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  • Cisco 7962

    Has anyone been able to install Cisco 7962 phones on their Shoretel as SIP extensions? Any assistance would be helpful.

  • #2
    1. Is this phone supported by Shoretel?
    2. Can you get any other SIP phones to work? Or this the only SIP phone you are trying to get to work?

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    • #3
      Other Sip devices work correctly. That model Cisco phone is not shown as being supported. I wondered if anyone had succeded in making this model work.

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      • #4
        Ah OK. I have no experience with it, sorry. I was just wondering if you need help with setting up a SIP extension in general, or with this actual phone.

        Comment


        • #5
          I have a 7960 series phone here which works fine. Do you have your tftp server up and working? Try using a config like this:
          Code:
          # SIP Configuration Generic File
          
          # Proxy Server
          proxy1_address:"10.173.214.5"; Can be dotted IP or FQDN
          proxy2_address:"10.173.214.5"; Can be dotted IP or FQDN
          proxy3_address:"10.173.214.5"; Can be dotted IP or FQDN
          proxy4_address:"10.173.214.5"; Can be dotted IP or FQDN
          proxy5_address:"10.173.214.5"; Can be dotted IP or FQDN
          proxy6_address:"10.173.214.5"; 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: 3600
          
          # Codec for media stream (g711ulaw (default), g711alaw, g729a)
          preferred_codec: g711ulaw
          
          # TOS bits in media stream [0-5] (Default - 5)
          tos_media: 5
          
          # Out of band DTMF Settings
          dtmf_inband: 1
          
          #(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
          stimer_t1: 500; Default 500
          msectimer_t2: 4000; Default 4
          secsip_retx: 10; Default 10
          sip_invite_retx: 6; Default 6
          timer_invite_expires: 180 ; Default 180 sec
          
          ####### New Parameters added in Release 2.0 #######
          
          # Dialplan template (.xml format file relative to the TFTP root directory)
          dial_template: dialplan
          
          # Time Server
          #(There are multiple values and configurations refer to Admin Guide for Specifics)
          sntp_server: ""; SNTP Server IP Address
          sntp_mode: anycast (default); unicast, multicast, or directedbroadcast
          time_zone: EST; 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 (default), 1-on, 2-off with no user control, 3-on with no user control)
          dnd_control: 0;
          
          # Caller ID Blocking
          #(0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
          callerid_blocking: 0; (Default is 0 - disabled and 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 is 0 - disabled and 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: 5060; 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: 1; 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: 1; 0-Disabled (default), 1-Enabled, 2-Privileged
          
          ####### New Parameters added in Release 4.0 #######
          
          #
          messages_uri: "102";
          
          # XML URLs
          # services_url: "http://10.173.214.2/Cisco/ShoreTelServices.xml"; URL for external Phone Services
          # directory_url: "http://10.173.214.2/Cisco/ShoreTelDirectory.xml"; URL for external Directory location
          logo_url: "http://10.173.214.2/shoretel_logo.bmp"; 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
          
          # Line 1
          line1_name: "Call"
          line1_displayname: "Cisco User"
          line1_shortname: "Call"
          line1_authname: "CUser"
          line1_password: "changeme"
          
          # Line 2
          line2_name: "Call"
          line2_displayname: "Cisco User"
          line2_shortname: "Call"
          line2_authname: "CUser"
          line2_password: "changeme"
          
          # Line 3
          line3_name: "Call"
          line3_displayname: "Cisco User"
          line3_shortname: "Call"
          line3_authname: "CUser"
          line3_password: "changeme"
          
          # Line 4
          line4_name: "Call"
          line4_displayname: "Cisco User"
          line4_shortname: "Call"
          line4_authname: "CUser"
          line4_password: "changeme"
          
          # Line 5
          line5_name: "Call"
          line5_displayname: "Cisco User"
          line5_shortname: "Call"
          line5_authname: "CUser"
          line5_password: "changeme"
          
          # Line 6
          line6_name: "Call"
          line6_displayname: "Cisco User"
          line6_shortname: "Call"
          line6_authname: "CUser"
          line6_password: "changeme"
          
          ####### New Parameters added in Release 2.0 #######
          
          # Phone Label (Text desired to be displayed in upper right corner)
          phone_label: "Cisco User - 126 - ShoreTel"; Has no effect on 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: "cisco"; Limited to 31 characters (Default - cisco)
          
          # User classification used when Registering [ none (default), phone, ip ]
          user_info: none

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          • #6
            Hi Palitto

            The codes will save as which format? And how sent to cisco phone?
            Can you tell me the step details?

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