 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: "cisco"; Limited to 31 characters (Default - cisco)
    
    # User classification used when Registering [ none (default), phone, ip ]
    user_info: phone
    SIPDefault.cnf
    Code:
    # Image Version
    image_version: "P0S3-8-12-00"
    
    # Proxy Server
    proxy1_address: "172.24.32.5"
    
    # Proxy Server Port (default - 5060)
    proxy1_port:"5060"
    
    # Emergency Proxy info
    proxy_emergency: "172.24.32.5" # IP address here alternatively
    proxy_emergency_port: "5060"
    
    # Backup Proxy info
    proxy_backup: "172.24.32.5"
    proxy_backup_port: "5060"
    
    # Outbound Proxy info
    outbound_proxy: ""
    outbound_proxy_port: "5060"
    
    # NAT/Firewall Traversal
    nat_enable: "0"
    nat_address: "172.24.32.5"
    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: "180"
    
    # 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: "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" ~np~# 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: "ntp2.usno.navy.mil" # IP address here alternatively
    time_zone: "MST"
    dst_offset: "0"
    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: "0"
    
    # 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: "0"
    
    # URL for external Phone Services
    services_url: "" # IP address here alternatively
    
    # URL for external Directory location
    directory_url: "http://172.24.32.5:5481/provisioning/cisco_phonebook.xml" # IP address here alternatively
    
    # URL for branding logo
    logo_url: "" # IP address here alternatively
    
    # Remote Party ID
    remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
    
     
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