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Call Forwarding from External Caller Not Working

kyle.koopman
Beginner
Beginner

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>

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