Synway SBO Series User manual

Synway SBO Series Gateway
SBO2000
Version 1.8.0
Synway Information Engineering Co., Ltd
www.synway.net

Synway Information Engineering Co., Ltd
SBO2000 Series Gateway User Manual (Version 1.8.0) Page i
Content
Content .................................................................................................................................................. i
Copyright Declaration.......................................................................................................................... iii
Revision History ...................................................................................................................................iv
Chapter 1 Product Introduction .................................................................................................. 1
1.1 Typical Application ................................................................................................................... 1
1.2 Feature List .............................................................................................................................. 1
1.3 Hardware Description .............................................................................................................. 2
1.4 Alarm Info................................................................................................................................. 3
Chapter 2 Quick Guide ................................................................................................................ 5
Chapter 3 WEB Configuration..................................................................................................... 7
3.1 System Login ........................................................................................................................... 7
3.2 Operation Info .......................................................................................................................... 7
3.2.1 System Info ...................................................................................................................... 7
3.2.2 IP Status ........................................................................................................................... 8
3.2.3 Call Monitor ...................................................................................................................... 9
3.2.4 Call Count....................................................................................................................... 10
3.2.5 Warning Info ................................................................................................................... 11
3.3 SIP Settings ........................................................................................................................... 11
3.3.1 SIP.................................................................................................................................. 11
3.3.2 SIP Trunk ........................................................................................................................15
3.3.3 SIP Register ................................................................................................................... 16
3.3.4 SIP Account ....................................................................................................................16
3.3.5 SIP Trunk Group ............................................................................................................. 17
3.3.6 Media Settings................................................................................................................18
3.4 Route Settings ....................................................................................................................... 20
3.4.1 IP to IP............................................................................................................................20
3.5 Number Filter .........................................................................................................................21
3.5.1 Whitelist .......................................................................................................................... 22
3.5.2 Blacklist .......................................................................................................................... 23
3.5.3 Number Pool .................................................................................................................. 23
3.5.4 Filtering Rule ..................................................................................................................23
3.6 Number Manipulation............................................................................................................. 24
3.6.1 IP to IP CallerID .............................................................................................................. 24
3.6.2 IP to IP CalleeID .............................................................................................................25
3.7 System Tools.......................................................................................................................... 25
3.7.1 Network ..........................................................................................................................25
3.7.2 Authorization .................................................................................................................. 26
3.7.3 Management .................................................................................................................. 26
3.7.4 IP Routing Table ............................................................................................................. 27
3.7.5 Access Control ...............................................................................................................27
3.7.6 Centralized Manage ....................................................................................................... 27
3.7.7 SIP Account Generator................................................................................................... 29
3.7.8 Recording Manage ......................................................................................................... 29
3.7.9 Configuration File ...........................................................................................................29
3.7.10 Signaling Capture...........................................................................................................30
3.7.11 Signaling Call Test.......................................................................................................... 30
3.7.12 Signaling Call Track........................................................................................................30
3.7.13 Network Speed Tester ....................................................................................................30
3.7.14 PING Test .......................................................................................................................31
3.7.15 TRACERT Test ...............................................................................................................31

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3.7.16 Modification Record........................................................................................................31
3.7.17 Backup & Upload............................................................................................................ 31
3.7.18 Factory Reset .................................................................................................................31
3.7.19 Upgrade.......................................................................................................................... 32
3.7.20 Account Manage ............................................................................................................32
3.7.21 Change Password .......................................................................................................... 33
3.7.22 Device Lock .................................................................................................................... 33
3.7.23 Restart ............................................................................................................................33
Chapter 4 Typical Applications .................................................................................................34
Appendix A Technical Specifications................................................................................................ 36
Appendix B Troubleshooting ............................................................................................................. 37
Appendix C Technical/sales Support ................................................................................................ 38

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Copyright Declaration
All rights reserved; no part of this document may be reproduced or transmitted in any form or by
any means, electronic or mechanical, without prior written permission from Synway Information
Engineering Co., Ltd (hereinafter referred to as „Synway‟).
Synway reserves all rights to modify this document without prior notice. Please contact Synway
for the latest version of this document before placing an order.
Synway has made every effort to ensure the accuracy of this document but does not guarantee
the absence of errors. Moreover, Synway assumes no responsibility in obtaining permission and
authorization of any third party patent, copyright or product involved in relation to the use of this
document.

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SBO2000 Series Gateway User Manual (Version 1.8.0) Page iv
Revision History
Version
Date
Comments
Version 1.0.0
2015-03
Initial publication.
Version 1.6.2
2015-09
New revision
Version 1.6.3
2016-01
New revision
Version 1.6.4
2016-09
New revision
Version 1.6.5
2017-06
New revision
Version 1.7.0
2018-06
New revision
Version 1.8.0
2020-01
New revision
Note: Please visit our website http://www.synway.net to obtain the latest version of this document.

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SBO2000 Series Gateway User Manual (Version 1.8.0) Page 1
Chapter 1 Product Introduction
Thank you for choosing Synway SBO Series Gateway Products (hereinafter referred to as „SBO
gateway‟)!
The SBO2000 gateway products are used to connect the IP telephony network or IP PBX,
providing such features as transcoding, routing, number filtering, number manipulation, etc.
1.1 Typical Application
Figure 1-1 SBOTypical Application
1.2 Feature List
Basic Features
Description
IP Call
Call initiated from IP to a designated SIP trunk, via routing and number
manipulation.
Number Manipulation
Peels off some digits of a phone number from left/right, or adds a prefix/suffix to a
phone number.
VoIP Routing
Routing path: from IP to IP.
Signaling & Protocol
Description
SIP Signaling
Supported protocol: SIP V1.0/2.0, RFC3261

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Voice
CODEC
G.711A, G.711U, G.729, G722, G723, iLBC, AMR-NB
DTMF Mode
RFC2833, SIP INFO, INBAND, RFC2833+Signaling,
In-band+Signaling
Network
Description
Network Protocol
Supported protocol: TCP/UDP, HTTP, ARP/RARP, DNS, NTP, TFTP, TELNET,
STUN
Static IP
IP address modification support
DNS
Domain Name Service support
Security
Description
Admin Authentication
Support admin authentication to guarantee the resource and data security
Maintain & Upgrade
Description
WEB Configuration
Support of configurations through the WEB user interface
Language
Chinese, English
Software Upgrade
Support of user interface, gateway service, kernel and firmware upgrades based
on WEB
Tracking Test
Support of Ping and Tracert tests based on WEB
SysLog Type
Three options available: ERROR, WARNING, INFO
1.3 Hardware Description
The SBO2000 gateway features 1U rackmount design and integrates embedded LINUX system
within the POWERPC+DSP hardware architecture. It has 2 Kilomega-Ethernet ports (LAN1 and
LAN2) on the chassis.
See the figures below for SBO2000 series‟ appearance:
Figure 1-2 Front View
Figure 1-3 Rear View
LAN1 Indicator
Alarm Indicator
Power 1 Indicator
LAN2 Indicator
Run Indicator
Power 2 Indicator
Reset Button
Power 1 Key
220V AC Power 1
Power 2 Key
LAN
Grounding Stud
220V AC Power 2
Console Port

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Figure 1-4 Left View
The table below gives a detailed introduction to the interfaces, buttons and LEDs illustrated
above:
Interface
Description
LAN
Amount: 2
Type: RJ-45
Bandwidth: 10/100/1000Mbps
Self-Adaptive Bandwidth Supported
Auto MDI/MDIX Supported
Console Port
Amount: 1
Type: RS-232
Baud Rate: 115200 bps
Connector: RJ45, USB
Data Bits: 8 bits
Stop Bit: 1 bit
Parity Unsupported
Flow Control Unsupported
Button
Description
Power Key
Power on/off the SBO gateway. You can turn on the two power keys at the same
time to have the power supply working in the hot-backup mode.
Reset Button
Restore the gateway to factory settings.
LED
Description
Power Indicator
Indicates the power state. It lights up when the gateway starts up with the power
cord well connected.
Run Indicator
Indicates the running status. For more details, refer to Alarm Info.
Alarm Indicator
Alarms the device malfunction. For more details, refer to Alarm Info.
Link Indicator
The green LED on the left of LAN, indicating the network connection status.
ACT Indicator
The orange LED on the right of LAN, whose flashing tells data are being
transmitted.
For the SBO2000 gateway, the console port is connected through a dual male USB cable, and
each USB port is switched to 4 console ports (i.e. 8 console ports in total). For other hardware
parameters, refer to Appendix A Technical Specifications.
1.4 Alarm Info
The SBO2000 gateway is equipped with two indicators denoting the system‟s running status: Run
Indicator (green) and Alarm Indicator (red). The table below explains the states and meanings of
Chassis Vent
Screw Holes for Foot Bracket
Power Vent

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the two indicators.
LED
State
Description
Run Indicator
Go out
System is not yet started.
Light up
System is starting.
Flash
Device is running normally.
Alarm Indicator
Go out
Device is working normally.
Light up
Upon startup: Device is running normally.
In runtime: Device goes abnormal.
Flash
System is abnormal.
Note:
The startup process consists of two stages: System Booting and Gateway Service
Startup. The system booting costs about 1 minute and once it succeeds, both the run
indicator and the alarm indicator light up. Then after the gateway service is successfully
started and the device begins to work normally, the run indicator flashes and the alarm
indicator goes out.
During runtime, if the alarm indicator lights up or flashes, it indicates that the device goes
abnormal. If you cannot figure out and solve the problem by yourself, please contact our
technicians for help. Go to Appendix C Technical/sales Support to find the contact way.

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Chapter 2 Quick Guide
This chapter is intended to help you grasp the basic operations of the SBO gateway in the
shortest time.
Step 1: Confirm that your packing box contains all the following things.
SBO2000 Gateway *1
Rubber Foot Pad *6, Screw for Angle Bracket *8, Gussets*2, Rear Gussets*2,
Grounding Wire*1, Straight-through Shielded Wires*2
220V Power Cord *2
Warranty Card *1
Installation Manual *1
Step 2: Properly fix the SBO gateway.
If you do not need to place the gateway on the rack, simply fix the 6 rubber foot pads. Otherwise,
you should first fix the 2 gussets onto the chassis and then install the chassis on the rack with the
rear gussets.
Step 3: Connect the power cord.
Make sure the device is well grounded before you connect the power cord. Check if the power
socket has the ground wire. If it doesn‟t, use the grounding stud on the rear panel of the device
(See Figure 1-3) for earthing.
Note: Each SBO gateway has two power interfaces to meet the requirement for power supply hot
backup. As long as you properly connect and turn on these two power keys, either power supply
can guarantee the normal operation of the gateway even if the other fails.
Step 4: Connect the network cable.
Step 5: Log in the gateway.
Enter the original IP address (LAN 1: 192.168.1.101 or LAN 2: 192.168.0.101) of the SBO
gateway in the browser to go to the WEB interface. The original username and password of the
gateway are both „admin‟. For detailed instructions about login, refer to System Login. We suggest
you change the initial username and password via „System Tools Change Password‟on the
WEB interface as soon as possible after your first login. For detailed instructions about changing
the password, refer to Change Password. After changing the password, you are required to log in
again.
Step 6: Modify IP address of the gateway.
You can modify the IP address of the gateway via „System Tools Network‟on the WEB interface
to put it within your company‟s LAN. Refer to Network for detailed instructions about IP
modification. After changing the IP address, you shall log in the gateway again using your new IP
address.
Step 7: Check the IP status.
After the configuration of signaling protocols, you can check the channel state via „Operation Info
IP Status‟. Refer to IP Status for detailed introductions.
Step 8: Set routing rules for calls.
Note: For your easy understanding and manipulation, all examples given in this step do not
involve registration.
Step 1: Configure the IP address of the remote SIP terminal which can establish conversations

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with the gateway so that the calls from other terminals will be ignored. Refer to „SIP Settings
SIP Trunk‟for detailed instructions. Fill in „Remote IP‟and „Remote Port‟with the IP
address and port of the remote SIP terminal which will initiate calls to the gateway. You may
use the default values for the other configuration items.
Example: Provided the IP address of the SIP trunk which calls in is 192.168.0.111 and the
port is 5060. Add SIP Trunk 0; set Remote IP to 192.168.0.111 and Remote Port to 5060.
Provided the IP address of the SIP trunk which calls out is 192.168.0.222 and the port is 5060.
Add SIP Trunk 1; set Remote IP to 192.168.0.222 and Remote Port to 5060.
Step 2: Add the SIP trunk configured in Step 1 into the corresponding SIP trunk group. Refer to
„SIP Settings SIP Trunk Group‟for detailed instructions. Select the SIP trunk configured in
Step 1 as „SIP Trunks‟. You may use the default values for the other configuration items.
Example: Add SIP Trunk Group 0. Check the checkbox before 0for SIP Trunks and keep
the default values for the other configuration items; add SIP Trunk Group 1. Check the
checkbox before 1for SIP Trunks and keep the default values for the other configuration
items.
Step 3: Add routing rules. Refer to „Route Settings IPIP‟for detailed instructions. Select SIP
Trunk Group[0] set in Step 2 as „Call Initiator‟and SIP Trunk Group[1] set in Step 3 as „Call
Destination‟. You may use the default values for the other configuration items.
Example: Select SIP Trunk Group[0] as Call Initiator and SIP Trunk Group[1] as Call
Destination. Keep the default values for the other configuration items.
Step 4: Initiate a call from SIP Trunk 0 configured in Step 1 to the IP address and port of the SBO
gateway. Thus you can establish a call conversation via SIP Trunk 1 with the IP terminal.
(Note: The format used for calling an IP address via SIP trunk is as follows: username@IP
address.)
Example: Provided the IP address of the SBO gateway is 192.168.0.101 and the port is 5060.
Provided 123 is a number which conforms to the number receiving rule of the remote device.
Initiate a call from SIP Trunk 0 to the IP address 192.168.0.101 (in the format:
[email protected]68.0.101) and you can establish a call conversation via SIP Trunk 1 to the
number 123.
Special Instructions:
The chassis of the SBO gateway must be grounded for safety reasons, according to
standard industry requirements. A simple way is earthing with the third pin on the plug or
the grounding studs on the machine. No or improper grounding may cause instability in
operation as well as decrease in lightning resistance.
As the device will gradually heat up while being used, please maintain good ventilation to
prevent sudden failure, ensuring that the ventilation holes are never jammed.
During runtime, if the alarm indicator lights up or flashes, it indicates that the device goes
abnormal. If you cannot figure out and solve the problem by yourself, please contact our
technicians for help. Otherwise it may lead to a drop in performance or unexpected
errors.

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Chapter 3 WEB Configuration
3.1 System Login
Type the IP address into the browser and enter the login interface. See Figure 3-1.
Figure 3-1 Login Interface
The gateway only serves one user, whose original username and password are both „admin‟. You
can change the username and the password via „System Tools Change Password‟on the WEB
interface. For detailed instructions, refer to Change Password.
After login, you can see the main interface.
3.2 Operation Info
Operation Info includes the following parts: System Info, IP Status, Call Monitor, Call Count
and Warning Info,showing the current running status of the gateway.
3.2.1 System Info
On the System Info interface, you can click Refresh to obtain the latest system information. See
below for details.
Item
Description
MAC Address
MAC address of LAN 1 or LAN 2.
IP Address
The three parameters from left to right are IP address, subnet mask and default
gateway of LAN 1 or LAN 2.
IPV6 Address
IPV6 address.
DNS Server
DNS server address of LAN 1 or LAN 2.

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Receive/Transmit
Packets
The amount of receive/transmit packets after the gateway‟s startup, including
three categories: All, Error and Drop.
Current Speed
The current speed of data receiving and transmitting.
Work Mode
The work mode of the network, including six options: 10 Mbps Half Duplex, 10
Mbps Full Duplex, 100 Mbps Half Duplex, 100 Mbps Full Duplex, 1000 Mbps Full
Duplex and Disconnected.
Network Type
The type of the network, including three options: Static, DHCP and PPPoE.
Runtime
Time of the gateway keeping running normally after startup. This parameter
updates every 2s.
CPU Temperature
Display the real time temperature of the CPU. The first is the temperature of the
Master and the last four are the temperature of the Slaver.
CPU Usage Rate
Display the real time usage rate of the CPU.
Current RTP Message
Data
Display the receiving and sending information of the current RTP data.
DCMS Working Status
Display the connecting status of the gateway and DCMS.
Recording Work
Status
Display the working status of the recording server docked by the gateway.
Serial Number
Unique serial number of an SBO gateway.
WEB
Current version of the WEB interface.
Gateway
Current version of the gateway service.
Uboot
Current version of Uboot.
Kernel
Current version of the system kernel on the gateway.
Firmware
Current version of the firmware on the gateway.
3.2.2 IP Status
The IP status interface shows the real-time status of each IP channel on the gateway. See below
for details.
Item
Description
Channel No.
Number of the IP channel on the device.
Status
Displays the channel state in real time. You can move the mouse onto the channel
state icon for detailed information about the channel and the call, such as: call
direction, calling party number and called party number. The channel states include:
State
Icon
Description
Idle
The channel is available.
Wait Answer
The channel receives the ringback tone and is waiting
for the called party to pick up the phone.
Ringing
The channel is in the ringing state.
Talking
The channel is in a conversation.
Pending
The channel is in the pending state
Dialing
The channel is dialing.
Wait Message
The channel is waiting for the message from remote
end.

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Note: The gateway provides the fuzzy search feature on this interface. After you click any
characters on the interface, and press the „F‟button, the search box will emerge on the right top of
this page. Then you can input the key characters and the gateway will locate the channel on
which there is an ongoing call that conforms to the fuzzy search condition.
Take an example: As shown in Figure 3-2, after we input the character 111 to the search box, and
click the Search button, the gateway does a fuzzy search and locates that the ongoing call whose
CalledID contains the character 111 occurs on Channel 2 and Channel 3 of Channel Group 0.
Figure 3-2 Search Calls
3.2.3 Call Monitor
On the Call Monitor interface, you can set a condition for call monitoring. The table below explains
the items on this interface.
Item
Description
Monitored CallerID,
Monitored CalleeID,
Monitored Remote
Address
Sets the condition for the call monitoring. You can set to monitor the calls by
CallerID, CalleeID or remote address.
Monitoring LAN Port
Selects the LAN port which is used to monitor the calls.
Channel No.
The number of the channel, which starts from 0.
Call Direction
The direction of the monitored call, including two options: IPPSTN and
PSTNIP.
Channel Status
The status of the channel which the monitored call locates at.
CallerID
The CallerID of the monitored call.
CalleeID
The CalleeID of the monitored call.
Start Time
The start time of the monitored call.
Duration
The duration of the monitored call.
Click the icon in the channel status column, and you can monitor the call in real-time. If your
computer is not installed with RemoteListener, click the icon and you will see a prompt asking you
to set the security level. Follow the instructions to configure the IE explorer: Open it and
click 'Tools > Internet Options >Security Tab'; then click 'Custom Level' and enable 'Initialize and

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script ActiveX controls not marked as safe for scripting'. If there is a shadow showing under the
icon, such as „ ‟, it means the monitoring goes successful. Click the icon again to cancel
the monitoring.
Note: If a channel has been monitored from the very beginning, the monitoring, even if not yet
cancelled, will terminate once the channel is removed from the monitor list.
3.2.4 Call Count
The Call Count interface lists the detailed information about all the calls counted from the startup
of the gateway service to the latest open or refresh of this interface. You can click Reset to count
the call information again, and click Download to download all the call logs. The table below
explains the items on this interface.
Item
Description
SIP Trunk
Address of the SIP trunk, i.e. the IP address or domain name of the remote SIP
terminal which will establish a call conversation with the gateway.
Description
More information about each SIP trunk group.
Current Number of
IP Call in
The number of calls currently coming in to the IP.
Connected Number
of IP Call in
The number of successfully connected calls coming in to the IP.
Total Nmuber of IP
Call in
The sum of all calls coming in to the IP.
Connection Rate of
IP Call in
The percentage of the number of successful IP incoming calls to the total number of
incoming calls.
Current Number of
IP Call out
The number of calls currently going out from the IP.
Connected Number
of IP Call out
The number of successfully connected calls going out from the IP.
Total Nmuber of IP
Call out
The sum of all calls going out from the IP.
Connection Rate of
IP Call out
The percentage of the number of successful IP outgoing calls to the total number of
outgoing calls.
Average Call Length
The average call length for all connected calls.
CPS
The number of new calls per second.
Release Cause
Reason to release the call.
Normal
Disconnection
Total number of the calls which are normally cleared.
Cancelled
Total number of the calls which are cancelled by the calling party.
Busy
Total number of the calls which fail as the called party has been occupied and
replies a busy message.
No Answer
Total number of the calls which fail as the called party does not pick up the call in a
long time or the calling party hangs up the call before the called party picks it up.
Routing Failed
Total number of the calls which fail because no routing rules are matched.
No Idle Resource
Total number of the calls which fail because no voice channel is available.

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Failed
Total number of the calls which fail as the called party number does not conform to
the number-receiving rule or for relative reasons.
Others
Total number of the calls which fail due to other unknown reasons.
Number
Count the number of channels in each state.
Percentage
The percentage of the calls with a release cause to total calls.
3.2.5 Warning Info
The Warning Information interface displays all the warning information on the gateway.
3.3 SIP Settings
SIP Settings includes six parts: SIP, SIP Trunk, SIP Register, SIP Account, SIP Trunk Group
and Media. SIP is used to configure the general SIP parameters; SIP Trunk is used to set the
basic and register information of the SIP trunk; SIP Register is used for the registration of SIP;
SIP Account is used for registering SIP accounts to the SIP server; SIP Trunk Group is to
manage SIP trunks by group; and Media is to set the RTP port and the payload type.
3.3.1 SIP
On the SIP Settings interface, you can configure the general SIP parameters. After configuration,
click Save to save your settings into the gateway or click Reset to restore the configurations. If a
dialog box pops up after you save your settings asking you to restart the service, do it immediately
to apply the changes. Refer to Restart for detailed instructions. The table below explains the items
on this interface.
Item
Description
SIP Address of WAN
IP address of WAN for SIP signaling, using LAN 1 by default.
SIP Signaling Port
The port monitored by SIP UDP/TCP. The value range is 5001-65535 and the
default value is 5060.
Note: It cannot overlap with the RTP port range in the media settings.
Recv 180 and Send
Tone
The gateway receives a 180SIP message and sends a custom ringback tone to
the caller. By default it is disabled.
Recv 183 and Stop
Tone
After receiving the 180 message, the gateway receives the 183 message and
stops the ringback tone. It is effective only when the feature Recv 180 and Send
Tone is enabled.
Send 100rel
Sets whether to send the 100rel field with the 180/183 message. The default
setting is disabled.
Hide CallerID
Sets whether to hide the CallerID, with the default value of Not Hidden.
Obtain CallerID from
There are four optional ways to obtain the calling party number: Username of
“From” Field, Displayname of “From”Field, P-Preferred-Identity Field,
P-Asserted-Identity Field. The default value is Username of “From”Field.
Obtain/Send CalleeID
from
There are two optional ways to obtain or send the called party number: from “To ”
Field or from “Request” Field. The default value is from “Request” Field.
Asserted Identity
Mode
Sets whether to have the invite message include some header information, two
options available now: P-Asserted-Identity and P-Preferred-Identity. The default
value is disabled.

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Number in From Field
not Manipulated
Once this feature is enabled, the callerID in the From field will not be manipulated,
with the default value of disabled.
Note: It is valid only when the configuration item Asserted Identity Mode is
enabled.
Prack Send Mode
Sets whether to return the prack message while receiving the 180/183 message
which carries the 100rel field. Three options are available: Disable, Supported and
Require, and the default setting is Disable.
NAT Traversal,
Traversal Type
Sets whether to enable the feature of NAT Traversal. By default, the feature is
disabled. There is only one optional traversal type: Port Mapping.
LAN1 Mapping
Address, LAN2
Mapping Address
The mapping address of the LAN1 and LAN2 in case the NAT traversal is
enabled. If the port mapping is selected as the traversal type, you are required to
set the mapping address on the router and fill in the corresponding information
here as well.
LAN1 Mapping
Address (RTP), LAN2
Mapping Address
(RTP)
The RTP mapping address of the LAN1 and LAN2 in case the NAT traversal is
enabled.
Always Use Mapping
Address
Once this feature is enabled, the gateway will be enforced to use the mapping
address set in the above configuration item to initiate calls. By default it is
disabled.
RTP Self-adaption
When this feature is enabled, the RTP reception address or port carried by the
signaling message from the remote end, if not consistent with the actual state, will
be updated to the actual RTP reception address or port. By default, this feature is
disabled.
UDP Header
Checksum
When this feature is enabled, the gateway will automatically calculate the check
sum of the UDP header during RTP transmission.
Rport
When this feature is enabled, the gateway will automatically add a corresponding
Rport field to the Via message of SIP. By default, it is disabled.
CPS
The number of new calls per second. By default it is 600.
Auto Reply of Source
Address
Once this feature is enabled, the gateway will reply the source address in the
invite message. The default value is disabled.
Calling Concurrency
Limit
Limit on call times of the same calling number. Once it is exceeded, the gateway
will reply 503.
Calling Concurrent
Number
Allowed call times of the same calling number.
Registration Related
Settings
When this feature is enabled, the available call time for each SIP registered
account as well as the SIP Registered Number Polling feature can be set. By
default it is disabled.
Time (min/month)
Specifies the call time for a SIP registered account.
SIP Registered
Number Polling
When this feature is enabled, the call is polled among SIP registered accounts. By
default it is disabled.

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Failed Count
It is valid only when the feature SIP Registered Number Polling is enabled. After a
number is called out and fails for set times, it will be kicked out of the cycle and
then allowed to re-join after Recover Time of Disable Account.
Recover Time of
Disable Account (m)
See the description of Failed Count.
Caller Prefix Grouping
When this feature is enabled, only if the calling number of the call matches the
caller prefix on the page of the SIP registered account will the rated time be used.
Caller over Clocking
(IP OUT)
Limit on the number of calls in a cycle for the calling number. By default this
feature is disabled.
Cycle (min)
The time of a cycle. It is only valid when the feature Caller over Clocking is
enabled.
Count Values
The allowed incoming calls within the set time of a cycle. It is only valid when the
feature Caller over Clocking is enabled.
Interval (ms)
The interval time for calls from a same calling number. After hangup, the gateway
needs to wait for some time before using this account. It is only valid when the
feature Caller over Clocking is enabled.
SIP Value of Reply
The abnormal SIP message returned by the gateway. The default value is 503.
SIP Account Numbers
The maximum number of SIP accounts must be set greater than the number of
existing SIP accounts.
SIP Account
Registration Interval
The interval between registrations of multiple SIP accounts. Range of value:
0~10000, with the default value of 0.
DSCP
Sets whether to enable the DSCP differentiated services code point. By default, it
is disabled.
Voice Media
Sets the priority of the voice media for DSCP. The voice media with a bigger value
has a higher priority. The value range is 0~63, with the default value of 46.
Signal Control
Sets the priority of the signal control for DSCP. The signal control with a bigger
value has a higher priority. The value range is 0~63, with the default value of 26.
Calls from SIP Trunk
Address only
Once this feature is enabled, the gateway will only accept the calls from the IP
addresses set in SIP Settings SIP Trunk. By default, it is disabled.
Match Call Count to
SIP Trunk based on
Source Address of
INVITE
Performs call count by matching the source address of the INVITE message. By
default it is disabled.
Hang up upon Call
Time-out
Sets whether to enable the feature to hang up the call once it is time-out, with the
default value of No.
Maximum Call
Overtime
Sets the maximum overtime for a call. Calculated by minute.
Working Period,
Period
The work period for the gateway. You can specify a certain period for the gateway
to make calls. By default, the gateway is allowed to make calls any time in the day
(24 Hours).

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Session Timer
Sets whether to enable the session refresh feature, with the default value of
disabled. Once this feature is enabled, you are required to enter the minimum
time and the timeout value.
Minimum Time
Sets the minimum time for refreshing the session. Value of range: 90~65535, with
the default value of 150.
Timeout
Sets the timeout value for refreshing the session. The value cannot be less than
that of Minimum Time, with the default value of 600.
Sip Trunk Heart
Sets whether to send the option message to the SIP trunk. The calls routed to this
trunk will be rejected directly if the times of no answer from the MGCF trunk
exceed the set value.
Trunk Heartbeat Cycle
The cycle to send the option message to the SIP trunk.
Trunk Heartbeat
Allowed Times of
NoResponse
The allowed times of SIP‟s no answer to the option message.
Early Media
Once this feature is enabled, the P-Early-Media field will be included in the Invite
message. The default value is disabled.
Early Session
Once this feature is enabled, the early-session field will be included in the Invite
message. The default value is disabled.
Support 100rel
Sets whether to carry 100rel in the Supported field of the request message for IP
calls out. By default it is disabled.
Not Wait ACK after
Sending 200 OK
Once this feature is enabled, the gateway does not need to wait the ACK
message after sending the 200OK message. The default value is disabled.
Match SIP Trunk Port
Sets whether to search SIP trunks by matching port number for IP calls in. By
default it is disabled.
The Percentage of
Registration Message
Sending Cycle to
Period of Validity
Sets the percentage of the sending cycle of the SIP registration message to the
validity period. Value of range: 1~200, with the default value of 70.
Maximum Wait
Answer Time
Sets the maximum time for the SIP channel to wait for the answer from the called
party of the outgoing call it initiates. If the call is not answered within the specified
time period, it will be canceled by the channel automatically. The default value is
60, calculated by s.
Maximum Wait RTP
Time
Sets the maximum time for the SIP channel to wait for the RTP packet. If no RTP
packet is received within the specified time period, the channel will enter the
pending state automatically and release the call. The default value is 0, calculated
by s.
Add Content to To
Field in INVITE
Message
Once this feature is enabled, “user=phone”will be added to the TO field of the
INVITE message. The default value is disabled.
Add Content (‘=’
replaced by ‘+’)
Sets the content to add to the TO field.
UserAgent Field
Sets the content of the UserAgent field. Currently, it only supports the English
uppercase and lowercase letters.

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3.3.2 SIP Trunk
On the SIP trunk settings interface, there is no SIP trunk information by default. A new SIP trunk
can be added by the Add New button on the bottom right corner of the list
The table below explains the items shown on the interface.
Item
Description
Index
The unique index of each SIP trunk.
Description
More information about each SIP trunk group.
Remote Address
Address of the SIP trunk, i.e. the IP address or domain name of the remote SIP
terminal which will establish call conversation with the gateway.
Remote Port
Port of the SIP trunk.
Local Network Port
The network port where the SIP trunk locates.
Local SIP Port
The local signaling port where the SIP trunk locates.
Display CODEC
Used to hide or unhide the CODECs with the packing time.
Outgoing Voice
Resource
Maximum number of voice channels for the outgoing calls allocated by the gateway
to the SIP trunk.
Incoming Voice
Resource
Maximum number of voice channels for the incoming calls allocated by the SIP
trunk to the gateway.
Working Period,
Period
The work period for the gateway. You can specify a certain period for the gateway
to make calls. By default, the gateway is allowed to make calls any time in the day
(24 Hours).
VOS1.1 SIP
Encryption
Sets whether to perform VOS1.1 encryption for SIP signaling, including three
modes: No Encryption, Gateway Encryption, and Client Encryption. The
default setting is No Encryption.
Encrypt Key
A key that encrypts SIP signaling.
VOS1.1 RTP
Encryption
Sets whether to perform VOS1.1 encryption for RTP. By default it is disabled.
Externally Bound
Enable
Sets whether to enable the Proxy feature. Once it is enabled, SIP messages will be
sent to the proxy address.
Externally Bound
Address
The proxy address.
Externally Bound
Port
The proxy port.
After configuration, click Save to save the settings into the gateway or click Close to cancel the
settings.
Click Modify to modify a SIP account. The configuration items on the SIP account modification
are the same as those on the Add New SIP Account interface.
To delete a SIP account, check the checkbox before the corresponding index and click the Delete
button. Check All means to select all available items on the current page; Uncheck All means to
cancel all selections on the current page; Inverse means to uncheck the selected items and
check the unselected. To clear all SIP accounts at a time, click the Clear All button.
Note: If no SIP trunk is configured, the configuration items such as SIP Register and SIP Trunk
Group will not be available.