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I set this up manually on the phone and it works just fine so config
files ... I attached the complete config files so maybe someone can
see what I am missing. ============ argon:/tftpboot# cat SIPDefault.cnf # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 ; # Proxy Server proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy5_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy6_address: "192.168.17.13" ; 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: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # 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 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 # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "192.168.17.11" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; 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: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) # 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-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable 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 ####### # XML URLs services_url: "" ; URL for external Phone Services directory_url: "" ; URL for external Directory location logo_url: "http://192.168.17.11/asterisk-tux.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 ####### New Parameters added in Release 4.4 ####### # Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) call_hold_ringback: 0 ; Default 0 (Disable ringback of held call) ======================== argon:/tftpboot# cat SIP00115C407FA3.cnf # SIP Configuration Generic File # Line 1 line1_name: Scott line1_authname: "scott" line1_password: "scott" # Line 2 line2_name: Scott1 line2_authname: "scott1" line2_password: "scott1" # Line 3 line2_name: "Line 2" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 4 line2_name: "Line 4" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 5 line2_name: "Line 5" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 6 line2_name: "Line 6" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" ####### 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: "User ID" # Line 2 Display Name (Display name to use for SIP messaging) line2_displayname: "User ID" ####### 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 classifcation used when Registering [ none(default), phone, ip ] user_info: none messages_uri: "_6101" argon:/tftpboot# Nabeel Jafferali wrote:
-- Scott Henderson ============================================================================ Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ============================================================================ |
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