Good Afternoon,
We use Anveo and VOIP.MS for our Cisco IP 7960 phones.
In order to use the phones, we have SIP & SEP files on one of our servers.
We just recently found out that external callers are not getting forwarded to a persons cell phone when they call their extension.
Here's our sipdefault.cnf file:
# Image Version
image_version: "P0S3-8-12-00"
# Proxy Server
proxy2_address: "sip.anveo.com"
# Proxy Server Port (default - 5060)
proxy2_port: "5010"
proxy1_address: "chicago3.voip.ms"
# Proxy Server Port (default - 5060)
proxy1_port: "5060"
# Emergency Proxy info
proxy_emergency: "chicago3.voip.ms"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "chicago3.voip.ms"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: "unprovisioned"
outbound_proxy_port: "0"
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: ""
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "120"
# 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: "0" ; 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"
# 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: "11 ; Default 11
sip_invite_retx: "7" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "*97"
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "unicast"
sntp_server: "0.pool.ntp.org"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
dst_stop_time: "2"
dst_auto_adjust: "1"
# 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: "1"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "0"
# URL for external Phone Services
services_url: "" ; URL for external Phone Services
directory_url: "http://*****" ; URL for external Directory location
logo_url: "http://*****" ; URL for branding logo to be used on phone display
# Remote Party ID
remote_party_id: 0 ; 0-Disabled (default), 1-Enabled
************************************************************************
SIP File - User's Phone Configs:
# SIP Configuration Generic File
# Line 1 appearance
line2_displayname: "Kim"
line2_name : "316297****"
line2_authname: "316297****"
line2_shortname : "Anveo"
line2_password:"****"
line1_name: "178805_****"
line1_authname: "178805_****"
line1_displayname: "Kim"
line1_password: "****"
line1_shortname : "voip.ms"
start_media_port: 20910 ; Start RTP range for media (default - 16384)
end_media_port: 20919 ; End RTP range for media (default - 32766)
voip_control_port: 5151 ; UDP port used for SIP messages (default - 5060)
nat_received_processing: "1"
####### 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: "Kim" ; 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: "****" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
# Time Server
sntp_mode: "unicast"
sntp_server: "0.pool.ntp.org"
time_zone: "CST"
dst_offset: "1"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
dst_stop_time: "2"
dst_auto_adjust: "1"
************************************************************************
SEP File - Standard Phone Configs:
<deviceProtocol>SIP</deviceProtocol>
<loadInformation model="IP Phone 7940">P0S3-07-2-00</loadInformation>
<transportLayerProtocol>1</transportLayerProtocol>
</device>
