[Asterisk-Users] Help sought: Cant get Cisco 7960 to register
on Asterisk
Doug Lytle
support at drdos.info
Sat Feb 5 15:10:19 MST 2005
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 = 4840 at sip
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 = 4841 at sip
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
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