Michael J. Tubby B.Sc. wrote:
Gents,
Following from a previous posting I've switched my 7060G at home from SCCP/Skinny to SIP but am stuck as I can't get it to register with Asterisk.
Mike,
This is from my configuration with a working 7960 under Asterisk 1.05, I have a 7940 under 1.03 with this same config, hope this helps:
[SIPDefault.cnf]
# SIP Default Generic Configuration File
# Image Version image_version: P003-07-3-00 image_version: P0S3-07-3-00
# Proxy Server proxy1_address: "192.168.100.55" ; Can be dotted IP or FQDN proxy2_address: "" ; Can be dotted IP or FQDN proxy3_address: "" ; Can be dotted IP or FQDN proxy4_address: "" ; Can be dotted IP or FQDN proxy5_address: "" ; Can be dotted IP or FQDN proxy6_address: "" ; 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: 900
# Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw
# 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: "./phone_configs/" ; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.168.100.55" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
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: 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: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged
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: 2 ; 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://192.168.100.15/icons/epi.bw.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-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off)
####### New Parameters added in Release 6.0 #######
# Dialtone Stutter for MWI stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled
# RTP Call Statistics (SIP BYE/200 OK message exchange) call_stats: 0 ; 0-Disabled (default), 1-Enabled
[sip.conf]
[4840] ; Line 1 type = friend host = dynamic auth=md5 username=4840 qualify=300 reinvite=no canreinvite=no nat=yes dtmfmode = rfc2833 context = sip mailbox = [EMAIL PROTECTED] secret=12345 disallow=all allow=ulaw allow=alaw callerid = 7960 #1 <4840>
[4841]; Line 2 type = friend host = dynamic auth=md5 username=4841 qualify=300 reinvite=no canreinvite=no nat=yes dtmfmode = rfc2833 mailbox = [EMAIL PROTECTED] context = sip secret=12345 disallow=all allow=ulaw allow=alaw callerid = 7960 #2 <4840>
[extensions.conf]
; (7960) - Test Line 1
exten => 4840,1,Macro(sip.extensions,${EXTEN},${EXTEN})
exten => 4840,2,Hangup(); (7960) - Test Line 2
exten => 4841,1,Macro(sip.extensions,${EXTEN},${EXTEN})
exten => 4841,2,Hangup()** My Macro section **
exten => s,1,NoOp(Dialing target ${ARG1} with rollover to voicemail ${ARG2})
exten => s,2,SetMusicOnHold(epi-cd)
exten => s,3,Dial(SIP/${ARG1},28,${VMOptions})
exten => s,4,Voicemail(u${ARG2})
exten => s,5,Hangup()
exten => s,106,Voicemail(b${ARG2})
exten => s,107,Hangup
_______________________________________________ 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
![http://192.168.100.15/icons/epi.bw.bmp"](http://192.168.100.15/icons/epi.bw.bmp"){kind=link}