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IP Phones NOt registering Remotely in different location

Hariram
Posts: 21
 Member Since:
2008-09-04
Submitted by Hariram on Sat, 11/15/2008 - 9:33pm.

Hi All,

In my local network all my Phones are up and running. I opened all the needed Port in my router/modem Ports TCP/UDP

69,
10000 - 20000,
4059,
5004-5082

Inside my company with different network it is registering without any issue and Phones are registering over the WAN without any issue. I just point the tftp server to my Public IP.

But when ever i try to register the IP phones in other location. it is not registering

1) Sometime SIPdefault.cnf and other are timeout. i understand this is internet connection issue. If i am wrong please correct me.
2) Mostly i am getting 'X' mark near to the extension.

I configured NAT to 'no' with the same configuration all my IP phones are registering through wan without any issue inside my company i tried with three different network. Please advise if any thing to check in my remote locations... May be bandwidth issue or packet drop i am not sure.

Thanks,

HARIRAM

‹ is there a tutorial on configuring a general sip devide with trixbox Nokia E51 wifi phone hangs on receiving calls ›

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locopedro
Posts: 20
 Member Since:
2007-02-21
I take it you have a cisco ip phone
Mon, 11/17/2008 - 8:34pm

If your trying to register cisco phones
make this changes in your SIPdefault.cnf

*****************SIPdefault.cnf******************

# Image Version
image_version: "P0S3-08-8-00"

# Proxy Server
proxy1_address: "External.IP.Address"

# Proxy Server Port (default - 5060)
proxy1_port:"5060"

# Emergency Proxy info
proxy_emergency: "External.IP.Address"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "External.IP.Address"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: "proxy_backup: "External.IP.Address"
outbound_proxy_port: "5060"

#very important that you set nat to enabled and place your external ip address there
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "External.IP.Address"
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"

# 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, g729)
preferred_codec: "g729"

# 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: "2" ; 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: "10" ; Default 11
sip_invite_retx: "6" ; 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: "External.IP.Address"
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: "http://External.IP.Address/xmlservices/index.php"

# URL for external Directory location
directory_url: "http://External.IP.Address/xmlservices/PhoneDirectory.php"

# URL for branding logo

logo_url: "http://External.IP.Address/cisco/bmp/trixbox.bmp"

# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled

*****************End of SIPdefault.cnf******************

That Should work on any 7960/7940 Cisco Phone.


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locopedro
Posts: 20
 Member Since:
2007-02-21
On another note
Mon, 11/17/2008 - 8:36pm

You might want to open port 123 so the phones can connect to the time server.


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16again
Posts: 370
 Member Since:
2007-03-04
Best to never use tftp on
Mon, 11/17/2008 - 11:47pm

Best to never use tftp on the internet!
Consider ftp http(s) or local config on phone itself

Why your setup doesn't work.......?
afaik, tftp uses a random port besides 69


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locopedro
Posts: 20
 Member Since:
2007-02-21
True TFTP only on Lan
Tue, 11/18/2008 - 2:51pm

yes, I agree TFTP should only be setup on LAN. You can use a tftp software like tftpd32 to provision the phones remotely.


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Hariram
Posts: 21
 Member Since:
2008-09-04
If i hard code the config on phone itself.
Tue, 11/18/2008 - 7:26pm

Hi All,

If i hard code the config on phone itself. what IP should i put on tftp server. Dont i need tftp server if i hard code the info. on remote Phones ?

Thanks,

HariRam


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atilio
Posts: 288
 Member Since:
2006-06-01
You wouldn't need one if you
Tue, 11/18/2008 - 7:58pm

You wouldn't need one if you manually configure them. It would take a while if you have lots of them. Now if there is a tftp server running where those remote phones are the phones might try to connect to it and get the configuration file from and getting unprovisioned. In the later case what I've done is manually enter an ip address other than that remote tftp server. So if that remote tftp server is 192.168.0.5. then I set it to 192.168.0.10 and manually configure the phone.


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dmanolis79
Posts: 16
 Member Since:
2007-07-09
All below information must
Wed, 11/26/2008 - 4:43pm

All below information must be in your SIPMACADDRESS.cnf File...

image_version: P0S3-08-10-00
# Cisco SIP Configuration

# Line 1 Settings
line1_name: "XXXX" ; Line 1 Extension\User ID
line1_displayname: "XXXX" ; Line 1 Display Name
line1_shortname: "XXXX"
line1_authname: "XXXX" ; Line 1 Registration Authentication
line1_password: "XXXX" ; Line 1 Registration Password

# Line 2 Settings
line2_name: "XXXX" ; Line 2 Extension\User ID
line2_displayname: "XXXX" ; Line 2 Display Name
line2_shortname: "XXXX"
line2_authname: "XXXX" ; Line 2 Registration Authentication
line2_password: "XXXX" ; Line 2 Registration Password

# Proxy
proxy1_address: "Trixbox External IP Address"
proxy1_port: "5060"

# Emergency Proxy info
proxy_emergency: "Trixbox External IP Address"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "Trixbox External IP Address"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: "Trixbox External IP Address"
outbound_proxy_port: "5060"

# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "your remote External IP"
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "1"

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Vitabyte" ; Has no effect on SIP messaging

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Phone prompt/password for telnet/console session
phone_prompt: "" ; Telnet/Console Prompt
phone_password: "" ; Telnet/Console Password

# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: phone

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "60"

# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "none"

# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"

# 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)

# 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 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec

# Setting for Message speeddial to UOne box
messages_uri: "*97"

# URL for external Directory location

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"

# Time Server
sntp_mode: "unicast"
sntp_server: "Trixbox External IP Address"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
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"

remote_party_id: 1 ; 0-Disabled (default), 1-Enabled

logo_url: "" ; URL for branding logo to be used on phone display

You do not need the SIPDefault.cnf File.

Now for TFTP use a different computer and port forward port 69 to that machine running TFTP software.

My 7960 works perfectly..

If you have any questions let me know.,


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