Cortelco 274701-VIP-PAK
Cortelco 274701-VIP-PAK
Cortelco 274701-VIP-PAK
IP Phone
User Manual
274701-VIP-PAK
Cortelco
Release 5.1
Contents
1 OVERVIEW .........................................................................................................4
1.1 INTRODUCTION .............................................................................................4
1.2 KEY FEATURES..............................................................................................4
1.3 HARDWARE SPECIFICATION ...........................................................................7
4 WEB CONFIGURATION.................................................................................30
4.1 NETWORK CONFIGURATION ON WEB ...........................................................31
4.1.1 Basic.....................................................................................................31
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4.1.2 Others...................................................................................................31
4.1.2.1 MAC Address...................................................................................31
4.1.2.2 NTP Server.......................................................................................31
4.1.2.3 Time Zone ........................................................................................32
4.2 H.323 CONFIGURATION ON WEB .................................................................32
4.2.1 H.323 Parameters ................................................................................32
4.2.2 Forward Mode .....................................................................................32
4.2.3 Advanced..............................................................................................33
4.2.3.1 Min Media Port :..............................................................................33
4.2.3.2 Max Media Port : .............................................................................33
4.2.3.3 Codec : .............................................................................................33
4.3 SIP CONFIGURATION ON WEB ......................................................................33
4.3.1 SIP Parameters ....................................................................................34
4.3.2 Forward Mode .....................................................................................34
4.3.3 Advanced..............................................................................................34
4.3.3.1 Reg From .........................................................................................34
4.3.3.2 Reg To ..............................................................................................34
4.3.3.3 Reg Expire .......................................................................................34
4.3.3.4 Min Media Port ................................................................................34
4.3.3.5 Max Media Port ...............................................................................34
4.3.3.6 Codec ...............................................................................................34
4.4 PHONE CONFIGURATION ON WEB ................................................................35
4.4.1 Prefix....................................................................................................35
4.4.2 Voice.....................................................................................................35
4.4.2.1 Ring Volume ....................................................................................35
4.4.2.2 Handset Volume ...............................................................................36
4.4.2.3 Handsfree Volume............................................................................36
4.4.2.4 CodecTxGain ...................................................................................36
4.4.2.5 Ring Type.........................................................................................36
4.4.2.6 RTPLowBW.....................................................................................36
4.4.2.7 Jitter Buffering .................................................................................36
4.4.3 Others...................................................................................................36
4.4.3.1 VAD .................................................................................................36
4.4.3.2 BG Noise Level................................................................................37
4.5 SYSTEM CONFIGURATION............................................................................37
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5.3 CHECK CALL HISTORY (INCOMING / OUTGOING / MISSED CALLS) .................38
5.4 REDIAL .......................................................................................................39
5.5 CALL FORWARD ..........................................................................................39
5.6 CALL TRANSFER .........................................................................................39
5.6.1 Blind Transfer ......................................................................................39
5.6.2 Attended Transfer.................................................................................39
5.7 CALL HOLD.................................................................................................40
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1 Overview
1.1 Introduction
Cortelco Inc. and its suppliers believe that the next generation of networks
based on VoIP technologies will significantly change the way people communicate
with each other. To get the greatest benefit from these technologies, the use of VoIP
should be simple to most users. Our mission is to make your VoIP experience just
as straightforward as using a normal telephone set.
With the built in LCD display, the user can easily configure the VOIP 2747 for
first time installation in just a few minutes. In addition to the advanced VoIP
functions and easy installation, the VOIP 2747 also provides rich telephone features
such as last number redial, speed dial, call forward/transfer, call history, volume
adjustment and speakerphone. Cortelco’s VOIP 2747 is your best VoIP solution for
the next generation of communication.
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o Gets IP from DHCP server using DHCP protocol or through ADSL
modem using PPPoE protocol, automatically reconnects when PPPoE
loses connection.
• Pass through NAT devices:
o Can make outgoing and incoming calls under any NAT device (even
under two layer NAT devices) when working with the specific
gatekeeper/proxy device.
• Remote software upgrade capability (via ftp):
o FTP protocol provides reliable remote upgrade via Internet.
• Advanced Digital Signal Processing (DSP) technology to ensure superior
audio quality:
o Hardware System on a Chip solution with built in DSP processor
ensures perfect voice quality.
• Supports G.723.1, G.729A/B, G.711 (A-law/U-law) voice codecs :
o Follows ITU-T standard for best compatibility.
• Supports supplementary services, including immediate (unconditional) call
forwarding, busy call forwarding, no answer call forwarding and call
hold/transferring.
• Provides call history:
o Records incoming call history, outgoing call history, missed (not
accepted) call history, and lets users make a direct call from the call
history log.
• Phone Book : 50 entries
• Speed dial : 10 entries
• Supports Silence Suppression, VAD (Voice Activity Detection), CNG
(Comfort Noise Generation) :
o Silence suppression can save about half of the network bandwidth
needed during normal VoIP conversation.
• Ping function supported :
o Pings another device on the Internet from the VOIP 2747 to make sure
the Internet connection is ok.
• System status display on the LCD panel :
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o User can easily observe if the VOIP 2747 is working normally and
monitor the system status (network status, registering status) from the
LCD panel display.
o A “PKT Trace” function is supported to display the packets received
on the LCD panel and helps network administrator find the problem on
line.
• Call with or without gatekeeper / proxy(direct IP dialing) :
o Follows standard SIP / H.323 protocol and is compatible with most
existing SIP proxy / H.323 gatekeepers.
• Provides easy configuration methods:
o Setting by keypads on the phone.
o Setting by web browser.
• Supports RFC-3261, H.323, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP,
DNS, DHCP, NTP/SNTP, FTP, PPP, PPPoE protocols.
• Interoperable with most of the existing SIP/H.323 VoIP devices (IP-phone,
gateway, gatekeeper, proxy, soft-switch, IP-PBX), including Microsoft
NetMeeting, Cisco gateways / gatekeepers:
o Please refer to the section 6.2/6.3 Interoperability List for a complete
listing.
• The WAN Port works with either straight through or crossover Ethernet cable.
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1.3 Hardware Specification
HARDWARE SPEC
Spec\Model VOIP 2747
Input: 100-240V AC
Universal Switching Power Adaptor
Output: +7V DC, 800mA
Dimension 19cm(W)x23cm(D)x9cm(H)
Weight 870 g
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2 Basic Installation
1.
7.
2.
3.
6. 5. 4.
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2.1.2 Rear and Back side Panel of the VOIP 2747
The rear and back side panel illustration is shown in figures 2 and 3;
main parts include:
3. 2. 1.
Figure 2. Rear Panel of the IP Phone
4.
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2.1.3 Keypad Definition
1. Light
The red light blinks when there is an incoming call and provides Message
Waiting indication.
2. LCD Display
Menu and all status displays for user.
3. “Í UP” Key
When the IP phone is in the menu selection mode, this key is used to scroll
up the menu items.
When the IP phone is editing menu item contents, this key is used as “ left
delete” to delete a digit upon each key press.
When the IP phone is in dial mode, the “Å” key is used as “delete” key.
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“Æ DOWN” Key
When the IP phone is in the menu selection mode, this key is used to scroll
down the menu items.
When the IP phone is editing menu item contents, this key is used as “ right
shift” to shift the cursor right a digit for each key press.
4. - IN VOL+
When the IP phone is in an idle state or user is talking on handset or speaker,
this key is used to increase/decrease the volume of the incoming voice.
The volume of speaker, handset and ring are separately adjusted according to
the mode of current usage. When in idle, “+” key increases the volume of
the ring tone, and “-” key decreases the volume of the ring tone; in
hands-free mode, “+” key increases the speaker volume, and “-” key
decreases the speaker volume; when in handset mode, “+” key increases the
volume of handset, “-” key decrease the volume of handset.
5. +OUT VOL-
Users are able to increase and/or decrease the volume which they transmit
out to a remote party.
6. SPEAKER
This key is pressed to switch between the using the handset and the speaker.
7. HOLD
This key places conversation on hold.
8. NET
When users are not successfully registered to their service provider, this
button light will blink. User can press this button to try to register again to
their service provider.
9. MESSAGE
This key allows retrieval of Voice Mail messages.
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10. Menu / OK
When the IP phone is in idle state, this key is used as a “Menu” key to bring
up the menu selection on the LCD display.
When inside the menu selection/setting on the LCD display, this key is used
as an “OK” key to enter into a lower level of menu selection or to accept the
edited item’s contents.
When the IP phone is in dial mode, the “OK/Menu” key is used as a “Dial
Out” key.
11. Cancel
When the IP phone is in the menu selection, this key is used to escape to an
upper level of the menu selection.
When the IP phone is in menu edit mode, this key is used to cancel current
edit and escape to an upper level of the menu selection.
When users need to enter information in text characters, press this button
and the alphabet shown on the keypads will be displayed.
13. TRANSFER
See Section 5.6.
When the IP phone is off-hook and this key is pressed immediately, the last
dialed number will be called out right away.
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15. Speed Dial M1 – M10
Users are able to store 10 specific phone numbers in the slots of M1 - M10.
Users are able to make a speed dial call to the specific party by pressing the
speed dial key from M1 – M10.
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2.2 LCD Menu List
View
¾ Network Value
¾ IP Address
¾ Network Mask
¾ Default Route
¾ DNS Server
¾ Ping
¾ Restart
¾ Image Version
¾ (Yes / No) PKT Trace
¾ H323
¾ Number
¾ Password
¾ H.323 ID
¾ (Yes / No) Reg to GK
¾ (Yes / No) RTPLowBW
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¾ SIP
¾ Disp Name
¾ Login ID
¾ Number
¾ Password
¾ (Yes / No) Proxy On
¾ If Yes:
¾ Proxy Address
¾ Proxy Port
¾ (Yes / No) Outbound Proxy
¾ If Yes:
¾ Outbound Proxy IP
¾ Outbound Proxy Port
¾ (Yes / No) RTPLowBW
¾ Forward Mode
¾ (Yes / No) Immediate
¾ If Yes:
¾ Immediate Number
¾ (Yes / No) Busy
¾ If Yes:
¾ Busy Number
¾ (Yes / No) No Answer
¾ If Yes:
¾ No Answer Number
¾ No Answer Time
Advanced
¾ System
¾ DSP Version
¾ Upgrade / DnLoad
¾ FTP Server IP
¾ Image File Name
¾ Upgrade Image
¾ Upgrade Loader
¾ Config Profile
¾ Debut Mode
¾ Dump Address
¾ Dump Size
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¾ Dump!
¾ Network
¾ MAC Address
¾ NTP Server
¾ Restart Count
¾ RTP Process
¾ (Yes / No) Bypass Server
¾ (Yes / No) Jitter Buffer
¾ (Yes / No) Auto Upgrade
¾ Phone Advanced
¾ Codec
¾ G.711u
¾ G.711a
¾ G.729
¾ G.723
¾ Voice
¾ (Yes / No) VAD
¾ BG Noise Level
¾ Volume
¾ Ring Volume
¾ Handset Volume
¾ Hand free Volume
¾ Codec Tx Gain
¾ Scrn Con (0-9)
¾ Ring Type (1-10)
¾ UI Mode
¾ Console
¾ Lcd
¾ Both
¾ SIP Advanced
¾ Protocol
¾ Min Media Port
¾ Max Media Port
¾ Reg From
¾ Reg To
¾ Reg Expire
¾ Reg Action
¾ Local Port
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¾ User Setting
¾ Platform
¾ Billing Server
¾ Login
¾ User Name
¾ Password
¾ Confirm Password
¾ Statistics
¾ User Statistics
¾ User Statistic
¾ Phone Statistic
¾ Phone Statistics
¾ Call Missed
¾ Call Received
¾ Call Dialed
¾ Additional
¾ International
¾ My Country Code
¾ Area Prefix Code
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2.3 Installation Environment
Step 1:
Connect your IP phone to its handset by plugging one end of the coil cord
into the handset and the other end into the jack on the bottom of the IP
phone. To connect to the internet, plug into the middle jack on the back of
the IP phone. This jack is marked WAN. See figure 4 above
Step 2:
If you would like to have your PC online at the same time, please connect
the far right port to your PC. This Jack is marked PC. See figure 4
above.
Step 3:
Please plug in your power adaptor to your IP phone and power source.
LCD of your IP phone will display “Starting…….” and then “SIP / Hi xxx”
menu within approximately 4 seconds.
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3 Configuration from Keypads
The VOIP 2747 series IP Phone is designed to be installed easily and is very
user-friendly so almost all the configurations can be done through the keypad and
LCD screen display on the phone set in a few minutes. In order to make a VoIP
call, please configure through the keypad as described in the following few
sections:
Notice:
(1). When needing to input a character in any menu item, please press that key
button quickly to switch between different characters to set the correct one.
(2). When the input mode is in digit mode (you can only input ‘0’ – ‘9’ and ‘*’,
‘#’), and you want to input a character, please press “TXT ←→ NUM” key
first to toggle to “character” input mode.
The first step in using the VOIP 2747 IP Phone is to set the network
configuration to let the IP Phone connect to internet. This step depends on
your network environment and phone models, so please use the proper
method in configuring the IP Phone for your internet connection.
Please press “ok” key to set Yes on (Yes / No) Dynamic IP when using
DHCP.
Most cable modem connections also use DHCP.
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3.1.2 PPPoE Method
Please press “ok” key to set Yes on (Yes/No) PPPoE when using PPPoE
method, then key in the account from the ISP.
PPPoE Username – please input the user name of the account given by your
ADSL ISP.
PPPoE Password – please input the password of the account given by your
ADSL ISP.
For the other network environments, the users will need to set the static IP
provided by their ISP or from the office MIS representative.
Under the “Static IP” submenu, please key in the IP address, network mask
and default router settings provided by your ISP or a private IP address.
Once all the network settings are complete, please restart your IP phone.
To check whether your internet connection is working properly, go to View Æ
Ping, then key in a public IP address to ping it. If the response is ok, then the
network settings are complete.
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3.2 Registration to GK / Proxy Server
After the network environment is set and connected to the Internet, you can
register the IP Phone to your H.323 gatekeeper or SIP Proxy server. Please choose
one of the following methods to register to a gatekeeper or proxy server. (Method
depends on your IP Phone model)
When the VoIP vendor/operator is running the H.323 system, use the VOIP
2747 to register to the gatekeeper. Configure the following parameters to do the
registration.
E. (Yes/No) Full RRQ – whether to register with full RRQ(* Default (No))。
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alive. Some gatekeepers require the devices to keep re-registering to it using
the full information, in this case, please set the “FullRRQ” to “Yes”.
Note: The VOIP 2747 can pass the NAT/router devices if the gatekeeper supports
this function. The IP Phone will automatically detect if the gatekeeper supports
this and do the proper configuration to work under this NAT environment.
When the VoIP vendor/operator is running the SIP system, use the VOIP
2747 to register to the proxy server. Configure the following parameters to do
the registration.
A. Number – please input the phone number (username) to register to the proxy
server.
D. (Yes / No) Proxy On – please select Yes and register to the proxy server.
After this item is enabled, two more menu items will appear. One is the
proxy server address, to set the IP address or domain name address of the
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proxy server. Another is the proxy server port, to set the port of the proxy
server; its value is usually 5060 unless specified by the service vendor.
If the IP Phone does not register to any proxy server, it still can call to
another IP phone by calling the IP address directly.
E. (Yes / No) Outbound Proxy – please set this item to Yes if the registration
needs to pass through the Outbound Proxy server.
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3.3 Registration / Startup message
When the network and registration configurations are set, please restart the IP
Phone. The LCD display on the IP phone will show one of the following
messages: (depends on whether the registration is ok or not.)
Hi (number)
Date Time
This means that the VOIP 2747 is working OK and is ready for
outgoing/incoming calls. The number inside braces () is the IP phone’s
number.
Hi (GK Off)
Date Time
This means that the VOIP 2747 is working OK and is ready for
outgoing/incoming calls. However, if the “Reg to GK” flag is “No”, then the
IP Phone does not need to register. In this case, you can call the IP address of
other IP Phone directly.
(b). When the IP Phone is registered to a SIP proxy server successfully, the LCD
screen will display the following message:
SIP (number)
Date Time
This means that the VOIP 2747 is working OK and ready for
outgoing/incoming calls. The number inside braces () is the IP phone’s
number.
This means that the VOIP 2747 is working OK and ready for outgoing /
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incoming calls. If the “Proxy On” flag is “No”, then the IP Phone does not
need to register. In this case, you can call the IP address of another IP Phone
directly.
(c). When the IP Phone is configured to use a gatekeeper or proxy server, but
has not yet registered successfully or failed, the LCD screen will display the
following message:
Registering (number)
Date Time
(1). Duplicate: this means that the registering number is duplicated with
others, or the IP Phone’s previous registration information is still kept
in the gatekeeper/proxy server if not unregistered last time (this could
happen when the IP Phone is powered off instead of restarted from the
menu item). If the previous registration information is not cleared, you
may need to wait about 4 minutes to let the IP Phone register
successfully again.
(2). Security : this means that the account (username / password / H.323
ID) is not correct, please check your account again.
When the RegFail message is displayed, the IP phone can not make
any calls. The menu selection and the on-hook / off-hook function will work
OK.
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(e). When the IP Phone fails in obtaining an IP address from the DHCP server, the LCD
screen will display the following message :
DHCPFail (number)
Date Time
(f). When the IP Phone is set to use the PPPoE method for network connection,
but has some problem in finding the PPPoE server (ADSL modem), the
LCD screen will display the following message :
PPPoE FindFail(number)
Date Time
(g). When the IP Phone is set to use the PPPoE method for network connection,
but has some problem in the PPPoE account, the LCD screen will display
the following message:
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username/password is not correct. The IP phone can not make any calls, but
the menu selection and the on-hook / off-hook function will work OK.
(i). When the IP Phone is enabled to “Immediate Forward”, the LCD screen will
display the following message :
FWD(number)
Date Time
This means that the VOIP 2747 is working OK and ready for
outgoing/incoming calls. For any incoming call, it will be forwarded to the
“Immediate Forward Number”, and this IP Phone will not ring. See Sect.
3.4.
The phone supports three different Call Forwarding modes. These modes are
selected using the “Configure”Æ”Forward Mode” option.
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3.4.2 Busy Forward
To enable Busy Forward mode, Press MENU/OK with Item 2 Busy selected.
The display will change to 2 (Yes) Busy and 3. Busy Number will appear. Press
DOWN Then MENU/OK to enter the target number for Busy Forwarding. The
display will show Busy Number =. Enter the number using the keypad and
press MENU/OK. Then press CANCEL three times. The display will show
SIP(Ext Number). If Busy forwarding is selected, all calls to the phone will be
sent to the target number when the phone is in use.
Under the “View” menu item, there are many submenu items. These are mainly
for information about the phone’s current status, but they also provide simple
functions to check the status of the network.
The “Ping” function is one of the most commonly used tools to determine the
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state of a network connection. By selecting “View”Æ”Ping” and keying in the
IP address or domain name of another device, the phone can check the status of
its connection to that device. This can also be used to check the status of the
Internet connection by pinging another device that is already on the public
Internet.
3.5.3 Restart
Setting the “View”Æ”PKT Trace” to (Yes) will cause the phone to display all the
signaling messages received on the LCD screen. This may be useful for
debugging.
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4 Web Configuration
The phone provides a web interface for configuration. Please note that the
web interface will only work with Internet Explorer. To use this configuration
method, enter the IP address of the phone into the browser address bar. The
following login window will appear:
root
Username:
****
Password:
enter
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4.1 Network Configuration on web
Clicking on the “Network” icon on the left banner will bring up following
page to allow network configuration.
4.1.1 Basic
Please see section 3.1 for an explanation of each field.
4.1.2 Others
View the MAC Address of the IP Phone. The MAC Address cannot be
changed at this screen.
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4.1.2.3 Time Zone
Used to specify the time zone. Clicking the “View” icon will
display a list of all available time zones. For example, the value
should be set to -6 for US Central Time.
Clicking on the “H.323” icon on the left banner will display the following page to
allow configuration of H.323 related parameters.
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4.2.3 Advanced
The minimum value of the range of the transmitted RTP packet’s port.
The maximum value of the range of the transmitted RTP packet’s port.
4.2.3.3 Codec :
Clicking on the “SIP” icon on the left banner will display the following page to
allow configuration of SIP related parameters.
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4.3.1 SIP Parameters
Please see section 3.2 for an explanation of each field.
4.3.3 Advanced
Modifies the “From:” field in the SIP / SDP messages. This field
should be left empty in most cases.
4.3.3.2 Reg To
Modifies the “To:” field in the SIP / SDP messages. This field should
be left empty in most cases.
The minimum value of the range of the transmitted RTP packet’s port.
The maximum value of the range of the transmitted RTP packet’s port.
4.3.3.6 Codec
The type of codec used to transmit RTP packets. In SIP calls, the type
of codec used is determined by the called party.
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4.4 Phone Configuration on web
Clicking on the “Phone” icon will display the following page to allow
configuration of phone related parameters.
4.4.1 Prefix
These entries are used for VoIP systems which require special treatment of
country codes, international codes, and area codes.
Any numbers placed in the “My Country Code” and “Area Prefix Code”
boxes will be appended to the front of dialed numbers.
Any numbers placed in the “International Code” box will be removed
from the front of dialed numbers.
In most cases, these values are not needed and the boxes should be left
empty.
4.4.2 Voice
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4.4.2.2 Handset Volume
4.4.2.4 CodecTxGain
4.4.2.6 RTPLowBW
Checking this box will cause the phone to decrease the voice packet
bandwidth in the transmit direction. This should only be used if the
internet bandwidth is not good enough to support normal packet
bandwidth.
Checking this box will increase the jitter buffer from 150ms to 400ms.
This may improve the received voice quality in some internet
connections.
4.4.3 Others
4.4.3.1 VAD
Checking this box will enable Voice Activity Detection. With this feature
enabled, the phone will detect if the user is talking and will not send
packets when the user is silent. This can help to decrease the bandwidth
requirement.
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4.4.3.2 BG Noise Level
Clicking on the “System” icon will display the following page to allow
configuration of system related parameters.
This page allows software upgrades, username/password changing, and rebooting
(restarting).
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5 Call Functions
To make a call, pick up the handset or press the SPEAKER button, dial the called
party’s number, and end with “#” key or RE/DIAL.
To answer a call when the phone is ringing, pick up the handset or press the
SPEAKER button.
The call history can be displayed on the LCD screen by pressing the “Å” key or
the “Æ” key when the phone is idle.
There are three kinds of call history:
Incoming is the record of numbers and time of the last 10 incoming calls when
answered
Outgoing is the record of numbers and time of the last 10 successful outgoing
calls.
Missed is the record of numbers and time of the last 10 incoming calls that were
not answered
When viewing the call history, press the “MENU/OK” key to do one of the
following three actions:
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5.4 Redial
To call the last dialed number (redial), press the “RE/DIAL” key immediately
after going off-hook.
Call forwarding functions can be set via the keypad or by web configuration.
Please refer to section 3.5 for more details.
This phone supports three types of call transfer. They are ringing transfer, blind
transfer and attended transfer.
When a call is incoming and the IP phone is ringing, by pressing the “Transfer”
key and then another IP phone’s number you can transfer the call immediately to
another party without answering the call.
With blind transfer, the caller is transferred to a third party without informing the
third party who is transferring the call.
Example: A calls B and A wants B to transfer the call to C. B presses the
“TRANSFER” button and waits to hear dial tone. Then B dials C’s number and
hangs up. A is now transferred to C.
With attended transfer, the caller is transferred to a third party after the third
party is informed.
Example: A calls B and A wants B to transfer the call to C. B presses the
“HOLD” and waits for dial tone. Then B dial the C’s number and waits for C to
answer. B can now inform C of the call transfer. When B hangs up, A is
transferred to C.
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5.7 Call Hold
As described above, the HOLD button allows attended transfer. If the number
of the third party is not dialed in 8 seconds, the phone will go into the “Hold”
state. By pressing HOLD again, the call can be continued
5.8.1.1 Add
After selecting the Phone Book, press “MENU/OK” to select Add. The display
will show “Name=”. Use the keypad to enter the name and press “MENU/OK”.
The display now shows “Number=.” Use the keypad to enter the number and
press “MENU/OK.” Press “CANCEL” twice to return to the idle state.
5.8.1.2 Edit
After selecting the Phone Book, press “DOWN” and then “MENU/OK” to select
Edit. The display will show all the entries in the Phone Book in alphabetical
order. Used the “UP” and “DOWN” keys to scroll to the desired name and
press “MENU/OK”. The display now shows 1.Name 2.Number. To edit the
name, press “MENU/OK”. To edit the number, press “DOWN” and then press
“MENU/OK”. After pressing “MENU/OK” the display will show the current
information stored for the name or number. Use the UP and DOWN keys and
the keypad to make changes, then press “MENU/OK.” Press “CANCEL” four
times to return to the idle state.
5.8.1.3 Delete
After selecting the Phone Book, press “DOWN” twice and then “MENU/OK” to
select Delete. The display will show all the entries in the Phone Book in
alphabetical order. Used the “UP” and “DOWN” keys to scroll to the desired
name and press “MENU/OK” to delete this entry. The display will now show
the new Phone Book list. Press “MENU/OK” to delete another entry or press
“CANCEL” three times to return to the idle state.
5.8.1.4 Delete All
After selecting the Phone Book, press “DOWN” three times and then
40
“MENU/OK” to select Delete All. The display will show “Are You Sure?”.
Press “MENU/OK” to delete all Phone Book entries. Press “CANCEL” twice
to return to the idle state.
6 Attachment
The followings devices that have been tested to be interoperable with the 2747 IP
Phone.
- Client/Terminal
Microsoft NetMeeting, Version 3.01
OpenH323 Phone, Version 1.6.0
WellTech, LANPhone 101
Sagitta IP Phone
RadVision H.323v4 Protocol Stack (IMTC 2002)
Leadtek Research Inc., BVP8770 Video Phone (IMTC 2002)
Sony Electronics, PCS-6000 / PCS-1600 (IMTC 2002)
Aethra Srl (Italy), Vega Pro (IMTC 2002)
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IBM Corporation, Java-based Voice Server (IMTC 2002)
First Virtual Communications, Click To Meet Web Endpoint (IMTC 2002)
Innovaphone, IP200 (IMTC 2002)
UniData iW200
Cisco7905 IP Phone
- GateKeeper
Cisco AS5350 Gatekeeper function
Citron Network Inc. Gatekeeper
Radvision ProLab GateKeeper Simulator, Version 1.0, October 2001
OpenH323 GateKeeper, Version 1.3.2
GnuGK
III (Institute, Institute for Information Industry) GateKeeper, Version 1.0
NexTone Communications Inc., MSW (IMTC 2002)
MediaDigm Technology Inc. (Canada), SureKeeper (IMTC 2002)
Lucent iMerge gatekeeper
Wintel gatekeeper
MKY gatekeeper
EZTone gatekeeper
Tellink gatekeeper
Hivocal gatekeeper
GCard gatekeeper
- Gateway
D-Link 4-port gateway
Vanguard Managed Solutions, Vanguard (IMTC 2002)
AudioCodes Ltd., MP108 (IMTC 2002)
Cisco Systems Inc., Test Bed#1 (IMTC 2002)
Polycom Israel, GW-25 (IMTC 2002)
Antek Tech., 2~16 port gateway
WellTech gateway
Quintum gateway
- MCU
OpenH323 MCU, Version 1.1.3
Polycom Israel, MGC-100 (IMTC 2002)
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- SoftSwitch
ZTE Telcom. Softswitch
WalkerSun softswitch
The following devices have been tested to be interoperable with the 2747 series IP
Phone.
Client/Terminal
-SCS-Client V1.00
-SIPS 2.0.43.11
-SJ-Phone 1.10.187c
-X-Lite – FWD V2.0
-Softphone – Simems V0.90Bata27
-Estara V3.0.0.15
-Cisco ATA-186
-Cisco 7905
-BCM SIP Phone
-GrandStream SIP Phone
Proxy Server
-Wparty SIP V0.5.0
-Wparty SIP V0.5.5.2
-Asterisk 0.5.0
-Linux – Vovida V1.5
-SER 0.8.10
-WalkerSun softswitch
Gentrice proxy server
Inphonex platform
DeltaThree platform
NEC SV7000 IP PBX
Trunking Gateway
-Cisco 5300
-Quintum D2400
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LIMITED WARRANTY
If you purchased this product new in the U.S. or Puerto Rico, CORTELCO warrants it against defects in material
and workmanship for a period of one (1) year from the date of original purchase. This warranty is in lieu of all
other express warranties. During the warranty period, CORTELCO agrees to repair or, at its option, replace the
defective product, or any part of it without charge for parts or labor. This is your exclusive remedy. This
warranty does not cover damage resulting from accident, misuse, abuse, improper installation or operation, lack
of reasonable care, the affixing of any attachment not provided by CORTELCO with the product and loss of
parts. The warranty is voided in the event any unauthorized person alters or repairs the unit.
Telephone companies and ISP’s use different types of equipment and offer various types of services to
customers. CORTELCO does not warrant that this product is compatible with the type of equipment of
any particular phone company, the ISP, or the services provided by them.
THIS WARRANTY GIVES YOU SPECIFIC LEGAL RIGHTS, AND YOU MAY HAVE OTHER RIGHTS, WHICH
VARY FROM STATE TO STATE. SOME STATES DO NOT ALLOW THE EXCLUSION OR LIMITATION OF
SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES OR LIMITATIONS ON HOW LONG AN IMPLIED
WARRANTY LASTS, SO THE ABOVE EXCLUSION AND LIMITATION MAY NOT APPLY TO YOU.
If failure occurs and your 2747 VoIP is in warranty, service shall be provided by returning it to
CORTELCO - Repair Center, 1703 Sawyer Road, Corinth, Mississippi 38834, shipping
prepaid. The product will be repaired or replaced if examination by us determines the product to be
defective.
2747001VIPPAK
Revision 5.1
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