S50 IP PBX User Manual

S50 IP-PBX
User Manual
SAHAB Technology Co., Ltd.

S50 IP-PBX User Manual
Table of Contents
1. INTRODUCTION ...................................................................................... 4
1.1 FEATURES ............................................................................................ 4

1.2 HARDWARE SPECIFICATIONS....................................................................... 5
1.2.1 Exterior Appearance ...................................................................... 5
2. SYSTEM SET UP ...................................................................................... 6
2.1 CONNECTION DRAWING ............................................................................ 6

2.2 CONNECTING ETHERNET LINE ..................................................................... 6
2.3 SUPPLYING POWER .................................................................................. 7
3. MANAGING S50 ....................................................................................... 8
3.1 ADMINISTRATOR LOGIN ............................................................................ 8

3.2 STATUS MONITOR ................................................................................... 9
3.2.1 Line Status ................................................................................... 9
3.3 BASIC ............................................................................................... 11
3.3.1 Extension ................................................................................... 11
3.3.2 Trunk ......................................................................................... 20
3.3.3 Outbound Routes ........................................................................ 31
3.3.4 Phone Provisioning ...................................................................... 33
3.4 INBOUND CALL CONTROL ........................................................................ 40
3.4.1 IVR ........................................................................................... 40
3.4.2 Queues ...................................................................................... 42
3.4.3 Custom Prompts ......................................................................... 47
3.4.4 Ring Groups ............................................................................... 48
3.4.5 Inbound Routes .......................................................................... 50
3.4.6 Blacklist ..................................................................................... 54
3.5 INTERNAL SETTINGS .............................................................................. 55
3.5.1 Options ...................................................................................... 55
3.5.23.5.2 Business Hours .................................................................... 57
3.5.3 Feature Codes ............................................................................ 58
3.5.4 SIP Settings ............................................................................... 61
3.5.5 Voicemail Settings ....................................................................... 66
3.5.6 DISA ......................................................................................... 69
3.5.7 Conferencing .............................................................................. 70
3.5.8 Paging Groups ............................................................................ 71
3.5.9 DNIS Settings ............................................................................. 72
3.5.10 PIN User settings ...................................................................... 73
3.5.11 Speed Dial Settings ................................................................... 74
3.5.12 Music on Hold Prompts ............................................................... 75
3.6 NETWORK SETTINGS .............................................................................. 75
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3.6.1 LAN Settings .............................................................................. 75

3.6.2 Static Route ............................................................................... 77
3.6.3 Firewall ...................................................................................... 78
3.6.4 DHCP Server .............................................................................. 81
3.6.5 VLAN Settings............................................................................. 81
3.6.6 DDNS Settings ............................................................................ 83
3.7 SYSTEM SETTINGS ................................................................................ 84
3.7.1 External Storage ......................................................................... 84
3.7.2 Password Settings ....................................................................... 85
3.7.3 System Prompts.......................................................................... 85
3.7.4 Date and Time ............................................................................ 86
3.7.5 Backup and Restore ..................................................................... 87
3.7.6 Reset and Reboot ........................................................................ 87
3.7.7 Firmware Update......................................................................... 88
3.7.8 Alert Settings ............................................................................. 89
3.8 REPORTS............................................................................................ 90
3.8.1 Call Logs .................................................................................... 90
3.8.2 System Logs ............................................................................... 91
3.8.3 System Info ............................................................................... 92
4. ACCESS MRI ......................................................................................... 93
5. ACCESS CDR......................................................................................... 97
6. USE S50 99
6.1 MAKE OUTBOUND CALL ........................................................................... 99

6.1.1 Sample Routing via PSTN Trunk .................................................. 100
6.1.2 Sample Routing via VoIP Trunk ................................................... 101
6.2 INCOMING CALL .................................................................................. 103
6.2.1 Sample Routing to an IVR .......................................................... 103
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1. Introduction
S50 IP-PBXanIP-PBX for Small Businesses/Home Office

S50 is a standalone embedded hybrid PBX for small businesses and remote
branch offices of larger organizations. S50 also offers a hybrid solution (a
combination of VoIP applications using legacy telecom equipment) alternative
for enterprises who are not yet ready to migrate to a complete VoIP solution.
Applications
1.1Features
Auto-provision Follow me
Blind Transfer Interactive Voice Response (IVR)
BLF Support Intercom / Zone Intercom
Blacklist Music On Hold
Call Detail Records(CDR) Music On Transfer
Call Forward Paging / Zone Paging
Call Parking PIN Users
Call Recording Queue
Call Pickup QOS
Call Routing Ring Group
Call Transfer Route by Caller ID
Call Waiting Skype Integration (Skype Connect)
Caller ID Speed dial
Conference Three-way Calling
Define Office Time Do Not Disturb(DND)
Direct Inward System Access(DISA) Voicemail
Distinctive Ringtone VLAN
Firewalls
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1.2Hardware Specifications
1.2.1 Exterior Appearance
Front Side
Figure1-1 S50 Front Panel Picture
No. Identifying
Green LEDIndicates the power connectionis normal.
Green LEDIndicates the server system is in working order
Green LEDIndicates the system is ready.
Green LEDIndicates the internet interface is in use
Red LEDstands for FXO port
Orange LED: Indicates presence of a BRI port.
Green LEDstands for FXS port
LED Dual - Red blinkFXO port isnt connecting PSTN line.
LED Dual - Red and Green blinkFXO port receive an incoming call.
LED Dual - Red and Green fast blinkFXO port is in talking.
LED Dual - Green and Red blinkFXS port is ringing.
LED Dual - Green and Red fast blinkFXS port is in talking.
2) Back Side
Figure1-2 S50Back Side Picture
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2. System set up
2.1 Connection Drawing
PSTN PSTN
FXO
IP
Ports
Provider
Route
FXS LAN
r
601 Ports
602
Normal IP Soft
Phone Phone Phone
Figure 2-1
2.2Connecting Ethernet Line
S50 provides two 10/100MEthernet ports with RJ45 interface and LED
indicator. Plug Ethernet line into S50s Ethernet port, and then connect the
other end of the Ethernet line with a hub, switch, router, LAN or WAN. Once
connected, check the status of the LED indicator. A yellow LED indicates the
port is in the connection process, and a green LED indicates the port is
properly connected.
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2.3 Supplying Power
S50 utilizes the high-performance switch power supply, which supplies the
required power for the unit.
AC Input: 100~240V
DC Output: 12V,1A
Please follow the steps below to connect the S50 unit to a power outlet:
1. Connect the small end of the power cable to the power input port on the
S50 back panel, and plug the other end of the cable into a 100VAC power
outlet.
2. Check the Power LED on the front panel. A solid green LEDindicatesthat
power is being supplied correctly.
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3. Managing S50
3.1 Administrator Login
From your web browser, input the IP address of the S50 server.
If this is the first time you are configuringS50, please use the default
settings below:
IP Address: http://192.168.5.150
Note: S50 supports multiple administrators in hierarchical mode

( Administrator, General Manager, CDR Manager)
Administrator
have all authority .
Username: admin , Password:xontel
General Manager
have basic authority; without the advanced authority to create VoIP trunks,
reset, update, backup and restore S50.
Username: user, Password:password
CDR Manager
only have the authority to check the call recordings.
Username: cdr, Password:password
Figure 3-1
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3.2 Status Monitor
3.2.1 Line Status
Figure 3.2.1
S50 Status Description:
Extensions:
1) :Extension is unregistered
2) :Extension is idle
3 :Extension is ringing
4 :Extension is busy in a call
5 : Extension is on hold
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Trunks:
VOIP Trunk:
Status
Unregistered: Trunk registration failed.
Registered: Succeed registration, trunk is ready for use.
Request Send: Registering.
Waiting: Waiting for authentication.
FXO Trunk:
Status
Idle: The port is idle.
Busy: The port is in use.
Disconnected: The port hasnt connected the PSTN line.
More detail message, please refer to the LED identifying of front panel.
BRI Trunk:
Status
Ok: The ports connect correctly.
Error: The port is error.
Disconnected: The port hasnt connected the PSTN line.
Service Provider:
Status
Ok: Succeed registration, trunk is ready for use.
Failed: Trunk registration failed.
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3.3 Basic
3.3.1 Extension
Extension has two types: Analog extensions (FXS) and VOIP extensions
(SIPextension).
Figure 3.3.1
3.3.1.1Analog Extensions (FXS)
EditAnalog Extensions
On the administration page of FXS extensions, click Edit on the extension that
you want to edit, and modify the following information on thepopup window.
1) General
Extension
The numbered extension, i.e. 1234, that will be associated with this particular
User / Phone.
Port
The extension correspond port.
Name
A character-based name for this user, i.e. Mostafa , Ahmed .
Caller ID
The Caller ID (CID) string will be used when this user calls another internal user.
2) Voicemail
Enable Voicemail
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Check this box if the user should have a voicemail account.
Voicemail Access PIN #

Voicemail Password for this extension, i.e. 1234.
3) Mail Setting
Enable Send Voicemail
Once enabled, the voicemail will be sent to the below email address as an
attachment.
Send Voicemail to Email Address

This option defines whether or not voicemails/Fax is sent to the Email address
asan attachment.
Note: Please ensure that all voicemail settings are properly configured on the
System Settings -> Voicemail Settings page before using this feature.
4) Flash
Hook Flash Detection
Sets the amount of time, in milliseconds, that must pass since the last
hook-flash event received by S50 before it will recognize a second event. If a
second event occurs in less time than defined byHook Flash Detection, then S50
will ignore the event. The default value of Flash is 1000 ms, and it can be
configured in 1ms increments.
5) Group
Pickup Group
If this extension belongs to a pickup group, any calls that ring this extension can
be picked up by other extensions in the same pickup group by dialing the Call
Pickup feature code (default *4).
Note: *4 is the default setting, it can be changed under Feature Codes ->
General -> Call Pickup.
S50 supports max 20 pickup groups.
6) Follow me (Call Forwarding)

This function sets inbound call forwarding on an extension. An administrator can
configure FollowMe for this extension.
7) Other Options
Call Waiting
Check this option if the extension should have Call Waiting capability.If this
option is checked, the When busy follow me options will not be available.
DND
Dont Disturb.
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User Web Interface

Check this option to allow the user to login to the S50 User Web interface, which
can be used to access voicemail and extension recordings. Users may login to
the S50 User Web interface by using their extension number and voicemail PIN
# as the login and password respectively.
Ring Out
Check this option if you want to custom the ring time. Tone will stop over the
time defined.
8) Volume Settings
Rxgain: The Volume sent to FXS extension.
Txgain: The Volume sent out by the FXS extension.
9) Mobility Extension
S50 allows you to use your mobile phone as extension. If you set your mobile
phone as mobility extension and then you call S50 with this mobility phone, you
will hear a dial tone. S50 will recognize your call as a call from an extension. You
can dial the number of other extensions (Your caller ID will be the number of
your extension) and use all outbound route that your extension can use of S50.
Mobility Extension Number

Don't forget to add the dial patterns of the outbound route at the beginning of
your mobile phone number when you fill in the mobility extension number
filed.E.g. if you want to set 15960XXXXXX as mobile extension, and the dial
pattern of the outbound route is 9; you should set 915960XXXXXX here.
Note: If callback is enabled in the inbound route, the mobility extension function
of this inbound route will be disabled.
10) Spy Settings

S50 allows extension to monitor/barge in other conversation. Once this feature
is enabled, the extension has the ability to monitor/barge in other calls using the
feature codes for each spy mode, refer to 'Feature Codes' page for more
information.
spy modes
There are 4 spy modes available for choice:
General spy: you have the permission to use the following 3 modes.
Normal spy: you can only hear the call, but can't talk
Whisper spy: you can hear the call, and can talk with the monitored extension
Barge spy: you can hear the call and talk with them both
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Note: for example, if 500 want to monitor extension 501, we need to enable the
'allow being spied ' for 501, and choose the spy mode for extension 500.
Then pick up 500 and dial feature codes + 501 to start monitoring when 501 is in a
call.
If 500 choose normal spy, it should dial*90501 to start monitoring;
if 500 choose whisper spy, it should dial *91501 to start monitoring;
if 500 choose barge spy, it should dial *92501 to start monitor;
if 500 choose general spy, it can dial *90501,*91501 or *92501 to start monitor.
11) Call Duration Settings

Setup the max cull duration for every call of this extension, but its only valid for
outbound calls. And if enter 0 or leave this blank empty, the value would be equal
to the max call duration configured in the Option Settings page.
Note: This setting will not be valid for internal calls.
Figure 3.3.1.1
3.3.1.2 VOIP Extension
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A VOIP extension is an SIP Account that allows an IP Phone or an IP Soft- Phone

client to register onS50.
1. Add VOIP Extension

Go to Extensions VOIP Extensions Create New Extension
1) General
Type
Extension type: SIP
SIP The extension sends and receives calls using the VoIP protocol SIP.
Extension
The numbered extension, i.e. 1234, that will be associated with this particular
User / Phone.
Password
The password for this extension, Ex: 12t3f6
Name
A character-based name for this user, EX: Mostafa , Ahmed
Caller ID
The Caller ID will be used when this user calls another internal extension.
2) Voicemail
Enable Voicemail
Voicemail Access PIN #

The voicemail Password for this extension, i.e. 1234.
3) Mail Setting
This option defines whether or not voicemails or faxes are sent to an Email
Address as attachment.

Once enabled, the voicemail will be sent to email as an attachment.
Email Address
Email address used to receive the voicemail or Fax.
Note: Please ensure that the section 'SMTP Settings For Voicemail'(in the
'Voicemail Settings') have been properly configured before using this feature.
4) Group
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Pickup Group
If this extension belongs to a pickup group, any calls that ring this extension can
be picked up by other extensions in the same pickup group by dialing the Call
Pickup feature code (default is *4).
Note: *4 is the default setting, it can be changed under Feature Codes ->
General -> Call Pickup.
5) Follow me (Call Forwarding)

Call forwarding for an extension can be configured here. The administrator can
configure FollowMe option for this extension. If you want to transfer the call to
an outbound number, please follow the dial pattern of outbound route filled in
the outbound number.
For example: transferring to your mobile phone number 123456789, the dial
pattern of outbound route is 9., you should fill in 9123456789 here.
6Other Options
.Call Waiting
Check this option if the extension should have Call Waiting capability.If this
option is checked, the When busy follow me options will not be available. The
call waiting function of IP phone has higher priority than S50s call waiting
function.
.DND
Dont Disturb. When DND is enabled for an extension, the extension will be not
available.
.User Web Interface

Check this option to allow the user to login to the S50 User Web interface, which
can be used to check voicemail and extension recordings. Users may login to
S50 User Web interface by using their extension number and voicemail PIN # as
the login and password respectively.
.Ring Out
Check this option if you want to customize the ring time. Ring tone will stop over
the time defined.
7)Spy Settings
S50 allows extension to monitor/barge in other conversation. Once this feature
is enabled, the extension has the ability to monitor/barge in other calls using the
feature codes for each spy mode, refer to 'Feature Codes' page for more
information.
spy modes
There are 4 spy modes available for choice:
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General spy: you have the permission to use the following 3 modes.
Normal spy: you can only hear the call, but can't talk
Whisper spy: you can hear the call, and can talk with the monitored extension
Barge spy: you can hear the call and talk with them both
Note: for example, if 500 want to monitor extension 501, we need to

enable the 'allow being spied ' for 501, and choose the spy mode for
extension 500.
Then pick up 500 and dial feature codes + 501 to start monitoring
when 501 is in a call.
If 500 choose normal spy, it should dial*90501 to start monitoring;
If 500 choose whisper spy, it should dial *91501 to start
monitoring;
If 500 choose barge spy, it should dial *92501 to start monitor;
If 500 choose general spy, it can dial *90501,*91501 or *92501
to start monitor.
8) VoIP Settings
NAT
This setting should be used when the system is using a public IP address to
communicate with devices hidden behind a NAT device (such as a broadband
router). If you have one-way audio problems, you usually have problems with
your NAT configuration or your firewall's support of SIP and/or RTP ports.
Qualify
Send check alive packets to IP phones
SIP Transport
This will be the transport method used by the extension. The options are UDP
(default) or TCP or TLS.
DTMF Mode RFC2833, Info, Inband, Auto.
Remote Register
Allow to register remote extensions.
This option is used to enhance the system security, its disabled by default.
9) IP Restriction
Enable IP Restriction
Check this option to enhance the VoIP security for S50.If this option is enabled,
only the permitted IP/Subnet mask will be able to register this extension
number. In this way, the VoIP security will be enhanced.
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Permitted IP address/Subnet mask

The input format should be IP address+/+Subnet mask.
e.g."192.168.5.100/255.255.255.255" means only the device whose IP
address is 192.168.5.100 is allowed to register this extension number.
e.g."192.168.5.0/255.255.255.0" means only the device whose IP address is
192.168.5.XXX is allowed to register this extension number.
10) Mobility Extension

S50 allows you to use your mobile phone as extension. If you set your mobile
phone as mobility extension and then you call S50 with this mobility phone, you
will hear a dial tone. S50 will recognize your call as a call from an extension. You
can dial other extension numbers (Your caller ID will be the number of your
extension) and dial out using all outbound route that your extension can use on
S50.
Mobility Extension Number

Don't forget to add the dial patterns of the outbound route at the beginning of
your mobile phone number when you fill in the mobility extension number
filed.E.g. if you want to set 15960XXXXXX as mobile extension, and the dial
pattern of the outbound route is 9; you should set 915960XXXXXX here.
Note: If callback is enabled on the inbound route, the mobility extension

function of this inbound route will be disabled.
11) Call Duration Settings

Setup the max cull duration for every call of this extension, but its only valid for
outbound calls. And if enter 0 or leave this blank empty, the value would be
equal to the max call duration configured in the Option Settings page.
Note: This setting will not be valid for internal calls.
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Figure 3.3.1.2
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2. Add Multiple Extensions
Go to Extensions VOIP Extensions Add Multiple Extensions.

1) Choose the number of extensionsthat you would like to create.
2) Choose the type of extension that you would like to create.
3)Fill in the starting extension number.
Figure 3.3.1.3
3. Edit VOIP Extension
Click Edit on VOIP Extension administration page or click Modify Selected

Extensions to editextension.
3.3.2Trunk
There are four types of trunks: Analog trunks (FXO),BRI Trunk, VoIP Trunk and
Service Provider trunks.
Figure 3.3.2
3.3.2.1 AnalogTrunk (FXO)
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1. Edit Analog Trunk (FXO)

On the Trunk administration page, click Edit on the selected trunk and modify
its properties in the popup window:
1General
Trunk Name
A unique label used to identify this trunk when listed in outbound rules,
incoming rules, etc. Ex: 'pstn5'
Volume Setting
Used to modify the volume level of this trunk . Normally, this setting does not
need to be changed.
2Busy Detection
Busy Detection
Busy Detection is used to detect far end hang-up or for detecting a busy signal.
Select Yes to turn this feature on.
Busy Count
If Busy Detection is enabled, it is also possible to specify how many busy tones
to wait for before disconnecting the call. The default is 4, but better results can
be achieved if set to 6 or even 8. Remember, the higher the number, the more
time will be required to release a channel. A higher setting lowers the
probability that you will encounter random hang-ups.
Busy Interval
The busy detection interval
Busy Pattern
If Busy Detection is enabled, it is also possible to specify the cadence of your
busy signal. In many Countries, it is 500msec on, 500msec off. Without Busy
Pattern specified, S50 will accept any regular sound-silence pattern that repeats
<Busy Count> times as a busy signal. If you specify Busy Pattern, then S50 will
further check the length of the tone and silence, which will further reduce the
chance of a false positive disconnect.
Frequency Detection
Used for Frequency Detection (Enable detecting the busy signal frequency or
not).
Busy Frequency
If the Frequency Detection is enabled, you must specify the local frequency.
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Polarity Detection
Configure if the call needs to be hung up when a polarity signal arrived
3Advanced Options
Caller ID Start
This option allows you to define the start of a Caller ID signal:
Ring:Start when a ring is received (Caller ID Signaling: Bell_USA, DTMF).
Polarity: Start when a polarity reversal is started (Caller ID Signaling:
V23_UK,V23_JP,DTMF).
Before Ring:Start before a ring is received (Caller ID Signaling: DTMF).
Caller ID Signaling
This option defines the type of Caller ID signaling to use. It can be set to one of
the following:
Bell: bell202 as used in the United States
v23_UK: suitable in the UK
v23_Japan: suitable in Japan
v23-Japan pure: suitable in Japan
DTMF: suitable in Denmark, Sweden, and Holland
.Caller ID Detection
For fxo trunks, this option forces S50 to clarify Caller ID incoming calls.
Figure 3.3.2.1
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3.3.2.2BRI Trunk
1. Edit BRI Trunk
1General
Trunk Name
A unique label used to identify this trunk when listed in outbound rules,
incoming rules, etc. Ex: 'BriTrunk1'
Signaling
Signaling method.
BRI-CPE: ISDN BRI in TE mode and Point to Point.
BRI-CPE-PTMP: ISDN BRI in TE mode and Point to multi Point.
BRI-NET: ISDN BRI in NET mode and Point to Point.
BRI-NET-PTMP: ISDN BRI in NET mode and Point to multi Point.
Switch Type
National: National ISDN type2 (common in the US)
ni1: National ISDN type 1
dms100: Nortel DMS100
4ess: AT&T 4ESS
5ess: Lucent 5ESS
euroisdn: EuroISDN
qsig: D-channel signaling protocol at Q reference point for PBX networking .
PRI Dial Plan

Sets an option required for some (rare) switches that require a dial plan
parameter to be passed. This option is ignored by most BRI switches. It may be
necessary on a few pieces of hardware. This option can almost always be left
unchanged from the default.
Reset interval
Sets the time in seconds between restart of unused channels. Some PBXs don't
like channel restarts. so set the interval to a very long interval e.g. 100000000
or 'never' to disable *entirely*. If you are in Israel, the following is important:
As Bezeq in Israel doesn't like the B-Channel resets happening on the lines, it is
best to set the reset interval to 'never' when installing a box in Israel. Our past
experience also shows that this parameter may also cause issues on local
switches in the UK and China.
PRI Local Dial Plan

Sets an option required for some (rare) switches that require a dial plan
parameter to be passed. This option is ignored by most BRI switches. It may be
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necessary on a few pieces of hardware. This option can almost always be left
unchanged from the default.
Over Lap Dial

Whether S50 can dial this switch using overlap digits. If you need Direct Dial-in
(DDI; in German "Durchwahl") you should change this to yes, then S50 will wait
after the last digit it receives.
PRI Indication
Tells how Device should indicate Busy() and Congestion() to the switch/user.
Accepted values are:
inband: Device plays indication tones without answering; not available on all
PRI/BRI subscription lines .
outofband: Device disconnects with busy/congestion information code so the
switch will play the indication tones to the caller. Busy() will now do same as
setting PRI_CAUSE=17 and Hangup().
Enable Facility
To enable transmission of facility-based ISDN supplementary services (such as
caller name from CPE over facility).
NSF
Used with AT&T PRIs.If outbound calls are being rejected due to "Mandatory
information element missing" and the missing IE is 0x20, then you need this
setting.
Echo Cancellation
Echocancel Obviously this disables or enables echo cancellation, it is
recommended to not turn this off.
Hide CallerID
If you want others to see your CID, please disable this option.
Codec
You can choose alaw or ulaw codes.
2) CallerID Prefix
International Prefix
When there are international calls coming in via this BRI trunk, the International
Prefix you have set here will be added before the CID. So you can know this is
an international call before you answer it.
National Prefix
When there are national calls coming in via this BRI trunk, the National Prefix
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you have set here will be added before the CID. So you can know this is a
national call before you answer it.
Local Prefix
When there are Local calls coming in via this BRI trunk, the Local Prefix you have
set here will be added before the CID. So you can know this is a local call before
you answer it.
Private Prefix
When there are Private calls coming in via this BRI trunk, the Private Prefix you
have set here will be added before the CID. So you can know this is a Private call
before you answer it.
Unknown Prefix
When there are calls with unknown number coming via this BRI trunk, the
Unknown Prefix you set here will be shown as the caller ID.
3) DOD Setting
Global DOD
Global direct outward dialing number.
DOD
Direct Outward Dialing Number.
Associated Extension
The extension make call out via BRI Trunk will display the associated DOD.
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Figure 3.3.2.2
3.3.2.3VOIP Trunk
. Add SIP Trunk

Input correct SIP information (provide by VOIP provider). Inaccurate
information will prevent the trunk from registering.
1) General setting
Type
SIP Identifies whether the trunk sends and receives calls using the VoIP
protocol SIP
Provider Name
A unique label to help you identify this trunk when listed in outbound rules,
incoming rules etc. Ex: 'Xontel'.
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Hostname/IP
Service providers hostname or IP address.5060 is the standardport number
used by SIPprotocol. Dont change this part if it is not required.
.Domain
VoIP providers server domain name.
Username
Username of SIP account. Used for SIP trunk registration.
.Authorization name
Used for SIP authentication. Leave this blank if not required.
Password
Password of SIP account.
.From User
All outgoing calls from this SIP Trunk will use the From User (In this case the
account name for SIP Registration) in From Header of the SIP Invite.
.Online number
Define the online number that expected by Skype Connectand some other SIP
service providers. Leave this field blank if it's no required.
Maximum Channels
Controls the maximum number of outbound channels (simultaneous calls) that
can be used on this trunk. Inbound calls are not counted against the maximum.
Leave blank to specify no maximum.
Caller ID
Specify the caller ID to use when making outbound calls over this trunk. The
caller ID set in the extension screen will override the caller ID set in the VOIP
trunk screen. Please note that not all the service providers support this feature.
Contact your service provider for more information.
Outbound Proxy Server

A proxy that receives requests from a client,even though it may not be the
server resolved by the Request-URI.
Codecs
Define the codec for this sip trunk and its priority
Note: codec can only display when edit itit is editedafter creating the trunk.
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Transport
This will be the transport method used by the SIP Trunk. This method is given by
the SIP trunk provider. The options are UDP (default) or TCP or TLS.
Enable SRTP
Define if SRTP is enabled for this trunk
Qualify
Send check alive packets to the sip provider.
DTMF mode
Set default mode for sending DTMF of this trunk.Default setting: rfc2833
2)DOD Setting
DOD
The extension make call out via SIP Trunk will display the associated DOD.
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Figure 3.3.2.3
3.3.2.4Service Provider
1. Add Service Provider

To Create the Service provider definition you need to complete the following
VoIP fields.
1) General
Type
SIP
SIP Identifies whether the trunk sends and receives calls by using the VoIP
protocol SIP.
Provider Name
A unique label would help to you identify this trunk. Ex: 'Provider2'.
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Hostname/IP
Service providers hostname or IP address.
Note: 5060 is the standardport number used by SIPprotocol, Dont change this
part if it is not required.
Maximum Channels
Controls the maximum number of outbound channels (simultaneous calls) that
can be used on this trunk. Inbound calls are not counted against the maximum.
Leave blank to specify no maximum.
.
Codecs
Define the codec for this sip trunk and its priority
Note: codec can only display when edit it after creating the trunk.
Transport
This will be the transport method used by the SIP Trunk. This method is given by
the SIP trunk provider. The options are UDP (default) or TCP or TLS.
Qualify
Send check alive packets to the sip provider.
DTMF mode
Set default mode for sending DTMF of this trunk.Default setting: rfc2833
2) DOD Setting
DOD
The extension make call out via BRI Trunk will display the associated DOD.
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Figure 3.3.2.4
3.3.3 Outbound Routes
Outbound routing defineshow outgoing calls are processed through the trunks.
Figure 3.3.3
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3.3.3.1 New Outbound Route
Click New Outbound Route and fill in the corresponding information in the
popup window.
1General
Route Name
Name of this Outbound Route. Ex: 'Local' or 'Long Distance' etc.
Dial Pattern
Outbound calls that match this dial pattern will use this outbound route. There
are a number of dial pattern characters that have special meanings:
X: Any Digit from 0-9
Z: Any Digit from 1-9
N: Any Digit from 2-9
[12345-9]: Any digit in the brackets (in this example, 1,2,3,4,5,6,7,8,9)
The .Character will match any remaining digits. For example, 9011. will match
any phone number that starts with 9011, excluding 9011 itself.
The ! will match none remaining digits, and causes the matching process to
complete as soon as it can be determined that no other matches are possible.
Example 1: NXXXXXXwill match any 7 digits phone number.

Example 2: 1NXXNXXXXXwill match a phone number starting with a 1,
followed by a 3-digit area code, and then 6 digit number.
Strip digits from front

Allows the user to specify the number of digits that will be stripped from the
front of the phone number before the call is placed. For example, if users must
press 0 before dialing a phone number, one digit should be stripped from the
dial string before the call is placed.
Prepend these digits before dialing

These digits will be prepended to the phone number before the call is placed. For
example, if a trunk requires 10 digit dialing, but users are more comfortable
with 7 digit dialing, this field could be used to prepend a 3 digit area code to all
7 digit phone numbers before calls are placed. When using analog trunks, a 'w'
character may also be prepended to provide a slight delay before dialing.
Password
The route password can be used to protect this route from being accessed
without a password.
Memory Hunt
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Round robin with memory, remembers which trunk was used last time, and then
use the next available trunk to call out.
Member Extensions
Defines the extensions that will be permitted to use this outbound route.
Member Trunks
Defines the trunks that can be used for this outbound route.
Figure 3.3.1
3.3.4 Phone Provisioning
The Auto Provision sub menu provides users a method to Auto Provision IP
Phone after the Express Setup process.
Note: Auto Provision functions fully test with these models:
Xontel (S10P , S11P , S12P . S16P , S17P , S18P , S19P , S20P , S21P , S22P ,
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S23P)
Snom ( 300,320,360,370)
Polycom (IP 6000,IP 7000,IP 32X,IP33X,IP430,IP450,IP550,IP560,VVX1500)
Cisco (IP7940,IP7960)
Aastra(480i,480i CT,6757i,6757i CT, 6737i)
Grandstream(GXP1450,GXP2100,GXP2110,GXP2120)
Escene(ES220,ES320,ES330,ES410,ES620)
Fanvil(C56,C58,C60,C62)
News:
When provisioning Xontel and Yealink IP phone, S50 is not needed to
be set as the only DHCP server any more.
Figure 3.3.4.2
3.3.4.3 Phone book

You can add your contacts here and provision them to your IP phone.
Figure 3.3.4.2
1) Add Contact
Type
There are three types: None, VIP and Deny list (Blacklist).
Group
There are 5 groups: None, Friends, Family, Work, Colleagues list.
Nick Name
You can set a nick name for this number.
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Favorite
Only works with snom phone.
Organization
Input the organization of this contact. Only works with snom phone.
Title
Input the title of this contact. Only works with snom phone.
Email
Input the email of this contact. Only works with snom phone.
Birthday
Input the birthday of this contact. Only works with snom phone.
First Name
Input the first name of this contact. Only works with snom phone.
Family Name
Input the family of this contact. Only works with snom phone.
Office Number
Input the office number here.
Mobile Number
Input the mobile number here.
Home Number
Input the home number here.
Sub Number
Add sub number of this contact. Only works with snom phone.
Note
Take some note of this contact. Only works with snom phone.
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Figure 3.3.4.2.1
2) Upload Phonebook
You can upload a phonebook before auto provision, which will be provisioned to
the IP phone when using auto provision feature to configure your IP phones. The
format of phonebook should be *.xml.
Note: All the existing phonebooks of the IP phone will be replaced automatically
if the phonebooks are configured in this way.
3.3.4.4 configure phone

Let's take provisioning yealink as an example
Create New Phone have two modes,
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Create New phone in webpage and Upload the IP Phones configure file.
1. Add new phone via webpage

Click Add Phone and fill in the corresponding information in the popup window.
1) General
Enabled
Choose yes or no to enable or disable this extension.
NewConfig
If the firmware version number is greater than or equal to 70,please enable this
option.
MAC address
Input the MAC address of IP phone
Name
Put the name of this Phone here.
Manufacturer
You can choose the Manufacturer of IP phone
Phone Type
Choose the model of your phone. Only for snom phone
Call Waiting
This call feature allows your phone to accept other incoming calls to an
extensionalready in an active call.
Key as Send
Configure the key as send, you choose # ,* or disable it
Auto redial
Enable the auto redial for IP Phone
Auto answer
Configure if auto answer is allowed for IP phone
Phone book
Enable the feature of phone book of IP phone
Line
Extension: Selected the extension number for IP Phone.
Label: It is shown on the LCD for users to identify the account.
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Line Active: You can choose on/off to enable/disable the account respectively.
Figure 3.3.4.4.1
2) Audio codec
In this section, we can design the allowed codec for IP phone
Figure 3.3.4.4.2
3) Line keys settings
Configure the DSS keys/Function Keys
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Figure 3.3.4.4.3
3.3.4.5Not configured phone

In this section, S50 will scan all the supported IP phones and display here, we
can click the 'MAC address' of IP phone and input the corresponding information
in the popup window, like the picture shows below
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Figure 3.3.4.5
3.3.4.6 Upload a file

Click Upload a file and choose the configure file of IP phonein the popup
window.
Note: the file format must be .cfg
Please edit the configuration files in advance before uploading.
3.4 Inbound Call Control
3.4.1IVR
When theres an inbound call aims at Auto Attendant, S50 will play an IVR
recording and route the caller to the requesteddestination (for example,
Welcome to XX company, for sales press 1, for technical support press 2, for
operator press 0, etc). The system will transfer the call to corresponding
extension according to DTMFdigits inputted by the user.
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Figure 3.4.1
3.4.1.1 Create IVR

Click Create New IVR.
1General
Number
S50 treats IVR as an extension;you can dial this extension number to reach the
IVR.
Name
A name for the IVR
Prompt
The prompt recording that will be played when this IVR is reached.
Repeat Count
The number of times that the selected IVR prompt will be played.
Key Timeout
Wait for the user to enter a new extension for a specified number of seconds.
Allow Dialing of Other Extensions

Allow the caller to dial other extensions other than the ones explicitly defined.
2Key Press Events

A list of actions that can be performed depending on the digit dialed by the user.
Time Out
Defines the timeout action. A timeout occurs after the IVR prompt has finished
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playing for the number of times specified by the 'Repeat Count' field.
Invalid
Defines the invalid action. The invalid action is triggered if the user enters a
DTMF digit that is not defined for this IVR.
Figure 3.4.1
3.4.2 Queues
Call Queues give users (i.e. call centers) an efficient means to have their calls
answered in the order they were received to deliver top tier customer service.
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Call queues allow calls to be sequenced to one or more agents.

Note: Dial 'Queue number + '*'' to log in or 'Queue number + '**'' to log out the
queue. For example, if the queue number is '680', then agent can dial '680*' to
log in or '680**' to log out.
1General
Queue Name
A name for the Queue.
Queue Number
Use this number to dial into the queue, or transfer callers to this number to put
them into the queue.
Queue Password
You can require agents to enter a password before they can login to this queue.
Queue Agent Timeout

The number of seconds an agent's phone can ring before we consider it a
timeout.
Queue Max Wait Time

The maximum number of seconds a caller can wait in a queue before being
pulled out. (0 for unlimited).
Queue Ring Strategy

This option sets the Ringing Strategy for this Queue.The options are
RingAll: Ring All available Agents simultaneously until one answers.
LeastRecent: Ring the Agent which was least recently called.
FewestCalls: Ring the Agent with the fewest completed calls.
Random: Ring a Random Agent.
RRmemory: Round Robin with Memory, Remembers where it left off in the last
ring pass".
2Agents
This selection shows all users. Selectinga user here makes them a agent of the
current queue.
3) Caller Position Announcements

Announce Position
Announce position of caller in the queue
Announce Hold Time

Enabling this option causes S50 to announce the hold time to the caller
periodically based on the frequency timer. Either yes or no; hold time will not be
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announced if <1 minute.
Frequency
How often to announce queue position and estimated hold time.
Note: '0 seconds' means disable the announcement
4) Periodic Announcements
Prompt
Select a prompt file to play periodically.
Frequency
How often to announce a prompt to the caller.
5) Events
If a caller presses the key while waiting in the queue, this setting selects which
action should process the key press.
6) Failover-Destination
Defines the failover action. A failover occurs after the user reach the Queue max
wait time.
7) Others
Music On Hold
Select the 'Music on Hold' Class for this Queue.
Leave When Empty

This option controls whether callers already on hold are forced out of a queue
that has no agents. There are two options.
Yes: Callers are forced out of a queue when no agents are logged in.
No: Callers will remain in a queue with no agents.
Join Empty
This option controls whether callers can join a call queue that has no agents.
There are two options,
Yes: Callers can join a call queue with no agents or only unavailable agents
No: Callers cannot join a queue with no agents
The default option is No.
Agent Announcement
Announcement played to the Agent prior to bridging in the caller.
Join Announcement
Announcement played to callers once prior to joining the queue.
Retry
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The number of seconds we wait before trying all the phones again.
Wrap-up time
How many seconds after the completion of a call an Agent will have before the
Queue can ring them with a new call. The default is 30.
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Figure 3.4.2
.
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3.4.3Custom Prompts
1. Record new Prompt

The administrator can use this screen to record customprompts by doing the
following:
1Click Record New Custom Prompt
2)Input the desired file name on the popup window and choose an extension to
call for recording (such as 500).
3) Click Record. The selected extension will ring and you can pick up the phone
to start recording.
Figure 3.4.3.1
2. Upload Prompt
The administrator can also upload prompts by doing the following:
1)Click Upload Prompt.
2) Click Browse to choose the desired prompt.
3) Click Uploadto upload the selected prompt.
Note: The file size must not be larger than 1.8MB, and the file must be WAV
format:
GSM 6.10 8kHz, Mono, 1Kb/s;
Alaw/Ulaw 8kHz, Mono, 1Kb/s;
PCM 8kHz, Mono, 16Kb/s.
Figure 3.4.3.2
Note: the maximum amount of custom prompt is 16.
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3.4.4 Ring Groups
Ring groups can be configured to balance the call traffic for multiple users
andgive callers a higher level of availability for incoming calls.Multiple ring
methods and voicemail are supported.
Figure 3.4.4
3.4.4.1 Create Ring Group

Click New Ring Group to enter into the Manage Ring Groups page
1General
Ring Group Name
This option defines a name for this group, i.e. Sales. Ring Group Name is a
label to help you identify this group in the group list.
Ring Group Number

This option defines the numbered extension that can be dialed to reach this
group.
Strategy
This option sets the Ringing Strategy for this Group. The options are as follows:
1. RingAll Simultaneously: Ring all available Extensions simultaneously.
2. RingSequentially:Ring each extension in the group one at a time.
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Seconds to ring each member

1. If the strategy is RingAll Simultaneously, it means set the number of
seconds to ring this group before routing the call according to the Destination if
No Answer settings.
2. If the strategy is Ring Sequentially, it means set the number of seconds to
ring a single extension before moving onto the next one.
2) Ring Group Members

An extension can be made a member of this ring group by moving it into the
Selected box.
3) DestinationIf No Answer
When all members on this group fail to answer the call, system will handle the
call according to the selected destination.
Figure 3.4.4.1
Note: the maximum amount of ring group is 16.Each group includes 20
members at most.
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3.4.5 Inbound Routes
Inbound routing processes incoming call traffic to destination

extensionsduringoffice hours or outsideoffice hours.
Figure 3.4.5
3.4.5.1 Create Inbound Route

Click New Inbound Route to enter to the Manage Inbound Routes page.
When an incoming call arrives, the system will first check fax detection, then
Holidays, at last Business Days.
1General
Route Name
A name for this inbound route. Ex: 'pstncallin' etc.
DID Number
Define the expected DID Number if this trunk passes DID on incoming calls.
Leave this field blank to match calls with any or no DID info. Only service
provider, E1 trunks, BRI trunks or SIP trunks need to be configured with this
setting.
You can also use pattern matching to match a range of numbers.The following
patterns may be used:
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Example 1: NXXXXXX will match any 7 digits phone number.

Example 2: 1NXXNXXXXX will match a phone number starting with a 1,
Extension
Define the extension for DID number. This field is only valid when you use BRI,
SIP, SPS or SPX trunk for this inbound router. You can only input number and
-in this field, and the format can be xxx or xxx-xxx. The count of the number
must be only one or equal the count of the DID number.
Caller ID Number
Define the Caller ID Number to be matched on incoming calls. Leave this field
blank to match any or no DID info.
You can also use a pattern match (e.g. 2[345]X) to match a range of numbers.
The following patterns may be used:
Example 1: NXXXXXXwill match any 7 digits phone number.

Example 2: 1NXXNXXXXXwill match a phone number starting with a 1,
Distinctive Ringtone
S50 support mapping to custom ring tone files. For example, if you configure the
distinctive ringing for custom ring tone to Family, the ring tone will be played
if the phone receives the incoming call.
Note: If you want to use feature Distinctive ringtone, please confirm youre
your IP phone support this feature firstly. Currently distinctive ringtone can be
compatible with XonTel and Snom phone.
2) Member Trunks
This area allows you to select which trunks will be member trunks for this route.
To make a trunk a member of this route, please move it to the 'Selected' box.
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4) Business Days
Define where the calls will be routed during Business Days.
Office Days
Select one defined business days office days.
Office Hours Destination

Configure where to route the incoming calls during office hours.
End Calls
Route the incoming calls to end calls, System will auto hang-up the call.
Extension
Route the incoming calls to a specific extension.
Voicemail
Route the incoming calls to extensions voicemail.
IVR
Route the incoming calls to a specific IVR.
Ring Group
Route the incoming calls to a specific Ring Group.
Conference Room
Route the incoming calls to a specific Conference Room.
DISA
Route the incoming calls to a specific DISA.
Queues
Route the incoming calls to a specific Queue.
Outbound Routes
Route the incoming calls to a specific outbound route.
This function is mainly used for the connection of two branches.

For example: Company A locates headquarters in the USA with a branch B in
China. A and B both have S50 phone systems.
Now if staff of A would like to make a call to a telephone or mobile phone in
China from the extension of A but via the FXS line of B, that can be done by this
configuration.
Non-office Hours Destination
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Configure where to route the incoming calls during non-office hours.
5) During Holidays
Define where the calls will be routed during Holidays.
Holiday
Select the which defined Holiday to use.When a time is defined in both Business
Days and Holidays, it will be treated as Holidays.
Destination
Configure where to route the incoming calls during holidays.
Figure 3.4.5.1
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3.4.6 Blacklist
Blacklist is used to block an incoming/outgoing call. If the number of

incoming/outgoing call is registered in the number blacklist, the caller will hear
the following prompt: The number you have dialed is not in service. Please
check the number and try again.The system will then disconnect the call.
Figure 3.4.6
3.4.6.1 Create Blacklist

Click New Blacklist to create a new number blacklist.
Figure 3.4.6.1
Blacklist number
Enter the number you would like to block.
Type
The number blocked for incoming or outgoing calls or both.
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3.5 Internal Settings
3.5.1 Options
1) General
Ring Timeout
Number of seconds to ring a device before handling the call as per the
extensions Follow Me settings . Default value is 30s.
.MAX call duration

The absolute maximum amount of time permitted for a call. A setting of 0
disables the timeout. Default value is 6000s.
.Maximum concurrent calls

Maximum concurrent calls limits. Default value 0 means no limit
Music on hold
Used to set hold music for the system.
Tone Region
Please select your country or nearest neighboring country to enable the default
dial tone, busy tone, and ring tone for your region.
Note: please reboot the system to take it effect.
HTTP bind port/Web Access Port

Port to use for HTTP sessions. Default: 80
Note: please reboot the system to take it effect.
FXO Mode
FXO ports operation mode .
Distinctive Caller ID
When incoming calls are routed from ring group/queue/IVR, the caller ID
displays with the name of ring group / queue/IVR for example
5503302(ringgroup_default)
Note: for displaying (To display) IVR's name, please press the key instead of
the extension number directly.
Enable Follow Me Prompt

When set Follow me to Transfer to number on the extension page (e.g. when
500 is busy, transfer to 501), while 500 is busy, the call will be transferred to
501. If Enable Follow Me Prompt choosing yes, there will be prompt please
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wait while trying to look at the person you are callingbefore transferring the call.
Otherwise, the call will be transferred directly without any prompt. Default: Yes.
Music on hold for Follow Me

Choose to play default music-on-hold, ringing tone or stay mute while Follow
Me.
Invalid Phone Number Prompt

Configure the prompt when the dialed phone number is invalid.
Busy Line Prompt

Configure the prompt when the dialed phone number is busy.
Dial Failure Prompt

Configure the prompt when dial failed due to conjunction no-available channel.
2) Extension Preferences
User Extensions
The default value is 500 to 616
Ring Group Extensions

Paging Group Extensions

Conference Extensions
IVR Extensions
Queue Extensions
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Figure 3.5.1
3.5.2 3.5.2 Business Hours
1) General
Enable or Disable Business Hours
2) Others
Enable Office Closed Timing
By dialing *81 (*81 is default) on an extension will force the office time closed
for the device whatever the general setting is.
Enable Office Timing

By dialing *82 (*82 is default) on an extension will force the office time enalbed
for the device whatever the general setting is.
Disable Office (closed) timing

By dialing *081 (*081 is default) on an extension will disable the Office Timing
or Office Closed Timing.
3) Business Days
You can setup the business hours here.
4) Holidays
You can setup the holidays here.
If a time period is configured as both Holidays and office hours, it will be treated
as Holidays.
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Figure 3.5.2
3.5.3Feature Codes
1) General
One Touch Record
A user may initiateor stop call recording by dialing *1during a call. (*1 is default
setting)
Extension for Checking Voicemail

Userscan check their Voicemail by dialing *2 on their phone (*2 is default
setting).
Voicemail for Extension

Users can leave a voicemail to other extensions by dialing # on their phone or
the incoming call could be forwarded to an extensions voicemail directly. (# is
default setting).
For example, extension 500 want to leave a message for extension 501, users
can use 500 dial#501 to enter the voicemail of 501.
Voicemail main menu

Userscan go to the main menu by dialing *02 (*02 is default setting).
Attended Transfer
Users may transfer anincoming call by dialing *3 on their phone (*3 is default
setting).
Attended Transfer Timeout

The time out of transferring a call
Blind Transfer
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Users may blind transfer an incoming call by dialing*03 on their phone (*03 is
default setting).
Call Pickup
Users may pickup an incoming call by dialing *4 on their phone (*4 is default
setting)
Extension Pickup
Users may pickup a specific extensions incoming call by dialing *04+extension
number on their phone (*04 is default setting)
Intercom
Define the feature code that is used to dial an extension in intercom mode. For
instance setting this value to *5 would allow you to initiate an intercom call with
extension 501 by dialing *5501.
Normal Spy
In this mode, you can only listen to the extension being spied, for example you
can dial *90501 to monitor extension 501
Whisper Spy
In this mode you can listen/whisper to the extension being spied, for example,
dialing *91501 to listen to extension 501, you can also talk with 501 too.
Barge Spy
In this mode, you can barge in both extensions involved the call, for example
dialing *92501 to barge in and talk with all the extensions inside
2) Call Park Preferences

Call Parking
User may park an incoming call on his own telephone by pressing *6(*6 is
default setting)
Extension range used to park calls

User may park an incoming call on a designated extension at first and then pick
up the call again on any other extension.
Number of seconds a call can be parked before it is recalled.

Defines the number of seconds that a call can be parked before it is recalled to
the station that parked it.
3) Call Forwarding Preferences

Reset to Defaults
Users may reset all call forward defaults by calling *70 on their phone (*70 is
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default setting).
Note: When reset to defaults. The call forwarding settings will be configured as
follows:
Always forward: Disabled
Busy forward to Voicemail: Enabled
No answer forward to Voicemail: Enabled
Do not disturb: Disabled
Enable Forward All Calls

Users may enable always forward by calling *71 on their phone (*71 is default
setting)
Disable Forward All Calls

Users may disable always forward by calling *071 on their phone (*071 is
default setting)
Enable Forward When Busy

Users may enable busy forward by dialing *72 on their phone (*72 is default
setting)
Disable Forward When Busy

Users may disable busy forward by calling *072 on their phone (*072 is default
setting)
Enable Forward No Answer

Users may enable no answer forward by calling *73 on their phone (*73 is
default setting)
Disable Forward No Answer

Users may disable no answer forward by calling *073 on their phone (*072 is
default setting)
Forward to number
Users may activate call forwarding by dialing this feature code, followed by the
extension or phone number to forward all calls to.
Note: Users may activate Forward to number by dialing *74 + phone number.
e.g.: by dialing *74501, all calls will be forwarded to extension 501.
Forward to Voicemail
Users may forwardthe call to Voicemail by calling *074 on their phone (*074 is
default setting)
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Enable Do Not Disturb

Users may enable do not disturb by calling *75 on their phone (*75 is default
setting)
Disable Do Not Disturb

Users may disable do not disturb by calling *075 on their phone (*075 is default
setting)
Figure 3.5.3
3.5.4 SIP Settings
1) General
UDP Port
Port use for sip registrations, Default is 5060.
TCP Port
TLS Port
RTP Port Start

Beginning of RTP port range
RTP Port End

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End of RTP port range
DTMF Mode
Set default mode for sending DTMF.Default setting: rfc2833
Max Registration/Subscription Time

Maximum duration (in seconds) of a SIP registration. Default is 3600 seconds.
Min Registration/Subscription Time

Minimumduration(inseconds)of a SIP registration.Default is 60seconds.
Default Incoming/Outgoing Registration Time

Default Incoming/Outgoing Registration Time: Default duration (in seconds)
of incoming/outgoing registration.
Register Attempts
The number of SIP REGISTER messages to send to a SIP Registrar before giving
up. Default is 8 times.
Register Timeout
Number of seconds to wait for a response from a SIP Registrar before timed out .
Default is 20 seconds.
Calling Channel Codec Priority

Once enabled, when dialing out via SIP/SPS trunks, the codec of calling channel
will be selected in preference. If not, S50 will follow the priority in your SIP/SPS
trunks.
Video Support
Support for SIP video or no. Default is yes.
Max Bit Rate

Configure the max bit rate for video stream. The default: 384kb/s
DNS SRV Look Up

Please enable this option when your SIP trunk contains more than one IP
address.
User Agent
To change the useragent parameter of asterisk, the default is S50, you should
change it if needed.
2) NAT
Note: Configuration of this section is only required when using remote
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extensions.
Enable STUN
STUN (Simple Traversal of UDP through NATs) is a protocol for assisting devices
behind a NAT firewall or router with their packet routing.
STUN Address
The STUN server allows clients to find out their public address, the type of NAT
they are behind and the internet side port associated by the NAT with a
particular local port. This information is used to set up UDP communication
between the client and the VOIP provider and so establish a call.
External IP Address
The IP address that will be associated with outbound SIP messages if the system
is in a NAT environment.
External Host
Alternatively you can specify an external host, and the system will perform DNS
queries periodically.
This setting is only required when your public IP address is not static. It is
recommended that a static public IP address be used with this system. Please
contact your ISP for more information.
External Refresh Interval

If an external host has been supplied, you may specify how often the system will
perform a DNS query on this host. This value is specified in seconds.
Local Network Identification

Used to identify the local network using a network number/subnet mask pair
when the system is behind a NAT or firewall.
Some examples of this are as follows:
'192.168.0.0/255.255.0.0' : All RFC 1918 addresses are local networks;
'10.0.0.0/255.0.0.0' : Also RFC1918;
'172.16.0.0/12':AnotherRFC1918withCIDRnotation;
'169.254.0.0/255.255.0.0' : Zero conf local network.
Please refer to RFC1918 for more information.
NAT Mode
Global NAT configuration for the system. The options for this setting are as
follows:
Yes = Use NAT. Ignore address information in the SIP/SDP headers and reply to
the sender's IP address/port.
No = Use NAT mode only according to RFC3581.
Never = Never attempt NAT mode or RFC3581 support.
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Route = Use NAT but do not include rport in headers.
Allow RTP Reinvite

By default, the system will route media steams from SIP endpoints through
itself. Enabling this option causes the system to attempt to negotiate the
endpoints to route packets to each other directly, bypassing the system. It is not
always possible for the system to negotiate endpoint-to-endpoint media
routing.
3Codecs
A codec is a compression or decompression algorithm that used in the
transmission of voice packets over a network or the Internet.
u-law: A PSTN standard codec, used in North America, that provides very
good voice quality and consumes 64kbit/s in each direction (receiving and
transmitting) of a VoIP call.
a-law: A PSTN standard codec, used outside of North America, that

provides very good voice quality and consumes 64kbit/s in each direction
(receiving and transmitting) of a VoIP call.
GSM: A wireless standard codec, used worldwide, that provides adequate

voice quality and consumes 13.3kbit/s in each direction (receiving and
transmitting) of a VoIP call. GSM is supported by many VoIP phones.
SPEEX:Speex is an Open Source/Free Software patent-free audio

compression format designed for speech. The Speex Project aims to lower
the barrier of entry for voice applications by providing a free alternative to
expensive proprietary speech codecs. Moreover, Speex is well-adapted to
Internet applications and provides useful features that are not present in
most other codecs.
G.722:G.722 is a wideband speech coding algorithms which supports the bit

rate of 64, 56 and 48kbps wideband. Its a broadband voice encoding of G
series.
G.726: A PSTN codec, used worldwide, that provides good voice quality and
consumes 32kbit/s in each direction (receiving and transmitting) of a VoIP
call. G.726 is supported by some VoIP phones.
ADPCM, G.729A, H261, H263, H263p, H264,MPEG4.

Note: If you would like to use G.729, please enter your license.
4QOS
QoS (Quality of Service) is a major issue in VOIP implementations. The issue
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is how to guarantee that packet traffic for a voice or other media connection
will not be delayed or dropped due interference from other lower priority
traffic.When the network capacity is insufficient, QoS could provide priority
to users by setting the value.
5) Advanced Settings
From Field
Where to get the caller ID in sip packet.
To Field
Where to get the DID in sip packet.
180 Ringing
It is set when the telecom provider needs. Usually it is not needed.
Remote Party ID
Whether send Remote-Party-ID on SIP header. Default no.
Allow Guest
Whether allow anonymous registration extension. Default: no.
This option is used to avoid some anonymous calls by hackers.
Pedantic
Enable pedantic parameter. Default: no.
Session -timers
Enable sesstion-timer mode, default: yes.
Sesstion-expires
The max refresh interval.
Sesstion-minse
The min refresh interval, which mustn't be less than 90 .
Sesstion-refresher
Choose sesstion-refersher, the default is Uas.
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sFigure 3.5.4
3.5.5 Voicemail Settings
1) General Voicemail Settings
a) Message Options
Max Messages per Folder
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Set the maximum number of messages that can be stored in a single voicemail
box.
Max Message Time

Set the maximum length of a single voicemail message.
Min Message Time

Set the minimum length of a single voicemail message. Messages below this
threshold will be automatically deleted.
Ask Caller to Dial 5

If this option is set, the caller will be prompted to press 5 before leaving a
message.
Operator Breakout from Voicemail

If this option is set, the caller can jump out of the voicemail and go to the
destination (IVR) you set by dialing 0.
b) Greeting Settings
Busy Prompt
Greeting played when the extension called is busy.
Skip greeting: Do not play a greeting.
Play busy greeting: play the extension busy greeting.
Unavailable Prompt
Greeting played when the extension called is Unavailable.
Play Unavailable greeting: play the extension Unavailable greeting.
Leave a Message Prompt

Greeting when ask the caller to dial 5 to leave a message.
Play busy greeting: play the extension busy greeting.
Play Unavailable greeting: play the extension Unavailable greeting.
c) Playback Options
Announce Message Caller ID
If this option is enabled, the Caller ID of the party that left the message will be
played back before the voicemail message begins playing.
Announce Message Duration

If this option is set, the duration of the message in minutes will be played back
before the voicemail message begins playing.
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.Announce Message Arrival Time

If this option is set, the arrival time of the message will be played back before
the voicemail message begins playing.
.Allow Users to Review Messages

Allow callers to review their recorded message before sending it to voicemail.
2) SMTP Settings for Voicemail

Note: If you want to send voicemail messages as email attachments, please
configure this section.
E-mail Address
The E-mail Address that S50 will use to send voicemail.
Password
The password for the email address used above.
SMTP Server
The IP address or hostname of an SMTP server that the S50will connect to in
order to send voicemail messages via email, i.e. mail.yourcompany.com.
Port
SMTP Port: the default value is 25.
Use SSL/TLS to send secure message to server

If the server of sending email needs to authenticate the sender, you need to
select the check box.
Note: Must be selected for Gmail or exchange server.
After filling out the above information, you can click on the Test Account
Settings button to check whether the setup is OK.
1) If the test is successful, you can use the email safely.
2) If test failed, please check the above information is correct or network is
proper.
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Figure 3.5.5
3.5.6 DISA
DISA (Direct Inward System Access) allows someone calling in from outside the
telephone switch (PBX) to obtain an internal system dial tone and make calls as
if they were using one of the extensions attached to the telephone switch. To use
DISA, a user calls a DISA number,which invokes the DISA application. The DISA
application in turn requires the user to enter aPIN number, followed by the
pound sign (#). If the PIN number is correct, the user will hear dialtone on which
a call may be placed. Obviously, this type of access has serious security
implications, and great care must be taken not to compromise your security.
Figure 3.5.6.1
To add a new DISA application, click the New DISA button.
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Figure 3.5.6.2
1) General
DISA Name
Give this DISA application a name to help you identify it.
PIN#
The password for this DISA.
Response Timeout
The maximum amount of time the system will wait before hanging up the call if
the user has dialed an incomplete or invalid number. Default is 10 seconds.
Digit Timeout
The maximum amount of time permitted between each digit when the user is
dialing an extension number. Default is 5 seconds.
2)Member Outbound Routes

Used to set the outbound routes that can be accessed from this DISA.
Note: the maximum amount of DISA is 16.
3.5.7 Conferencing
Conference Calls increase employee efficiency and productivity, and provide a

more cost-effective way to hold meetings.
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Extension
This is the number dialed to reach this Conference Room.
Admin
Admin can kick a user out and can lock the conference room.
Pin #
Set a PIN # that must be entered in order to access this conference room (i.e.
1234).
Figure 3.5.7
3.5.8 Paging Groups
Paging is used to make an announcement over the speakerphone to a phone or

group of phones. Targeted phones will not ring, but instead answer immediately
into speakerphone mode. Please note that this section is for configuring paging
groups. If you would like to configure Intercom settings, please open the Other
Settings -> Feature Codes screen.
This feature is supported by the following SIP phones:
Yealink's T28,T26,T22,T20,T10T,T9CM. Other SIP devices may also work with
this feature but are not officially supported.
Note:A paging group can have a maximum of 20 members.
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Figure 3.5.8.1
Paging Group Number

Defines the numbered extension that may be dialed to reach this group.
Duplex
Paging is typically one way for announcements only. Checking this will make
paging duplex, allowing all users in the group to talk and be heard by all.
Figure 3.5.8.2
3.5.9 DNIS Settings
DNIS (Dialed Number Identification Service) is a telephone service that

identifies for the receiver of a call the number that the caller dialed.
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Note:
1. DID number is not available in PSTN/GSM/UMTS trunks
2. If DID is not configured here, all the calls via this trunk will show the DNIS
instead of the original caller ID.
Figure 3.5.9
3.5.10 PIN User settings
PIN User are used to manage lists of PINs that can be used to access restricted
features such as Outbound Routes. If user use PIN User call out, system will
auto record the PIN in the call detail records.
1) Options
Access Code
.Dial this code to access PIN.
Prompt for Entry

Prompt caller enter the PIN Number.
Prompt for Entry Failure
Prompt the caller when an invalid PIN is entered.
2) PIN User
S50 can store a number of PIN Users. PIN Users may be used to keep track of
calls in relation to particular activities or clients. They can also be used to keep
track of calls by particular users or sets of users.
When a PINis entered while making a call or during a call, that PINis included
in the call records output by the system.
PINentered are checked against those stored by the system. If an invalid PINis
entered, the PINis requested again.
The system administrator can configure certain numbers or types of numbers
to require entry of aPINbefore you can continue making a call to such a number.
The system administrator can also configure you to have to enter a PINbefore
making any external call.
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Name
A character-based name for this PIN list, i.e. XontelPIN.
PIN List
Enter a list of one or more PINs, One PIN per line.
Available Outbound Routes

The outbound routes which can be used by the pin user.
Figure 3.5.10.
3.5.11 Speed Dial Settings
1) Options
The prefix of speed dial
The prefix should be dialed before the speed dial number. Default is *99.
2) Add new speed dial.

Source Number
The speed dial number.
Destination Number
The number you want to call.
e.g. The source number is123. The destination number is 5503305.The prefix
number is*99.You can use an extension with any type to dial *99123, then it will
call to number 5503305.
Note: Dont forget to add the outbound dial prefix if you would like to dial the
speed dial number through trunk.
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Figure 3.5.11
3.5.12 Music on Hold Prompts
The administrator can upload on hold music as follows:

1) Click Upload Music on Prompt .
2) Click Browse to choose the desired audio file.
3) Click Upload to upload the selected file.
Note: The sound file format should be as follows:
GSM 6.10 8kHz, Mono, 1Kb/s;
Alaw/Ulaw 8kHz, Mono, 1Kb/s;
PCM 8kHz, Mono, 16Kb/s.
Figure 3.5.12
3.6 Network Settings
3.6.1 LAN Settings
DHCP
If this option is set, S50 will use DHCP to get an available IP address from
your local network. Not recommended.
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Enable SSH
This is the advance way to access the device, you can use the putty software
to access the device.In the SSH access, you can do more advance setting
and debug.
Port: the default is 8022,
Hostname
Set the host name for S50.
IP Address
Set the IP Address for S50.
Subnet Mask
Set the subnet mask for S50.
Gateway
Set the gateway for S50.
Primary DNS
Set the primary DNS for S50.
Secondary DNS
Set the secondary DNS for S50.
IP Address2
Set the second IP Address for S50.
Subnet Mask2
Set the second subnet mask for S50.
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Figure 3.6.1
3.6.2 Static Route
S50 will have more than one internet connection in some situations but it has
only one default gateway. You will need to set some Static Route for S50 to force
it goes out through different gateway when access to different internet.
The default gateway priority of S50 from high to low is OpenVPNWAN
portLAN port.
1) Route table
The current route rules of S50
Destination
The destination network to be accessed to by S50
Subnet Mask
Specify the destination network portion.
Gateway
Which gateway S50 will go through when access to the destination network.
Metric
The cost of a route is calculated by using what are called routing metric. Routing
metrics are assigned to routes by routing protocols to provide measurable
statistic which can be used to judge how useful (how low cost) a route is.
Interface
Define which internet port to go through.
2) Static Route Rules

You can add new static route rules here.
Figure 3.6.2
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3.6.3 Firewall
Firewalls are used to prevent unauthorized Internet users from accessing

private networks connected to the Internet, especially intranets. All
messages entering or leaving the intranet pass through the firewall, which
examines each message and blocks those that do not meet the specified
security criteria.
In order to ensure the safe operation of the system, we suggest to enable
and configure firewall.
Figure 3.6.3
1) Enable Firewall
Enable the firewall to protect the device. You should reboot the device to let
the firewall run successfully.
2) Common Rules
Name
A name for this rule , e.g. HTTP.
Description
Simple description for this rule. E.g.: Accept the specific host to access the web
interface for configuration.
Protocol
The protocols for this rule.
Port
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Initial port should be on the left and end port should be on the right.
The end port must be equal to or greater than start port.
IP
The IP address for this rule. The format of IP address is: IP/mask
Ex:192.168.5.100/255.255.255.255forIP 192.168.5.100
Ex:216.207.245.47/255.255.255.255 for IP 216.207.245.47
Ex:192.168.5.0/255.255.255.0 for IP from 192.168.5.0 to 192.168.5.255 .
MAC Address
The format of MAC Address is XX:XX:XX:XX:XX:XX, X means 0~9 or A~F in hex,
the A~F are not case sensitive.
Action
Accept: Accept the access from remote hosts.
Drop: Drop the access from remote hosts.
Ignore: Ignore the access.
Figure 3.6.3.1
3) Auto Defense
Port
Auto defense port, e.g.: 8022.
Protocol
Auto defense protocol, TCP or UDP.
Rate
The maximum packets or connections can be handled per unit time.
E.g.: (Port: 8022 Protocol: TCP Rate: 10/minute) means maximum 10 TCP
connection to port 8022 can be handled per minute, the eleventh connection will
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be refused directly.
Figure 3.6.3.2
4) IP Blacklist
You can set some packets accept speed rules here. When a IP address which
hasnt been accepted in common rules sends packets faster than the allowed
speed, it will be set as black IP address and blocked automatically.
a) New Rule
Port
Auto defense port
Protocol
Auto defense protocol. TCP or UDP.
IP Packets
Allowed IP packets number in the specific time interval.
Time interval
The time interval to receive IP packets. For example, IP packets 90,time
interval 60 means 90 IP packets are allowed in 60 seconds.
Figure 3.6.3.3
b) IP Blacklist Manage
You can manage the IP addresses which are blocked automatically here.
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Figure 3.6.3.4
5) Other Settings
Disable Ping
Enable this item, net ping from remote hosts will be dropped.
Drop All
When you enable Drop All feature, system will drop all packets or connection
from other hosts if there are no other rules defined. To avoid locking the devices,
at least one TCP accept common rule must be created for port used for SSH
access, port used for HTTP access and port sued for CGI access
3.6.4 DHCP Server
Dynamic Host Configuration Protocol (DHCP) is a network protocol that enables

a server to automatically assign an IP address to a computer from a defined
range of numbers (i.e., a scope) configured for a given network. You can set a
local network NTP server for S50 here too.
Note: When using Phone Provisioning for Grandstream IP phone, Enter the IP
address of the server directly, e.g.:192.168.5.150; for other phones using the
default configuration.
Figure 3.6.4
3.6.5 VLAN Settings
A VLAN(Virtual LAN) is a logical local area network (or LAN) that extends beyond
a single traditional LAN to a group of LAN segments, given specific
configurations.
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1) VLAN Over Lan
NO.1
Click the NO.1 you can edit the first VLAN over Lan.
VLAN Number
.The VLAN Number is a unique value you assign to each VLAN on a single device.
VLAN IP Address
Set the IP Address for S50VLAN over Lan.
VLAN Subnet Mask

Set the Subnet Mask for S50VLAN over Lan.
Default Gateway
Set the Default Gateway for S50VLAN over Lan
NO.2
Click the NO.2 you can edit the first VLAN over Lan.
VLAN Number
.The VLAN Number is a unique value you assign to each VLAN on a single device.
VLAN IP Address
Set the IP Address for S50VLAN over Lan.
VLAN Subnet Mask

Set the Subnet Mask for S50VLAN over Lan.
Default Gateway
Set the Default Gateway for S50VLAN over Lan.
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Figure 3.6.5
3.6.6 DDNS Settings
DDNS(Dynamic DNS) is a method / protocol / network service that provides the

capability for a networked device, such as a router or computer system using
the Internet Protocol Suite, to notify a Domain Name System (DNS) name
server to change, in real time, the active DNS configuration of its configured
hostnames, addresses or other information.
Enable DDNS
DDNS Server
Select the DDNS server you sign up for service.
User Name
User name the DDNS server provides you.
Password
User accounts password .
Host Name
Note:DDNS allows you to access your network using domain names instead of
IP address.The service manages changing IP address and updates your domain
information dynamically.You must sign up for service through dyndns.org,
freedns.afraid.org, www.no-ip.com, www.zoneedit.com
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Figure 3.6.6
3.7 System Settings
3.7.1 External Storage
The External Storage feature is used to extend storage space. Once configured,
the files (voicemail, call recording files, CDR files) created before the configured
days will be moved to the Net-Disk.
Auto-Backup extends the allocated disk space for backing up critical files. When
properly configured, S50 will move all qualified files to a Windows PC every 30
minutes. For the voicemail files and recoding files, they must be created before
Auto-Backup has been configured.
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Figure 3.7.1
3.7.2 Password Settings
1) Change Password
User
S50 support 3 levels users, choose one of them to change its password.
Enter Old Password

The default password is password.
Enter New Password
Retype New Password

To change the password, enter the new password and click update.The system
will then prompt you re-login using your new password.
2) CDR setting
Whether enable CDR User.
Figure 3.7.2
3.7.3 System Prompts
S50 have prompts of many languages. You can download the appropriate
language you need.
Note:
Auto-detection is highly recommended. But if you prefer to download via HTTP
or TFTP server, please contact the local dealer for the prompts.
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Figure 3.7.3
3.7.4 Date and Time
Set the date and time for S50.
Time Zone
You can choose your time zone here.
Daylight Saving Time

Set the mode to Automatic or disabled
Automatically Synchronize With an Internet Time Server

Input the NTP server so that S50 will update the time automatically.
Set Date & Time Manually
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Figure 3.7.4
3.7.5 Backup and Restore
Figure 3.7.5
3.7.6 Reset and Reboot
Reboot System
Warning:Rebooting the system will terminate all active calls!
Reset to Factory Defaults

Warning: A factory reset will erase all configuration data on the system.
Please do not turn off the system until the RUN light begins blinking. Any power
interruption during this time could cause damage to the system.
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Figure 3.7.6
3.7.7 Firmware Update
Upgrading of the firmware is possible through the Administrator web interface

using a TFTP Server or an HTTP URL.
Enter your TFTP Server IP address and firmware file location, then click start to
update the firmware.
Note
1. If enabled Reset configuration to Factory Defaults, System will restore to
factory default settings.
2. When update the firmware, please dont turn off the power.
Figure 3.7.7
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3.7.8 Alert Settings
If the device is attacked, the system will notify users the alert via call or E-mail.
The attack mode include IP attack and Web Login.
Figure 3.7.8.1
1.IPATTACK
When the system is attacked by IP address, the firewall will add the IP to auto
IP Blacklist and notify the user if it match the protection rule.
1Phone Notification Settings
PHONE Notification
Whether enable phone notification.
Number
The numbers could be set for alert notification, users can setup multiple
extension and outbound phone numbers. Please separate them by ;.
e.g. 500;9911, if the extension has configured Follow Me Settings, the call
would go to the forwarded number directly.
Attmtpts
The attempts to dial a phone number when there is no answer.
Interval
The interval between each attempt to dial the phone number. Must be greater
than 3 seconds, the default value is 10 seconds.
Prompt
Users will hear the prompt while receiving the phone notification.
2E-mail Notification Settings
Note: Please ensure that all voicemail settings are properly configured on the
System Settings -> Voicemail Settings page before using this feature.
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E-mail Notification
Whether enable E-mail Notification
Recipient Name
The recipients for the alert notification, and multiple email addresses are
allowed, please separate them by ;.
e.g. [email protected]; [email protected] .
Subject
The subject of the alert email.
Email Content
Text content support for predefined variables. Variable names and
corresponding instructions are as follows:
$(HOSTNAME) Host name

$(LOCALIP) Local IP address
$(SOURCEIP) Attack source IP address
$(DATETIME) Occurred
$(USERNAME) User nameWEBLOGIN effective
$(DESTMAC) Attacks destination MACIPATTACK effective
$(DESTPORT) Attacks destination Port numberIPATTACK effective
$(PROTOCOL) Protocol typeIPATTACK effective
$(INTERFACE) Network interface nameIPATTACK effective
Figure 3.7.8.2
2.WEBLOGIN
Web Login Alert NotificationEnter the password incorrectly five times to login
S50 Web interface will be as attack, the system will limit the IP login within 10
minutes and notify the user.
3.7.8.3
3.8 Reports
3.8.1 Call Logs
The call Log captures all call details, including call time, caller number, callee
number, call type, call duration,etc. An administrator can search and filter call
data by filter the call logs by call date, caller/callee, trunk, duration, billing
duration, status, communication type and pin user.
www.sahabtec.com
Figure 3.8.1
3.8.2 System Logs
You can download and delete the system logs of S50.
Enable Hardware Log

Save the infomation of hardware; (up to 4 log files)
Enable Normal Log

Save the prompt information; (up to 16 log files)
Enable Web Log

Save the history of web operations (up to 2 log files)
Enable Debug Log

Save debug information (up to 2 log files)
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Figure 3.8.2
3.8.3 System Info
General:
Information about hardware version , firmware version and system uptime.
LAN:
Information about hostname, MAC address, IP address,gateway, Primary
DNS and Secondary DNS.
Disk Usage:
Disk usage information.
Memory Usage:
Memory usage information.
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Figure 3.8.3
4. Access MRI
MRI (S50 Recording Interface).Users may access MRI by logging into the MRI
web interface with their username (extension number) and voicemail password.
4.1 Allow users to access MRI

The extensions User Web Interface option must be checked before the
associated user can log into MRI.
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Figure 4-1
4.2 User login

Users can access the MRI web interface by navigating to the S50IP address
using a web browser. If you are unsure of this address, please contact your
network administrator.
www.sahabtec.com
Figure 4-2
4.3 Voicemail
Users can check, delete, move and download voicemail files here.
Figure 4-3
4.4 Record
Users can play, delete and download recorded files here.
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Figure 4-4
4.5 Voicemail Settings

Voicemail password: newvoicemail box password.
Enter again to confirm: confirm new voicemail box password.
Email Address: Email address use to receive the voicemail or Fax.
Note: Please ensure that the section 'SMTP Settings For Voicemail'(in the
'Voicemail Settings') have been properly configured before using this feature.
Enable Voicemail
Once enabled, the voicemail or Faxex will be sent to email as an attachment.
Figure 4-5
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4.6 Settings
You can do some basic setting here. Such as call forwarding, DND, Mobile
Extension Number.
Figure 4-6
5. Access CDR
CDR (S50 Recording Interface).Users may access CDR to research call logs.
Username: cdr
Password: password
5.1 Call Logs

The call Log captures all call details, including call time, caller number, callee
number, call type, call duration,etc. An administrator can search and filter call
data by filter the call logs by call date, caller/callee, trunk, duration, billing
duration, status, communication type.
www.sahabtec.com
Figure 5-1
5.2 Password Settings

Change the password for cdr user.
Figure 5-2
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6. Use S50
PSTN PSTN
FXO
Lines IP
Provider
Route
FXS LAN
r
601 Lines
602
Normal IP Soft
Phone Phone Phone
6.1 Make outbound call
To make an outbound call, we need to add trunk first. There are five types of
VoIP Trunk:
Analog Trunk:FXO ports of S50, connected to a local PSTN.

BRI Trunk: BRI ports of S50, connected to a local PSTN.
VoIP Trunk:Connected to remote VOIP service server.
Service Provider: Connected to service provider server.
What are FXO and FXS?
FXS (Foreign exchange Station) is an interface which drives an analog

telephone or FAX machine. FXS interfaces deliver power, provide ringing, and
use FXO signaling.FXS interfaces are what allow you to hook telephones and
other analog devices to your PBX
FXO (Foreign exchange Office) is an interface that connects to a phone line to

supply your PBX with access to a public telephone network. FXO interfaces use
FXS signaling.FXO interfaces allow you to connect your PBX to real analog
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phone lines.
6.1.1 Sample Routing via PSTN Trunk
Lets route all inside extensions through an analog trunk by dialing 9. In

Outbound Routes, add a new outbound route as below.
Figure6-1
As we can see from the outbound route of pstnout, all phone numbers
starting with 9 will have their first digit stripped off (digit 9) and will be sent
to the PSTN (port 1-2).
After we have configured the above, we can dial 9 + local number to dialout
via a PSTN line.
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Note: Setting number prefix to wild card X and setting Strip to 0 digits from
the front will allow all calls to go through this outbound route.
6.1.2 Sample Routing via VoIP Trunk
Lets configure all inside extensions to dial 0 through the VoIP Trunk.
1. Add VoIP service provider

Before we do add this, please make sure you have a VoIP Trunk account.
TrunksVoIP Trunk SIP Trunk
Enter your account information on this page, and click Save.
Figure6-2
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2. Add Outbound Routes
As we can see from the Outbound Route of voipout, all phone numbers
starting with 0will have their first digit stripped off (digit 0) and will be sent
to the SIP Trunk.
Figure6-3
Nowthat we have added two outbound dialing rules, any call starting with 9
will be routed to the PSTN, and any number starting with 0 will be routed to
the SIP Trunk.
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6.2 Incoming call
6.2.1 Sample Routing to an IVR
Lets configure an incoming call to route to the IVR. In the IVR itself, lets
configure digit 0to route the call to extension 500, and digit 1 to route the
call to extension 501.
. Add IVR
To add a new IVR, go to IVR Create New IVR
Figure6-4
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2. Add Inbound Routes
As we can see from the Inbound Route ofallin, all incoming calls will be sent
to the IVR.
Figure6-5
Figure 1-1
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S50 IP-PBX

SAHAB Technology Co., Ltd.

1.1 FEATURES ............................................................................................ 4

2. SYSTEM SET UP ...................................................................................... 6

2.1 CONNECTION DRAWING ............................................................................ 6

3. MANAGING S50 ....................................................................................... 8

3.1 ADMINISTRATOR LOGIN ............................................................................ 8

3.6.1 LAN Settings .............................................................................. 75

4. ACCESS MRI ......................................................................................... 93

6.1 MAKE OUTBOUND CALL ........................................................................... 99

S50 IP-PBXanIP-PBX for Small Businesses/Home Office

1.2.1 Exterior Appearance

Figure1-1 S50 Front Panel Picture

Figure1-2 S50Back Side Picture

2.1 Connection Drawing

2.2Connecting Ethernet Line

2.3 Supplying Power

3.1 Administrator Login

Note: S50 supports multiple administrators in hierarchical mode

3.2 Status Monitor

3.2.1 Line Status

S50 Status Description:

4 :Extension is busy in a call

3.3.1.1Analog Extensions (FXS)

Check this box if the user should have a voicemail account.

Voicemail Access PIN #

Send Voicemail to Email Address

6) Follow me (Call Forwarding)

User Web Interface

Mobility Extension Number

10) Spy Settings

11) Call Duration Settings

Note: This setting will not be valid for internal calls.

3.3.1.2 VOIP Extension

A VOIP extension is an SIP Account that allows an IP Phone or an IP Soft- Phone

1. Add VOIP Extension

Voicemail Access PIN #

Enable Send Voicemail

5) Follow me (Call Forwarding)

.User Web Interface

Note: for example, if 500 want to monitor extension 501, we need to

DTMF Mode RFC2833, Info, Inband, Auto.

Permitted IP address/Subnet mask

10) Mobility Extension

Mobility Extension Number

Note: If callback is enabled on the inbound route, the mobility extension

11) Call Duration Settings

Note: This setting will not be valid for internal calls.

2. Add Multiple Extensions

Go to Extensions VOIP Extensions Add Multiple Extensions.

3. Edit VOIP Extension

Click Edit on VOIP Extension administration page or click Modify Selected

1. Edit Analog Trunk (FXO)

1. Edit BRI Trunk

PRI Dial Plan

PRI Local Dial Plan

Over Lap Dial

. Add SIP Trunk