FREQUENTLY
ASKED QUESTIONS:
Q:
What are the differences between BudgeTone-101, 102 and 102D models?
Q:
Can I call another IP phone using its IP address directly without a proxy?
Q:
Where to get the latest software release for BudgeTone 100 series SIP phones?
Q:
How do I upgrade my BT100 series phone's firmware?
Q:
How do I setup my tftp server on the phone?
Q:
Why is my phone's LCD keep lighting up without showing any date?
Q:
Why is my phone showing date "1900-01-02"?
Q:
How to reset my BudgeTone 100 phone to factory default setting?
Q:
What is Outbound proxy? Should I put an Outbound proxy in the field?
Q:
What is the difference between "User ID" and "Authentication ID"?
Q:
What if my SIP URI domain is different from the SIP proxy server FQDN (Fully
Qualified Domain Name)?
Q
What Codec should I use for my IP phone?
Q:
What number should I use for "Voice Frames per TX"?
Q:
What is "Early Dial"? Should I use it?
Q:
What is STUN? Should I use STUN?
Q:
Do I still need to put in "Outbound" proxy if my phone is working under
STUN?
Q:
Which other 3rd party SIP applications and products are BudgeTone SIP phones
compatible with?
Q:
Do you have other colors than the white one?
Q:
Which SIP based IP telephone service providers do you currently support?
Q:
How do I setup my IP Phone for Delta3/iconnect network?
Q:
How do I setup my IP Phone for nikotel network?
Q:
How do I setup my IP Phone for MCI(test) network?
Q:
How do I setup my IP Phone for telic.net network?
Q:
How do I setup my IP Phone for go2call network?
Q:
How do I setup my IP Phone for FWD service?
Response
Code List
Q:
What are the differences between BudgeTone-101, 102 and 102D models?
A: Within
the BudgeTone-100 series family, model 101 and 102 have the same software
functions and the only difference is that model 101 has 1 (one) Ethernet
interface and model 102 has 2 (two) Ethernet interfaces.
Model 102D
will have a better character based LCD, 2 (two) Ethernet interfaces, and
more software functions such as 3-way conferencing, support for SIMPLE,
support for more vocoders, future support for power-over-Ethernet, ear
phone interface, etc.
Q:
Can I call another IP phone using its IP address directly without a proxy?
A:Yes.
Direct IP-to-IP calling is supported. Please refer to Users Manual for
details.
Q:Where
to get the latest software release for BudgeTone 100 series SIP phones?
A: If you
uses Ovislink IP phone service (P2P mode), you alway gets the latest firmware
automatically.
Q:
How do I upgrade my BT100 series phone's firmware?
A: First
configure your tftp server on the phone and then power cycle or reboot
the phone. Please consult user manual for detail.
Q:
How do I setup my tftp server on the phone?
A: There
are 2 ways setting up your tftp server.
1) From
the phone's web page.
2) From
phone keypad, press menu button and down arrow to item number 6, press
menu button one more time to get into the "Editing" mode. If the tftp server
IP displayed is not the one you want, enter the entire 12 digits of the
IP address of your tftp server. e.g., if your tftp server is 192.168.1.100,
enter 192168001100.
Q:
Why is my phone's LCD keep lighting up without showing any date?
A: The
most possible problem will be that the phone is not getting responses from
the NTP server. Check your network connection, DNS server or try to use
another NTP server.
Q:
Why is my phone showing date "1900-01-02"?
A: This
is probably because either DNS is not resolving the NTP server correctly,
or the NTP server is not responding. This 1900-01-02 is the default date
shown under this circumstance.
Q:
How to reset my IP phone to factory default setting?
A:1. set
up the phone to have NO tftp server and then boot without network
2. Press
the Menu and then the upward arrow key, you will see the Reset option on
the LCD. At this point, enter the full 12 digit of the MAC address or Product
ID of your phone (printed at the back of your phone). e.g., for "A/B/C",
press the "2" button until you see "A" or "B" or "C" appears, for "D/E/F",
press the "3" button until you see the corresponding character. Then continue
to enter the complete 12 digit of the MAC address. After that, press "Menu"
again.
Q:
What is Outbound proxy? Should I put an Outbound proxy in the field?
A: An
Outbound proxy is mostly used in presence of a firewall/NAT to handle the
signaling and media traffic across the firewall. Generally, if you have
an outbound proxy and you are not using STUN or other firewall/NAT traversal
mechanisms, you can use it. However, if you are using STUN or other firewall/MAT
traversal tools, do not use an outbound proxy at the same time.
Q:
What is the difference between "User ID" and "Authentication ID"?
A: User
ID is the user part of the SIP address of the phone and this is usually
the information displayed as Caller ID on the LCD. e.g., typically it is
a phone number or extension number or a user's name. Authentication ID
is an ID used strictly for authentication purpose when the phone attempts
to contact the SIP server. This may or may not be the same as User ID.
Q:
What if my SIP URI domain is different from the SIP proxy server FQDN (Fully
Qualified Domain Name)?
A: With
firmware 1.0.3.60 and later, you can put the your SIP URI domain name into
the SIP Server field, and put the actual sip server FQDN into Outbound
Proxy field. The phone will use the domain name in SIP Server as
part of SIP URI but send and receive SIP messages through the SIP proxy
server defined in the Outbound Proxy field.
Q
What Codec should I use for my IP phone?
A: By default,
PCMU(G711u) will be used. Both PCMU and PCMA will give you toll quality
but their bandwidth consumption is also the highest (64kbps). If your network
bandwidth is low, you can choose other lower-bit-rate codec such as G723
or G729 which will give you near toll quality at much smaller bandwidth
consumption (G723 consumes 5.3/6.3kbps and G729 consumes 8kbps). If bandwidth
is not a concern and you want good voice quality, try using PCMU or PCMA,
or even the new wide band codec G722 (64kbps) which will provide hi-fidelity
voice that is better than toll quality.
Q:
What number should I use for "Voice Frames per TX"?
It depends
on what codec you choose and balance between bandwidth utilization and
impact of packet loss. The bigger this value, the higher bandwidth utilization
because more voice frames are packed into the payload field of a UDP/RTP
packet and thus the network header overhead would be lower. However, the
impact of a packet loss on perceived voice quality will be bigger.
For PCMU/PCMA,
the default is 2 and max is 10
For G723,
the default is 1 and max is 32
For G726-32,
the default is 2 and max is 20
For G729,
the default is 2 and max is 64
For G728,
the default is 4 and max is 64
Q:
What is ¡°Early Dial¡±? Should I use it?
A: When
you dial a number, if you do not press the "Dial" ( "Redial") or "#" key
if it is configured to function as the "Send" key at the end of your dialed
string, the phone will wait for about 4 seconds before timeout and then
sends the actual INVITE message. If you set "Early Dial" to be YES, then
the phone will attempt to send out INVITE at each key input using the entered
dial string collected so far. If the SIP server supports 484 Incomplete
Address response, the phone will keep trying with each new key entry until
the complete dialed string is entered. This will essentially eliminate
the 4-second wait time mentioned above.
Please
note that this option can be used ONLY when the SIP server supports 484
Incomplete Address response. Otherwise, any other negative responses from
the SIP server (such as 404 Not Found) will cause immediate termination
of the call.
Q:
What is STUN? Should I use STUN?
A: STUN
stands for Simple Traversal of UDP over NAT. It is a protocol which enables
an IP phone to detect the presence and type of NAT behind which the phone
is placed. An IP phone that supports STUN can intelligently modify the
private IP address and port in its SIP/SDP message by using the NAT mapped
public IP address and port through a series of STUN queries against a STUN
server located on the public Internet. This will allow SIP signaling and
RTP media to successfully traverse a NAT without requiring any configuration
changes on the NAT.
STUN presents
a working solution for most NATs that are not symmetric NAT, e.g., most
of the SOHO routers have non-symmatric NAT and in this case, it is OK to
use STUN. However, STUN does NOT work with symmetric NAT and if your routers
have built-in symmetric NAT, do not use STUN.
Note: NOT
ALL SIP PROXY SERVER WILL WORK with A STUN TRANSLATED SIP MESSAGES, PLEASE
CONSULT YOUR SERVICE PROVIDER FOR DETAIL.
Q:
Do I still need to put in "Outbound" proxy if my phone is working under
STUN?
A: NO.
Q:
Which other 3rd party SIP applications and products are BudgeTone SIP phones
compatible with?
A: We have
been active participants in the SIPit events and have done extensive tests
successfully with a number of 3rd party SIP products directly or indirectly
through our customers or partners. A partial list of other products with
which we have successfully tested basic and in some cases advanced features
include:
-
Cisco (7960/7905 IP phones,
ATA186, SIP proxy, 5300/3640, etc)
-
Microsoft (Messenger, RTC Server)
-
DynamicSoft (SIP proxy)
-
Broadsoft (softswitch)
-
Santera/Tekelec (softswitch)
-
Siemens (IP phone)
-
Nortel (softswitch, softphone)
-
Intel (gateway)
-
Lucent (media server)
-
Alcatel
-
Jasomi (border controller)
-
iptel/SER (open source SIP proxy)
-
Digium/Asterisk (open source
IP PBX)
-
vovida.org/VOCAL (open source
IP PBX)
-
Mitel (IP phone)
-
Pingtel (IP phone, softphone)
-
Teledex (IP phone)
-
Dlink (IP phone)
-
Ingate/Intertex (SIP aware firewall)
-
Hotsip (SIP proxy, STUN server)
-
Radvision
-
Sylantro
-
Hughes Software Systems
-
Avaya
-
Ericsson
-
Nokia
-
Sharp
-
TI/Telogy
-
eDial (conferencing server)
-
octave (conferencing server)
-
Cosmocom (contact center application)
-
IVR Technologies (software based
IVR system)
-
SoftFront (SIP proxy, UA)
-
Vegastream (gateway)
-
UTStarcom (SIP proxy, media
gateway)
-
Tangerine (SIP proxy/registrar)
and many
others...
Q:
Do you have other colors than the white one?
A: We will
have another dark gray color option in the near future.
Q:
Which SIP based IP telephone service providers do you currently support?
A: We have
tested with and support the following service providers' SIP network:
MCI
Nikotel
Delta3
Telic.net
Go2Call
Free World
Dialup
This list
will continue to expand and please check for updates from time to time.
Q:
How do I setup my IP Phone for Delta3/iconnect network?
A: typical
configuration is:
SIP
Server: natrelay.deltathree.com outbound proxy: leave it blank User ID:
xxxxxx (your Delta3 account number) Authentication/Login ID: xxxxx (same
as above, your Delta3 account number) Password: xxxxx (your Delta3 password)
Dial
plan: 6666
Q:
How do I setup my IP Phone for nikotel network?
A: typical
configuration is:
SIP
Server: calamar0.nikotel.com
Outbound
proxy: leave it blank
User
ID: xxxxx (your nikotel account number)
Authentication
ID: same as your User ID
Password:
your nikotel password
NAT
Traversal: YES (WITHOUT setting the STUN server)
Q:
How do I setup my IP Phone for MCI(test) network?
A: typical
configuration is:
SIP
Server: siptest.mci.com
Outbound
proxy: (use an outbound proxy if MCI provides one for you)
User
ID: xxxxx (your MCI assigned account/phone number)
Authentication
ID: (Your MCI assigned id, i.e., foo)
Password:
your MCI password
NAT
Traversal: No (You need to set up your STUN server if you don't have
outbound proxy)
Note: MCI
Proxy server seems to respond our phone client SIP messages correctly.
Q:
How do I setup my IP Phone for telic.net network?
A: typical
configuration is:
SIP
Server: sip.telic.net
Outbound
proxy: (Use outbound proxy, it will not work under STUN for now)
User
ID: xxxxx (your Telic.net account number)
Authentication
ID: same as your User ID
Password:
your Telic.net password
Note: STUN
is not working yet against Telic.net's SIP proxy server for now.
Q:
How do I setup my IP Phone for go2call network?
A: typical
configuration is:
SIP
Server: voip01.go2call.com
Outbound
proxy: (Should leave it blank, because it's a GW)
User
ID: xxxxx (your Go2Call PIN number)
Authentication
ID: same as your User ID
Password:
xxxxxxx (Your Go2Call password)
NAT
Traversal: YES (WITHOUT setting the STUN server)
Q:
How do I setup my IP Phone for FWD service?
A: typical
configuration is:
SIP
Server: fwd.pulver.com
Outbound
proxy: 192.246.69.247:5082 (used only when behind firewall, otherwise
leave it blank)
User
ID: xxxxxx (your FWD account number)
Authentication/Login
ID: xxxxx (same as above, your FWD account number)
Password:
xxxxx (your FWD password)
NAT
Traversal: No (You need to set up your STUN server if you don't have
outbound proxy)