Atcom Phone Manual
Atcom Phone Manual
Version: 1.2
2015-03-12
Rainbow1 IP Phone User Manual
Release note
Version Changed note Editor Date
1.0 First Release Aimee 2014-5-12
1.1 1.5 Management and Aimee 2014-8-25
Maintenance
support capture network traffic
4.3 SIP Setting
Modify the default value of RTP
Packet Size to 20
Add codec payload and name for
G726r16,G726r24,G726r32,G726r40
4.4 Account
Add parameter RPort Enable
Modify Codec configuration
4.6.1 Manual Update
Add Pcap feature
1.2 Pictures stefen 2014-3-12
All pictures are replaced to the latest
one
4.1 Basic
Add wizard module
4.5.1 Daylight Saving Time
Modify the parameters and configure method
4.5.2 Speed dial
Delete the speed dial module
4.6.1 Manual Update
Delete some parameters
4.6.4 Debug
Add the Debug module
5.2 Upgrade firmware under safe mode
Modify the way to upgrade firmware under
safe mode
Content
Contact ATCOM .................................................................................................................................................................5
7. Abbreviations..................................................................................................................................................................44
Contact ATCOM
Overview of ATCOM
ATCOM is the leading VoIP hardware manufacturer in global market. We have been
keeping innovating with customer’s needs oriented , working with partners to
establish a total solution for SMB VoIP with IP phone , IP PBX and Asterisk cards
With over 10 years’ experience of R&D , manufacturing and service in network and
VoIP filed ; mission of creating the biggest value for IP terminals , we commit
ourselves in supplying the competitive IP phone and other terminals for IP PBX ,
softswitch , IMS , NGN providers and carriers; supplying the competitive total VoIP
solution for SMB market. We keep improving the customer’s experience and creating
the bigger value with our reliable products. Until now, our VoIP products have been
available in 100+ countries and used by millions of end users.
Contact Sales
Address Area C, A2F , Block 3 ,Huangguan Technology Park ,
#21 Tairan 9th Rd, Chegongmiao , Futian District ,
Shenzhen China
Tel + (86) 755-83018618-8806
Fax + (86) 755-83018319
E-mail [email protected]
1. Overview of Rainbow 1
Rainbow1
1.1 Interfaces
Power input: DC 5V, 1000mA or POE
LAN: RJ45 port
PC: RJ45 port
Headset jack 1 : RJ9 port
Handset jack 1 : RJ9 port
1.2 Hardware
LCD: 132×52
FLASH: 16M
RAM: 16M
CPU: 262MHz Dual Core
LED indicator: 1 Status Light
1.3 Software
Sip 2.0 (RFC3261) and other related SIP RFCs
1 SIP lines
STUN
Jitter Buffer, VAD,CNG
G.711A/U, G722, G.723, G.726-16, G.726-24, G.726-32, G.726-40, G.729 ,
L16, iLBC
Echo Cancellation
SIP Domain name, Authentication
DTMF(inband, RFC2833, info)
Call transfer, Call forward, 3-way conference, Call hold, Call back
DND(Do Not Disturb), Auto answer, Blacklists, Block Call-ID, Block
Anonymous call, Dial plan, IP call
Phone book with 100 white records and 50 black records, 100 answered calls,
100 missed calls, 100 dialed calls
Update via HTTP, FTP, TFTP, PNP
Syslog
SNTP
WEB access with different login level
Multi-language: English, Chinese, Farsi, French, German, Hebrew, Italian,
Portuguese, Russian, Spanish, Turkish
Soft button: soft button * 3
Redundancy SIP server
1.4 Network
LAN/PC:Support bridge mode
Support PPPoE(ADSL,cable modem used for Internet connecting)
Support VLAN(DATA VLAN and VOICE VLAN)
Support L2TP VPN
LAN support Primary and Secondary DNS
LAN support DHCP Client
Support QoS
1.6 Protocol
IEEE 802.3 /802.3 u 10 Base T / 100Base TX
PPPoE: PPP over Ethernet
DHCP: Dynamic Host Configuration Protocol
SIP RFC3261, RFC3262, RFC3263, RFC3264, RFC3265, RFC2543, RFC3489,
RFC3842, RFC3515, RFC2976, RFC3428, RFC2327, RFC2782, RFC1889
TCP/IP: Transfer Control Protocol/Internet Protocol
RTP: Real-time Transport Protocol
RTCP:RTP Control Protocol
DNS: Domain Name Server
TFTP: Trivial File Transfer Protocol
HTTP:Hypertext Transfer Protocol
FTP:File Transfer Protocol
Operation temperature: lower than 60° C
Storage temperature: lower than 60° C
Humidity: 10 to 90% no dew
1.9 Installation
review its current IP address by pressing key on idle state. To access the web
interface, you can input the IP address in IE browser. E.G. The IP address of your
Rainbow1 is 192.168.1.100, you can input 192.168.1.100 and press enter key on
your browser to access its webpage. There are two login level:
User
Admin
No password is set for those two accounts with factory settings. You can click admin
button on the right corner of the webpage to switch from the user to admin mode. To
set the password for user and admin login you can firstly login as admin and enter
the Network--->Advance page as following.
Installation instruction
1. Desktop installation
A. Put the bottom side of the IP phone upside and press one-side joints of stand
bracket into the slot, please refer the picture as below:
ATCOM TECHNOLOGY CO., LIMITED www.atcom.cn 9
Rainbow1 IP Phone User Manual
B. Press the other side joints into the slot according to the direction of the arrow
2. Wall-hung Installation
A. Put the bottom side of the IP phone upside and press one-side joints of wall-hung
stand bracket into the slot, please refer the picture as below:
ATCOM TECHNOLOGY CO., LIMITED www.atcom.cn 11
Rainbow1 IP Phone User Manual
B. Press the other side joints into the slot according to the direction of the arrow
D. Press the other side joints into the slot according to the direction of the arrow
E. Knock in nails or screws on the wall according to the proportion of the distance
between the hanging holes as below:
ATCOM TECHNOLOGY CO., LIMITED www.atcom.cn 13
Rainbow1 IP Phone User Manual
2. Keypad of IP Phone
Volume
Soft Key
Navigation
Menu Voice
OK
Mail
Cance
Mute
l
Headset
Hold
Dial pad
Redial
Speaker
Icon Description
The extension is registered
The extension is unregistered
There is a new voice mail
There is an incoming call
The call is held
A. Answer by handset
Pick up the handset and talk with the caller. If you want to hang up, just put back the
handset. When you are talking with the handset and want to switch to speaker mode
or headset mode, please press key or key, and then put down the
handset.
B. Answer by speaker
Press key and talk with callers by built-in Micro-phone and Speaker. If you
want to hang up, please press key again. Switch calling or talking into
handset mode by lifting the handset under speaker mode. Press key will
switch calling or talking into headset mode.
C. Answer by headset
Keep your microphone connected with the RJ9 headset jack, when there is an
incoming call, press key and talk with the caller. If you want to hang up,
please press key again. Pressing key can change calling or talking
into speaker mode, and lifting the handset switches to handset mode.
Press key and input a phone number. Press soft key "Send" to dial the
number. When caller hear the tones of “du~~du~~” and the phone number your
dialed is being displayed on the LCD, the phone at the side of being called should be
Press Key and input a phone number. Press soft key "Send" to dial the
number. When caller hear the tones of “du~~du~~” and the phone number your
dialed is being displayed on the LCD, the phone at the side of being called should be
ringing. If the called party answers this calling, the call is established and the calling
timer is started immediately.
1. Press key and input the keypad password 123 to enter the menu and choose
“Directory” option. Press "Select" soft key and then find the contact person by
navigation keys. When the certain contact person is highlighted, press "Dial" or
just pick up the handset to call this number.
2. Pick up the handset, press "Directory" soft key, then select the contact person and
press "Dial" soft key.
3. Pick up the handset, press and enter “Directory”, then select the contact
person and press “Dial” soft key.
1. Press key and input the keypad password 123 to enter the menu and choose
“History” option, then enter sub-directory “Dialed Calls”, “Received Calls” or ”
Missed Calls” to select one of call history entry, and press "Dial" soft key or pick
up handset to call this number.
2. Pickup the handset, press "History” soft key, then select one of call history entry,
and press "Dial" soft key to call this number.
3. Pickup the handset, press and enter “Call History” ,then select one of call
history entry, and press "Dial" soft key to call this number.
Press key at speaker model, the current calling will be hung up.
3. Headset Hang up
Press key at headset model, the current calling will be hung up.
4. Hang up one line call
Press the hook to hang up the current calling when 2 calls happened simultaneously.
While on calling, press the "Transfer" soft key to hold the current call and phone the
third party. Then dial the target number you want to transfer to on the activated line
and press “Send” soft key to call that number. After the target party answers the call,
press "Transfer" soft key again to complete the transfer.
3.7 Voicemail
Rainbow1 has a key for entering voicemail box. Press key to enter
the menu to configure voicemail number if never configure it previously. Otherwise,
the voicemail number will be called after press it. If you want to modify it after
configured it, please go to the Account webpage to modify the voicemail number.
The input audio will be not transmitted to peer phone after pressing key, and
the phone will be muted even switched among different modes of speaker, handset
The current calling will be hold by pressing soft key “Hold” or key. And the
held call will be resumed after pressing soft key “Resume” or key or the
corresponding line key. Even on 3-way conference calling, the 3-way conference
calling will be held after pressing “Hold” key, and be resumed to 3-way conference
after pressing “Hold” Key again. Remember the conversation is still on hold without
being ended even if hung up under the status of hold.
1. After the third party answers the call, pressing "Conf" key again to establish the 3-
way conference.
2. 3-way conference initiator can press "Exit" soft key to quit from the conference and
leave the other two parties still in the conversation.
3. If the initiator hangs up the call or press the “End Conf” soft key, the conference
will be ended and the calling between the other two parties will be hung up.
Press "History" soft key or key when Rainbow1 is standby, all the incoming(->),
outgoing(<-) and missed calls(!) will be listed. There is other ways to check them:
A. Missed call
1. Press key.
2. Press key and key to select “Call History” then press “Select” soft key.
3. Press key and key to select “Missed Calls” then press “Select” soft key.
4. Press key and key to browse the missed call record. If there is no record,
the LCD display will be indicated “List is Empty”.
B. Answered call
1. Press key .
2. Press key and key to choose “Call History” and then press “Select” soft
key.
3. Press key and key to choose “Answered Calls” and then press “Select” soft
key.
4. Press key and key to browse the answered call records. If there is no
record, the LCD display will be indicated “List is Empty”.
C. Dialed call
1) Press key.
2) Press key and key to select "Call History" and then press “Select” soft
key.
3) Press key and key to select “Dialed Calls” and then press “OK” soft key.
4) Press key and key to browse the dialed call records. If there is no record,
the LCD display will be indicated “List is Empty”.
Press key, then the status of the phone will be displayed on the screen and you
will see the current IP address of the phone.
Dial Plan
Dial plan contains a series of digit sequences, separated by the ‘|’ character. The
collection of sequences is enclosed in parentheses ‘(‘ and ‘)’.
Default: (*xx.|xxxxxxxxxxxx.)
When user dials a series of digits, Rainbow1 will response in below way:
• No candidate sequences matched, the number will be rejected and “call ended” will
be displayed on the screen. For instance, if the default dial plan only supports digits,
any ‘*’ character or letters input will be rejected.
More than one candidate sequences matched, Rainbow1 will wait for more digits
input.
When input timeout occurs, Rainbow1 will dial the digits input already.
When input ‘#’ character, Rainbow1 will dial the input digits immediately.
Example:
(xxxxxxx|[*#]xxxx|9,1xxxxxxxxxx|00xxx!) contains 4 subsequences:
1. Allow to dial numbers with 7 digits
2. Allow to dial numbers with 4 digits and start by ‘*’ or ‘#’
3. Allow to play an “outside line” dial tone after pressing ‘9’ and dial numbers with 11
digits and start by 1
4. Forbid to dial numbers with 5 digits and start by 00
4. Web settings
Input the IP address in the web browser and press ‘Enter’ key to access Rainbow1's
user webpage.
Click "admin" which is on the right top corner to enter administrator webpage.
4.1 Basic
4.1.1 Status
4.1.2 Wizard
Wizard is an interface to configure network access type and SIP accounts quickly.
There are 3 ways available when configuring Internet Port:
After IP address filled manually, click the button “Next” to enter the web page of
“SIP Settings”, or click the button “Back” to back to web page of “Internet Port”.
3. If choose PPPoE, you should input the username/password (provide by ISP) of
PPPoE manually and then click the button “Next” to enter the web page of
“PPPoE Settings”,
After the username/password of PPPoE filled manually, click the button “Next” to
enter the web page of “Account Configuration”, or click the button “Back” to back
to web page of “Internet Port”.
The configuration of SIP Settings will be saved to Line 1 automatically.
Click the button “Finish” to save configurations, and click “Back” back to the
previous web page.
4.2 Network
4.2.1 Basic
There are 3 ways to connect to the internet: DHCP, Static and PPPoE, please choose
one according to your own situation.
2. Static IP
3. PPPoE
Press ‘Submit’ button after finishing setting and all the settings info will be saved and
taken effect after Rainbow1 reboots.
4.2.2 Advance
Web Server
Enable Web Server: Enable or disable web access. If choose "no", you’re not able
to access Rainbow1’s webpage.
Admin password: Set password for admin webpage access. Input ‘http://ip-
address/index.asp’ in the web browser to access admin’s login webpage after
setting the admin password, then input username(admin) and password to access
the admin’s webpage.
User password: Set password for user webpage access. Input ‘http://ip-
address/user.asp’ in the web browser to access user’s login webpage after setting
the user password, then input username(user) and password to access the user’s
webpage.
HTTP port: set port for HTTP access (defaults to 80)
For example, Rainbow1's IP is 192.168.1.223
HTTP port was set as 100, you have to type "http://192.168.1.223:100" in web
browser to enter Rainbow1’s webpage.
VPN
After apply, the phone will be reboot. The VPN IP address will be shown on the
System Status webpage.
VLAN
Port Link
Choose the network type and port link for LAN and PC
1. LAN Port Link: Auto negotiate, full duplex 10Mbps, full duplex 100Mbps, half
duplex 10Mbps, half duplex 100Mbps.
2. PC Port Link: Auto negotiate, full duplex 10Mbps, full duplex 100Mbps, half duplex
10Mbps, half duplex 100Mbps.
Qos
Syslog
1. Sip T1: RFC 3261 T1 value (RTT). Range: 0 – 64 sec, defaults to 0.5
2. Sip T2: RFC 3261 T2 value (Maximum retransmit interval for non-INVITE requests
and INVITE responses). Range: 0 – 64 sec, defaults to 4
3. Sip T4: RFC 3261 T4 value (Maximum duration a message will remain in the
network). Range: 0 – 64 sec, defaults to 5
4. Reg Retry Intvl: Interval to wait before the phone retries registration again after
encountering a failure condition during last registration. Range: 0 –65535,
defaults to 8
5. Sub Retry Intvl: Interval to wait before the phone retries subscriber again after
encountering a failure condition during last subscriber. Range: 0 –65535, defaults
to 10
RTP Parameters
1. RTP Port Min: Minimum port number for RTP transmission and reception. Range:
1–65535, defaults to 16384
2. RTP Port Max: Maximum port number for RTP transmission and reception. <RTP
Port Max> should be at least 2 larger than <RTP port Min>.Range: 1–65535,
defaults to 16482
3. RTP Packet Size(ms): Packet size in milliseconds, which can be 10ms, 20ms,
30ms, 40ms, 60ms,defaults to 20.
1. Enable Stun: Select “Yes” or “No” to enable or disable using stun to discover NAT
mapping.
2. Stun Server: Set stun server, which can be IP address or domain name.
4.4 Account
Rainbow1 has 1 line which is enabled to register by default.
SIP
1. Display Name: This name will be displayed on the LCD. It will show the User ID
instead if leave Display Name as blank.
2. User ID: Username of sip account.
3. Authenticate ID: Normally is the same as User ID, but blank is acceptable.
4. Password: Password of SIP account.
5. SIP Server: SIP server address, support both IP address and domain name.
6. SIP Port: SIP server port, defaults to 5060.
7. SIP Redundancy Server: SIP redundancy server address. It can be configured
manually and also can be auto configured as the server address carried in the
DNS SRV record if there is a DNS which supports specifying the location of the
server or domain for SIP protocol. Only when Rainbow1 fails to register to the
SIP Server, it will try to register to the SIP Redundancy Server.
8. Use Outbound Proxy: Select “Yes” or “No” to enable or disable outbound proxy.
9. Outbound Proxy Server:Set address of outbound proxy server. All signaling
requests will be sent to outbound proxy server firstly.
10. Outbound Proxy Port: Outbound proxy server port.
11. Local SIP Port: The SIP port which used by phone, defaults to 5060
12. Register Expires: Register expiration time, defaults to 300 seconds.
13. Subscribe Expires: Subscriber expiration time, defaults to 3600 seconds.
14. Transport Type: UDP/TCP/TLS. Defaults to UDP.
15. SIP 100Rel Require: Select “Yes” or “No” to enable or disable 100Rel. If
enabled, 100rel parameters will be added to the SIP request to support PRACK.
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Rainbow1 IP Phone User Manual
16. RPort Enable: Select “Yes” or “No” to enable or disable rport.
17. Session Timer Enable:Select “Yes” or “No” to enable or disable Session Timer.
18. Early Update Enable:Select “Yes” or “No” to enable or disable Early Update.
19. Caller ID Display: Select “Yes” or “No” to enable or disable Caller ID display
20. AutoSubscribeMWIEnable:Select “Yes” or “No” to enable or disable
SubscribeMWI
21. Server List:Choose the server type.
22. DNS Mode: Choose the DNS mode
23. BLF List URI: Set BLF list URI when Rainbow1 cooperates with Broadworks
(Broadsoft).
Codec Configuration
DTMF Configuration
1. DTMF Tx Method: Select the method to transmit DTMF signals to the remote
end: Inband, RFC2833, SIP INFO. Defaults to RFC2833.
2. DTMF Display: Select “Yes” or “No” to enable or disable display DTMF.
Dial Plan
Dial Plan: Configure dial rule for SIP account, please refer to dial plan.
4.5.1 Preference
Output Volume(1~8)
1. Handset Volume: Specify handset volume grade
2. SpeakerPhone Volume: Specify speaker volume grade
3. Headset Volume: Specify headset volume grade
4. Ring Volume: Specify ring tone volume grade
Input Gain
1. Handset Gain: Specify handset gain, the bigger the gain, the louder the other
party heard.
2. SpeakerPhone Mic Volume: Specify speaker gain, the bigger the gain, the
louder the other party heard.
3. Headset Volume: Specify headset gain, the bigger the gain, the louder the other
party heard.
LCD
1. Backlight Level: select the backlight level
2. Backlight Time(Seconds): select the backlight time
3. Contrast: select the contrast level
4. Keypad Password: set keypad access password
1. Interdigit Long Timer: If the numbers or characters input have not finished and
do not full match the dial plan, it will be not dialed out automatically until time
out. Range: 0 – 64 sec
2. Interdigit Short Timer: If the numbers or characters input are full matched the
dial plan, it will be dialed out automatically until time out. Range: 0 – 64 sec
2. Set Local Time(HH:mm:ss) : manually set local time or click to adjust local
time. Format: hour/minute/second. e.g. 12:00:00.
Daylight Saving Time
4.5.2 Features
Call Forward
1. Do Not Disturb: Select “Yes” or “No” to enable or disable DND (Do Not
Disturb).When DND enabled, all the incoming calls will be rejected. At this
4.5.3 Voice
Echo Cancellation
1. VAD:Select “Yes” or “No” to enable or disable VAD (Voice Active Detection). If
enable, RTP packets will not be sent when Rainbow1 is mute.
2. CNG:Select “Yes” or “No” to enable or disable CNG (Comfort Noise Generator).
If enable, comfortable noise will be sent to the remote end to let it perceive the
conversation is still active when Rainbow1 is mute.
Jitter Buffer
Rainbow1 is able to buffer incoming voice packets to minimize out-of-order packet
arrival. This process is known as jitter buffer.
1. Type: Choose type of jitter buffer. When choose Fixed, the size of jitter buffer is
fixed. When choose Adaptive, the size of jitter buffer is the sum of Minimum
Delay and the size of RTP packets.
2. Min Delay: The minimum delay of the jitter buffer.
3. Max Delay: The maximum delay of the jitter buffer.
4. Normal Delay: This is used to set fixed jitter buffer which should be between
Min Delay and Max Delay.
Administer can upload 2 user define ring. The ring file should be wav (8k, 8bit, u-law)
and no larger than 200 KBytes.
4.5.5 Tone
4.6 Update
1. Provisioning Server: The address to save control file for auto upgrading, it can
filled by http、https、tftp server、ftp server,for example,
tftp://192.168.1.111/upgrade_control_file.xml
2. User Name: The user name to access the file server
1. Reboot: Reboot will terminate all active calls, and restart the phone in a several
seconds.
4.6.4 Debug
Rainbow 1 support the function of capturing packages, click Start to begin capturing,
and Stop after finishing it, then click Export to download the file.
If the Pcap Feature is not enabled when something wrong happened with the phone,
you can click the Download button to get the syslog file, and then send it to ATCOM to
help you to solve issues.
6. Trouble Shooting
6.1 The phone can’t register successfully
1. Check the IP address, and if the mode of WAN port is DHCP, please make sure
the DHCP server is in service.
2. Check the gateway.
3. Check the DNS.
4. Make sure the information of the account is consistent with which offered by the
service supplier.
5. Make sure the SIP server is on.
6. Check the port of the SIP server whose default value is 5060.
6.3 Only one part can hear the voice during the call
1. Make an IP dial-up call to make sure the telephone receiver and microphone are
normal.
2. Enable STUN on web page.
3. Set STUN server as stun.sipgate.com.
4. Click ‘submit’ and wait for the phone to restart.
5. Try to make calls again.
7. Abbreviations
DND : Do Not Disturb
CFWD : Call Forward
Bxfer : Blind Transfer
Conf : Conference
Num : Number
DelChr : Delete Char
Y/N : Yes/No
SIP:Session Initiate Protocol
RTP:Real-time Transport Protocol
SDP:Session Description Protocol
VPN:Virtual Private Network
VLAN:Virtual Local Area Network
QoS:Quality of Service
Syslog : System log
UDP:User Data Protocol
TCP:Transmission Control Protocol
TLS:Transport Layer Security Protocol
BLF:Busy Lamp Field
DNS:Domain Name System
SRTP:Secure Real-time Transport Protocol
NTP:Network Time Protocol
VAD:Voice Activity Detection
CNG:Comfort Noise Generator