IAD Series FXS VoIP Gateway

IAD-16S VoIP Gateway User Manual
IAD-16S VoIP Gateway User Manual
note : same document is applicable for IAD-24FXS (24S)
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IAD-16S VoIP Gateway User Manual
Revision Record
File Name
IAD-16S VoIP Gateway User Manual
Document Version
V2.2
Firmware Version
1.18.02.20
Date
2016/02/16
Revised by
Technical Support Department
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IAD-16S VoIP Gateway User Manual
Preface
Welcome
Thanks for choosing IAD-16S VoIP Gateway! We hope you will make optimum use of this flexible,
rich-feature VoIP-to-FXS gateway. Please read this document carefully before install the gateway.
About this manual
This manual provides information about the introduction of the gateway, and about how to install,
configure or use the gateway.
For interoperability with different IPPBX/Softswitch platform, you can refer to relevant configuration
guide of different systems.
This manual is written with reference to the default configurations of the IAD-16S VoIP Gateway.
Intended audience
This manual is aimed primarily at network and system engineers who will install, configure and
maintain the gateway.
System engineers are persons who customize the configurations to meet the requirements of users.
Parts of the document containing description of telephony features are aimed at users who are the
persons who will actually use the gateway.
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Contents
1 Introduction of IAD-16S . ............................................................................................................. 1
1.1 Overview. ......................................................................................................................................... 1
1.2 Equipment Appearance .................................................................................................................... 1
1.3 Ports and Connector......................................................................................................................... 2
1.4 Functions and Features .................................................................................................................... 3
1.4.1 Protocol standard supported ........................................................................................................ 3
1.4.2 Voice and Fax parameters ............................................................................................................. 3
1.4.3 Supplementary service .................................................................................................................. 3
2 Basic Operations .......................................................................................................................... 5
2.1 Methods to Number Dialing ............................................................................................................. 5
2.2 Direct IP Calls .................................................................................................................................... 5
2.3 Call Holding. ..................................................................................................................................... 6
2.4 Call Waiting. ..................................................................................................................................... 6
2.5 Call Transfer. .................................................................................................................................... 6
2.5.1 Blind Transfer ................................................................................................................................ 6
2.5.2 Attended Transfer ......................................................................................................................... 6
2.6 Three-way Calling .............................................................................................................................. 7
2.7 Description of Feature Codes ............................................................................................................ 7
2.8 Sending and Receiving Fax ................................................................................................................ 8
2.8.1 T. 38 and Pass-Through ................................................................................................................. 9
2.9 Local IVR Operation ........................................................................................................................... 9
2.9.1 Inquire IP address .......................................................................................................................... 9
2.9.2 Factory Reset ................................................................................................................................. 9
2.9.3 Configure LAN Port’s IP Address ................................................................................................... 9
3 Configurations on Web Interface. .............................................................................................. 11
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3.1 Network Connection ....................................................................................................................... 11
3.2 Preparations for Login ..................................................................................................................... 11
3.3 Log in Web Interface ....................................................................................................................... 12
3.4 Navigation Tree ............................................................................................................................... 13
3.5 State and Statistics .......................................................................................................................... 13
3.5.1 System Information ..................................................................................................................... 13
3.5.2 Registration Information ............................................................................................................. 16
3.5.3 TCP/UDP Statistics ....................................................................................................................... 16
3.5.4 RTP Session Statistics .................................................................................................................. 17
3.5.5 CDR Statistics ............................................................................................................................... 17
3.6 Quick Setup Wizard ......................................................................................................................... 17
3.7 Network Configuration .................................................................................................................... 17
3.7.1 Local Network.............................................................................................................................. 17
3.7.2 VLAN(Virtual Local Area Network)......................................................................................... 18
3.7.3 DHCP Option................................................................................................................................ 20
3.7.4 Qos. ............................................................................................................................................. 20
3.7.5 LAN Qos . ..................................................................................................................................... 21
3.7.6 ARP . ............................................................................................................................................ 21
3.8 SIP Server . ....................................................................................................................................... 21
3.9 Port . .............................................................................................................................................. 24
3.10 Advanced . ..................................................................................................................................... 26
3.10.1 FXS/FXO Parameters ................................................................................................................. 26
3.10.2 Media Parameter ...................................................................................................................... 28
3.10.3 SIP Parameters .......................................................................................................................... 30
3.10.4 Fax Parameter ........................................................................................................................... 35
3.10.5 Digit Map ................................................................................................................................... 36
3.10.6 Feature Codes............................................................................................................................ 37
3.10.7 System Parameter ..................................................................................................................... 38
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3.10.8 Action URL ................................................................................................................................. 40
3.11 Call & Routing ................................................................................................................................ 41
3.11.1 Wildcard Group ......................................................................................................................... 41
3.11.2 Port Group ................................................................................................................................. 41
3.11.3 IP Trunk. .................................................................................................................................... 43
3.11.4 Routing Parameter .................................................................................................................... 43
3.11.5 IP -> Tel Routing ........................................................................................................................ 44
3.11.6 Tel-IP/Tel Routing ...................................................................................................................... 45
3.11.7 IP – IP Routing ........................................................................................................................... 46
3.12 Manipulation Configuration ......................................................................................................... 46
3.12.1 IP -> Tel Callee ........................................................................................................................... 47
3.12.2 Tel -> IP/Tel Caller ..................................................................................................................... 48
3.12.3 Tel-IP/Tel Callee......................................................................................................................... 49
3.13 Routing rule examples ................................................................................................................... 49
3.13.1 Route any calls from any IP to specific port . ............................................................................. 49
3.13.2 Route any calls from any IP to specified port group . ................................................................ 50
3.13.3 Route any calls from any port to specific SIP IP trunk. .............................................................. 51
3.14 Maintenance ................................................................................................................................. 53
3.14.1 TR069. ....................................................................................................................................... 53
3.14.2 SNMP (Simple Network Management Protocol). ...................................................................... 53
3.14.3 Syslog . ....................................................................................................................................... 55
3.14.4 Provision .................................................................................................................................... 57
3.14.5 Cloud server .............................................................................................................................. 58
3.15 User Manage ................................................................................................................................ 58
3.16 Remote Server .............................................................................................................................. 59
3.17 Record Parameter ........................................................................................................................ 59
3.18 Security . ....................................................................................................................................... 59
3.18.1 WEB ACL .................................................................................................................................... 59
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3.18.2 Telnet ACL.................................................................................................................................. 60
3.18.3 Passwords .................................................................................................................................. 60
3.18.4 Encrypt . .................................................................................................................................... 61
3.19 Tools . ........................................................................................................................................... 61
3.19.1 Firmware upload ....................................................................................................................... 61
3.19.2 Data Backup............................................................................................................................... 62
3.19.3 Data Restore .............................................................................................................................. 63
3.19.4 Ping Test .................................................................................................................................... 63
3.19.5 Tracert Test ............................................................................................................................... 64
3.19.6 Outward Test ............................................................................................................................. 65
3.19.7 Network Capture ....................................................................................................................... 66
3.19.8 Factory Reset ............................................................................................................................. 70
3.19.9 Device Restart ........................................................................................................................... 71
4 Glossary .................................................................................................................................... 72
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1 Introduction of IAD-16S
1.1 Overview
IAD-16S VoIP gateway provides voice services based on IP network. It’s a cost-effective and flexible
solution for SOHO (Small Office-Home office), remote office, medium-sized enterprise and enterprise
with multiple branches.
The gateway connects to analog telephone, fax and traditional analog PBX with standard voice
interfaces and provides high quality voice service.
The gateway, based on standard SIP protocol is compatible with leading IP PBX, soft-switch and
SIP-based platform.
The FXS analog gateway available in the following configurations:
Model
IAD-16S
Voice Channels
16
FXS Ports
16
Physical Port Labels
0-15
For detailed hardware and software features, please refer to “product specifications”.
1.2 Equipment Appearance
Front View
Back View
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1.3 Ports and Connector
Port Name
Connector
Description
Power Jack
Power Jack
To connect 100-240V AC 50-60HZ power supply
LAN Port
RJ45
to connect to the IP network over a DSL modem or Router or
a LAN switch
FXS Ports 0-15
RJ11
FXS ports to connect standard analog phone or FAX machine
or a PBX
Console Port
RJ48
Console port is used to carry out maintenance-related
configurations
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1.4 Functions and Features
1.4.1 Protocol standard supported
•
SIP V2.0 (RFC 3261,3262,3264)
•
SDP (RFC 2327)
•
REFER (RFC 3515)
•
RTP/RTCP (RFC 1889,1890)
•
STUN (RFC 3489)
•
ARP/RARP (RFC 826/903)
•
SNTP (RFC 2030)
•
TFTP/HTTP/HTTPS
•
DNS/DNS SRV (RFC 1706/RFC 2782)
•
VLAN 802.1P/802.1Q
1.4.2 Voice and Fax parameters
•
G.711A/U law, G.723.1, G.729AB,iLBC,AMR
•
Comfortable Noise Generation (CNG)
•
Voice Activity Detection (VAD)
•
Echo Cancellation (G.168)
•
Adaptive Dynamic Jitter Buffer
•
Voice and fax gain control
•
Modem
•
T.38/Pass-through
•
DTMF Mode: Signal/RFC2833/INBAND
1.4.3 Supplementary service
•
Call waiting
•
Call transfer(Blind transfer, Attend transfer,)
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•
Quick pickup
•
Call Forwarding Unconditional
•
Call Forwarding on No Reply
•
Hotline
•
Call hold
•
DND
•
Three-way calling(1/2/4 port support)
•
Voice mail
•
Direct IP Call
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2 Basic Operations
2.1 Methods to Number Dialing
Dial mobile phone or extension number
Dial the number directly and wait for 3 seconds (Default “No dial timeout”);
Dial the number directly and press #.
2.2 Direct IP Calls
The IAD-16S gateway allows users to directly call through IP address. Under this circumstance, the
user only needs an analog phone which is connected to a FXS port of the gateway, and calls can be
established without register.
Calls can be established through IP address as long as one of the following conditions is met.
Both the IAD-16S and other VoIP device have public IP addresses;
The IAD-16S and other VoIP device use private IP addresses of a same LAN;
The IAD-16S and other VoIP device can be connected through a router and use public or private IP
addresses (with necessary port forwarding or DMZ).
Operation Process:
Step1: Pick up the analog phone and then dial “*47”;
Step2: Enter the target IP address.
【Note】: No dial tone will be played between step 1 and step 2
Example:
Assume that the target IP address is 192.168.0.160, user need to dial *47 and then 192*168*0*160.
After that, press the “#” key or wait 3 seconds. Then signaling interaction is completed and ringing
can be heard .
【Note】:You cannot make direct IP calls between FXS0 to FXS1 of a same IAD-16S since they are
using same IP addresses. Call through IP address is only routed to the default destination port 5060.
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2.3 Call Holding
Place a call on hold by pressing the “flash” button on the analog phone (if the phone has the button).Press
the “flash” button again to release the previously held caller and resume conversation. If no “flash”
button is available, use “hook flash” instead.
2.4 Call Waiting
If a calling party places a call to a called party which is otherwise engaged, and the called party has
the call waiting feature enabled, the calling party will hear a IVR voice ‘Please hold on,
the subscriber you dialed is busy’ and the called party will hear three beeps.
By pressing the flash button or the flash hook, the called party is able to switch between the new
incoming call and the current call.
2.5 Call Transfer
2.5.1 Blind Transfer
Blind transfer is used to transfer call to a third party without informing the caller. Assume that A and
B are in a conversation. A wants to blind Transfer B to C:
A presses FLASH on the analog phone to hear the dial tone;
Then A dials *87 and C’s number and # (or wait for 4 seconds);
A will hear the confirm tone. Then, A hangs up, and B and C enter into a conversation.
Note:
“Call features enable” must be set to “Yes” on WEB configuration page. Caller A can place a call on
hold and wait for one of the three situations:
A quick confirmation tone (similar to call waiting tone) which follows the dial tone. This indicates
the transfer is successful. At this point, Caller A can either hand up or make another call.
A quick busy tone which follows a restored call (on supported platforms only). This means the
transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy
tone indicates the transfer has failed.
Continuous busy tone. This means the call has timed out.
2.5.2 Attended Transfer
Attended transfer allows the transferring party either connects the call to a ringing phone (ringback
heard) or speaks with the third party before transferring the call to the third party.
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Assume that A and B are in conversation. Caller A wants to attended transfer B to C:
A presses FLASH on the analog phone and wait for dial tone;
Then dial C’s number followed by # (or wait for 3 seconds);
If C answers the call, A and C are in conversation. Then A can hang up to complete the transfer;
If C does not answer the call, A can press “flash” to resume call with B.
2.6 Three-way Calling
Three-way calling:
A calls B,B picks up the phone, then A and B enters into conversation;
A presses the hook flash, and the call between A and B is placed on hold. Then C calls A and A
answers the call.
A presses hook flash again, then the calls between A and B and between A and C are placed on
hold. At this time, if A presses 1 , conversation between A and B is resumed; if A presses 2, conversation
between A and C is resumed; if A presses 3, A,B and C enter into conversation.
2.7 Description of Feature Codes
The IAD-16S gateway supports all traditional and senior phone function. It provides feature codes for
easy maintenance and easy entry to phone functions.
Feature Codes
Corresponding Function
*158#
Dial *158# to inquiry the IP address of LAN port
*114#
Dial *114# to inquire port account
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*150*
Dial *150* to set the way of obtaining IP address
*157*
Dial *157*0 to set route mode; dial *157*1 to set bride mode
*152*
Dial *152* to set IPv4 address
*153*
Dial *153* to set subnet mask
*156*
Dial *156* to set default gateway’s IP address
*193#
Dial *193# to renew the IP address
*166*000000#
Dial *166*000000# to reset to factory defaults
*111#
Dial *111# to restart the gateway
*#
Dial *# to place a call on hold
*47*
Dial *47* to establish a call through IP address
*51#
Dial *51# to enable ‘call waiting’ feature
*50#
Dial *50# to disable ‘call waiting’ feature
*87*
Dial *87* to blind transfer a call
*72*
Dial *72* to enable ‘unconditional call forwarding’ feature
*73#
Dial *73# to disable ‘unconditional call forward’ feature
*90*
Dial *90* to enable ‘busy call forwarding’ feature
*91#
Dial *91# to disable ‘busy call forwarding’ feature
*92*
Dial *92* to enable ‘no answer call forwarding’ feature
*93#
Dial *93# to disable ‘no answer call forwarding’ feature
*78#
Dial *78# to enable DND
*79#
Dial *79# to disable DND
*200#
Dial *200# to access voice mail
Flash/Hook
Used to switch between incoming calls. If the phone is not in session,
flash/hook will switch a new channel for a new call.
2.8 Sending and Receiving Fax
The IAD-16S gateway supports four fax modes:
T.38 (FoIP)
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Pass-Through
Modem
Adaptive
2.8.1 T. 38 and Pass-Through
T.38 is the preferred fax mode because it is more reliable and works well in most network conditions.
If the service provider supports T.38, please use this method by selecting T.38 as fax mode (default).
If the service provider does not support T.38, pass-through mode may be used. If you have problems
with sending or receiving Fax, toggle the Fax Tone Detection Mode setting.
2.9 Local IVR Operation
2.9.1 Inquire IP address
Connect analog phone to FXS ports of the IAD-16S gateway, then pick up the phone. After dialing
tone, dial *158# to inquire the IP address of LAN port.
2.9.2 Factory Reset
Pick up the phone, and then dial *166*000000#. After hearing a voice prompt of ‘setting successfully’,
hang up the phone and the gateway is reset to factory defaults.
2.9.3 Configure LAN Port’s IP Address
Before configuration, please ensure:
The gateway is power on;
Device has been connected to network;
Telephone is connected to FXS port of the IAD-16S gateway.
Configure dynamic IP address by DHCP:
Pick up the phone, dial *150*2# and then hang up the phone.
If the voice prompt indicates ‘setting successfully’, please restart the gateway after 10 seconds.
Configure Static IP address:
Take the configuration of IP address ‘172.16.0.100’ as example.
Pick up the phone, dial *150*1# and then hang up the phone.
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Then configure IP address and mask as follow:
• Configure IP address
Pick up the phone, dial *152*172*16*0*100# and then hang up the phone.
• Configure subnet mask
Pick up the phone, dial *153*255*255*0*0# and then hang up the phone.
• Configure gateway IP address
Pick up the phone, dial *156*172*16*0*1# and then hang up the phone.
• Query the IP address of the IAD-16S gateway:
Pick up the phone, dial *158#.
If the gateway uses PPPoE method to get IP address,the IP address needs to be configures through
web browser.
【Note】:The telephone will play voice prompt “setting successfully” if the step is correct。
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3 Configurations on Web Interface
3.1 Network Connection
Connect the IAD-16S gateway to the network according to the following network topology, and dial
*158 to query the IP address of the gateway.
DSL Modem/ Router
LAN Switch/ Hub
IAD-16S
IP Network
PC
3.2 Preparations for Login
Modify the IP address of the PC to make it at the same network segment with the IAD-16S device,
since the default IP address of the gateway is 192.168.11.1.
Take Windows 7 as an example, the IP address of PC is changed into 192.168.11.10:
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Check the connectivity between the PC and the gateway. Click Start  Run of PC and enter cmd to
execute ‘ping 192.168.11.1’ to check whether the IP address of the IAD-16S gateway runs normally.
3.3 Log in Web Interface
Open a web browser and enter the IP address of the LAN port of the IAD-16S (the default IP of LAN
port is 192.168.11.1). Then the login GUI will be displayed. Both the default username and password
are admin.
It is advised to modify the username and password for security consideration.
Enter default username and password: admin/admin, then click “Log in” to enter into the Web
interface. And then you can see the following web interface.
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3.4 Navigation Tree
The web management system of the IAD-16S VoIP gateway consists of the navigation tree and
detailed configuration interfaces.
Choose a node of the navigation tree to enter into a detailed configuration interface.
3.5 State and Statistics
3.5.1 System Information
On the System Information interface, you can view the information of device ID, MAC address,
network mode, IP addresses, version information, sever register status and so on.
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Figure 3.5-1 System Information
Explanation of items on System Information interface
Device ID
A unique ID of each device. This ID is used for warranty and cloud server authentication.
MAC address
Hardware address of the LAN port
The IP address of the gateway is shown.
DHCP: Obtain IP address automatically. IAD-16S is regarded as a DHCP client, which sends a
IP Address
broadcast request and looks for a DHCP server from the LAN to answer. Then the first
discovered DHCP server automatically assigns an IP address to the IAD-16S from a defined
range of numbers.
Static IP Address: Static IP address is a semi-permanent IP address and remains associated
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with a single computer over an extended period of time. This differs from a dynamic IP
address, which is assigned ad hoc at the start of each session, normally changing from one
session to the next.
If you choose static IP address, you need to fill in the following information:
 IP Address: the IP address of the LAN port of the IAD-16S;
 Subnet Mask: the netmask of the router connected the IAD-16S;
 Default Gateway: the IP address of the router connected the IAD-16S;
PPPoE: PPPoE is an acronym for point-to-point protocol over Ethernet, which relies on two
widely accepted standards: PPP and Ethernet. PPPoE is a specification for connecting the
users on an Ethernet to the Internet through a common broadband medium, such as a
single DSL line, wireless device or cable modem. PPPOE IP address refers to IP address
assigned through the PPPoE mode.
If you choose PPPoE, you need to fill in to fill in the following information:
 Username: the account name of PPPoE
 Password: the password of PPPoE
 Server Name: the name of the server where PPPoE is placed
DNS Server
IP address of DNS server and default gateway information is displayed.
DBORegStatus
Whether the IAD-16S gateway is registered or not.
System Uptime
The running time of the IAD-16S since it is powered on.
Succeed: the IAD-16S gateway is sync to NTP server successfully;
NTP Status
Failed: the IAD-16S gateway fails to be sync to NTP server. Then you should check network
connection and the NTP server.
Network Traffic Statics
Total bytes of message received and sent by network port.
Usage of Flash
Detailed usage of Flash memory
Usage of RAM in Linux
Detailed RAM usage of Linux core
Usage of RAM in AOS
Detailed RAM usage of AOS
Current
The software version that runs on the gateway. Model name, version number and the
Version
Software
software development date are displayed.
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Backup
Software
Backup software is for the purpose of backup. When the current software fails, the backup
Version
software version will work.
U-boot Version
U-boot version
Kennel version
Linux Kennel version
FS Version
File system version
Hint Language
The current language of the IAD-16S gateway
3.5.2 Registration Information
Figure 3.5-2 Port and Port Group Registration Information
Primary/Secondary User status:
Registered: the port is registered to SIP server successfully;
Unregistered: the port fails to be registered to SIP server.
3.5.3 TCP/UDP Statistics
Figure 3.5-3 TCP/UDP Statistics Information
The above interface shows the statistical number of sending or receiving packets over TCP, and the
number of sending or receiving packets over UDP since the IAD-16S is booted up.
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3.5.4 RTP Session Statistics
Figure 3.5-4 RTP Session Statistics
The above interface shows real-time RTP session information, including: port, payload type, packet
period, local port, peer IP, peer port, sent packets, receive packets, lost packets, jitter and duration.
3.5.5 CDR Statistics
CDR (Call Detail Record): is a data record produced by a telephone exchange or a telecommunication
device, which contains the details of a telephone call that passes through the device.
On the Status & Statistic  CDR interface, details of all calls through the ports of the IAD-16S are
displayed. The CDR function can be enabled on this interface.
3.6 Quick Setup Wizard
Quick setup wizard guides user to configure the device step by step. User only needs to configure
network, SIP server and SIP port in the Quick Setup Wizard interface. Basically, after these three
steps, user is able to make voice call via the IAD-16S device.
3.7 Network Configuration
3.7.1 Local Network
The IAD-16S only works in the bridge network mode. It serves as a 16-port Ethernet switch. Under
this network mode, user only needs to configure the IP address of LAN port and DNS.
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Figure 3.7-1 Local Network
When “Obtain IP address automatically” is selected, the gateway will obtain IP address by DHCP.
When “Use the following IP address” is selected, user needs to configure a static IP address.
When “PPPoE” is selected, user needs to fill in the account and password offered by ISP.
【Notes】:
 If DHCP is selected to obtain IP address, please ensure DHCP server in the network works normally.
 After the configurations are finished, please restart the gateway for the configurations to take effect.
3.7.2 VLAN(Virtual Local Area Network)
In order to control the impacts brought by broadcast storms, user can divide VLANs into three groups,
namely VLAN1, VLAN2 and VLAN3. There are kinds of VLAN, including data VLAN, voice VLAN and
management VLAN. Different kind of VLAN has different messages.
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802.1Q
The IEEE 802.1Q standard defines the architecture for Virtual Bridged LANs, the services provided in
Virtual Bridged LANs and the protocols and algorithms involved in the provision of those services.
No Quality of Service mechanisms are defined in this standard, but an important requirement for
providing QoS is included in this standard, e.g. the ability to regenerate user priority of received frames
using priority information contained in the frame and the User Priority Regeneration Table for the
reception Port.
802.1P
IEEE 802.1P standard, describes important methods for providing QoS at MAC level. IEEE 802.1p is in fact
quite good. Lower priority level packets are not sent, if there are packets in queued in higher level queues.
IEEE 802.1p describes no admission control protocols. It would be possible to give Network Control
priority to all packets and the network would be easily congested.
Figure 3.7-3 VLAN parameter configuration
Explanations of the parameters in VLAN interface:
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The gateway supports three VLANs at most. Please enable VLAN according to actual
VLAN1/VLAN2/VLAN3
needs.
If the checkboxes on the right of data, voice and management of VLAN1 are selected, it
Data/Voice/Management, means data messages, voice messages and management messages are subject to the
network setting, 802.1Q VLAN1 ID and 802.1P Priority of VLAN1.
802.1Q VLAN ID(0-4095)
Set an ID to identify a VLAN based on 802.1Q protocol.
802.1p Priority(0-7)
Set the priority of a VLAN based on 802.1P protocol.
Set a DHCP IP address or static IP address for a VLAN, and set the IP address of the DNS
Network Setting
server used by the VLAN.
【Note】:User needs to restart the gateway for the configurations to take effect.
3.7.3 DHCP Option
3.7.4 Qos
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3.7.5 LAN Qos
3.7.6 ARP
ARP is address resolution protocol. ARP helps user get the MAC address of a device through its IP address.
Under TCP/IP network environment, each host is assigned with a 32-bit IP address, but MAC address
needs to be known for message transmission in the physical network. ARP is a tool that converts IP
address into MAC address.
Figure 3.7-9 ARP Parameters
3.8 SIP Server
Introduction of SIP Server:
1)SIP server is the main component of VoIP network and is responsible for establishing all the
SIP calls. SIP server is also called SIP proxy server or register server. Both IPPBX and softswitch
can act as the role of SIP server.
2)Usually, SIP server does not participate in media processing. Under SIP network, media
always use end-to-end negotiating. Simple SIP server is only responsible for the establishment,
maintenance and cleaning of sessions, while relatively-complex SIP server (SIP PBX) not only
provides basic calling and conversational support, but also offers rich services such as Presence,
Find-me and Music On Hold.
3)SIP server based on Linux platform, such as: OpenSER、sipXecx,VoS,Mera etc.
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4)SIP server based on windows platform, such as :mini SipServer、Brekeke,VoIPswitch etc.
5)Carrier-grade soft switch platform, such as Cisco, Huawei, ZTE etc.
Figure 3.8-1 Configuration Interface for SIP Server
Explanation for SIP parameters:
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Primary SIP Server Address
Primary SIP Server port
The IP address or domain name of the primary SIP server. They are
provided by VoIP service provider.
The Service port of the primary SIP server. It is 5060 by default.
It is used to avoid excessively frequent registrations.
Registration Expires
When the time that is set expires, terminals will send register request to
the primary SIP server. The time is 1800s by default.
Heartbeat is used to check the connection between terminal and SIP
Heartbeat
server.
Secondary SIP Server address
Secondary SIP Server port
The IP address or domain name of the backup SIP server. They are
provided by VoIP service provider.
Service port of the backup SIP server. It is 5060 by default.
It is used to avoid excessively frequent registrations.
Registration Expires
When the time that is set expires, terminals will send register request to
the backup SIP server. The time is 1800s by default.
Heartbeat is used to check the connection between terminal and SIP
Secondary SIP heartbeat
server.
Outbound proxy IP address or domain name provided by VoIP service
Outbound Proxy Address
provider.
Outbound Proxy Port
Retry
Interval
Default outbound proxy port is 5060.
when
Registration failed
Registration times per second
SIP Transport Type
The retry interval time after a registration fails. Default: 30s
The maximum number of registrations in a second. 0 means no limitation
for registrations.
The way of SIP-based transmission. It can be UDP, TCP and Auto. Default:
UDP.
Use Random Port
The SIP port for providing services for terminal is chosen by random.
SIP Local Port
Default SIP local service port is 5060.
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3.9 Port
Figure 3.9-1 Port Configuration Interface
Explanations for port parameters:
Port
Port number
Disable port
Whether to disable port temporally
Registration
Whether to enable registration for the port
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Primary /Secondary SIP
Display Name
Primary /Secondary SIP account description. It is used to identify the SIP account
Primary /Secondary SIP
User account information provided by VoIP service provider (ITSP). Usually in the
User ID
form of digit similar to phone number or actually a phone number.
Primary/Secondary SIP
SIP service subscriber’s authenticate ID used for authentication. It can be
Authenticate ID
identical to or different from SIP User ID.
Primary/Secondary
Authenticate password
Offhook Auto-dial
Auto-dial Delay Time
DND
Caller ID
Number for CFU
Number for CFB
Number for CFNRy
SIP password which registers to soft switch/SIP server
An extension or phone number is pre-assigned here so that the number is
automatically dialed as soon as user picks up the phone
How long the auto-dial number is prolonged. If it is set as 3s, the auto-dial
number is dialed after 3 seconds pass.
Do not disturb, the phone won’t receive any calls in case it enabled
Enable or disable caller ID for corresponding port. If it is disabled, the caller ID
for the calls through the port won’t be displayed.
Call forward unconditional. All incoming calls will be forwarded to pre-assigned
number automatically
Call forward on busy. If the line is busy, the call will be forwarded to pre-assigned
number automatically
Call forward no reply. If the call is not answered, the call will be forwarded to
pre-assigned number automatically
Call Waiting
If call waiting is enabled, a special tone is sent if another caller tries to reach you
Play Call Waiting Tone
If call waiting tone is enabled, caller will hear special tone.
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3.10 Advanced
3.10.1 FXS/FXO Parameters
FXS parameters include: timeout Call Progress Tone, Timeout for Dialing, Send Polarity Reversal etc.
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Figure 3.10-1 Configuration Interface for FXS Parameters
Explanation for FXS parameters:
With the help of dialing timeout, you can limit the time between two digits
while users are typing the digits of a number through an extension. If the
Timeout for dialing
timeout expires, the gateway will consider the dialing has finished and will try to
send message to SIP server. Default value is 4 seconds.
Timeout
for
This parameter determines how long the caller party will wait for answer when
answer(Outgoing call)
making outgoing calls through a phone.
Timeout
This parameter determines how long the phone rings when there are incoming
for
answer(Incoming call)
If this parameter is enabled, the situation will be detected when there is no RTP
No RTP Detected
Period
without
calls
packets received during the set time period.
RTP
Packet
Call Process Tone
Auto Gain Control
The time period when there is no RTP packets received.
The signal tone standard after a phone is picked up. Choose national standards
from the drop-down box. Default value is the United States.
Whether to enable automatic gain control
If polarity reversal is enabled, call tolls will be calculated based on the changes in
Send Polarity Reversal
voltage. If polarity reverse is disabled, you need to set the time for offhook
detection and call tolls will be calculated starting from the set time.
If ‘Detect Hook Flash’ is enabled, you need to set a minimum time and a
maximum time. If a phone’s hook flash is pressed for a time period greater than
Detect Hook flash
the set minimum time but less than the maximum time, the action is considered
as a ‘hook flash’ operation. If a phone’s hook flash is pressed for more the set
maximum time, the action is considered as ‘hang up the phone’.
CID Type
There are two CID types, namely DTMF and FSK.
Message Type
There are two call display types including SDMF and MDMF
Message Format
Send CID before Ringing
The call display format in analog phone. It can be “Display Name and CID”, “CID
only”, or “Display Name only”; default value is “Display Name and CID”
If this parameter is enabled, the gateway send Caller ID to phone before ringing,
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otherwise the caller ID will be displayed after ringing.
Delay of sending CID
The time how long the caller ID will be delayed when the caller ID is set to be
after Ringing
displayed after ringing. Default value is 500ms.
CFNRy Timeout
Timeout for ‘call forwarding on no answer’ service
SLIC Setting
Impedance matched with analog phone.
Long Line Support
Whether to enable ‘Long Analog Extension Line’.
3.10.2 Media Parameter
Media parameters mainly include: RTP start port, DTMF parameter, Preferred Vocoder, etc.
Figure 3.10-2 Configuration Interface for Media Parameters
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Explanation of media parameters:
Use Random Port
If this parameter is enabled, the gateway will choose a port by random as the
start port for RTP.
RTP Start Port
Default RTP start port is 8000
DTMF Method
Include SINGAL, INBAND and RFC2833
RFC2833 Payload Type
Payload value, default value is 101
DTMF Gain
Default value is 0 DB
DTMF Send Interval
The interval for sending DTMF signal. The default value is 200ms.
Send Flash Event
Coder Name
If this parameter is enabled, the gateway will send flash event to remote
terminal, and thus user does need to handle it locally
The gateway supports G729, G711U, G711A and G723. When outgoing calls are
made, G.729 will be used.
Payload Type
Each kind of coding has a unique load value, refer to RFC3551.
Packetization Time
The time for voice packaging
Rate
Voice data flow rate; It is defaulted by system.
Silence Suppression
Default value is ‘disabled’.
If this parameter is enabled, VoIP transmission
bandwidth can be saved, and meanwhile network congestion can be avoided.
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3.10.3 SIP Parameters
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Figure 3.10-3 SIP Parameter Configuration Interface
Explanation of SIP parameters:
SUBSCRIBE
for
MWI
(Message
Whether to enable ‘voicemail message waiting indicator’; it is
Waiting Indicator)
realized in the way of NOTIFY
MWI Subscription Expires
MWI subscription expiry time; Default value is 3600s.
Voicemail User ID
The user ID for access to voicemail box
RFC3407 Support
Whether to enable RFC3407 support.
IP-to-IP Call
If this parameter is enabled, user can dial IP address through a phone
to call destination gateway.
If this parameter is enabled, ‘user=phone’ will be contained in URI.
URI Includes user=phone
When calls are routed to PSTN network, the called number will be got
from user name. Default value is ‘not enable’.
INVITE
with”P-Preferred-Identity”
If this parameter is enabled, ‘P-Preferred-Identity’ Header will be
Header (RFC3325)
added in INVITE message for anonymous call (Support RFC3325).
Only Accept Call from ACL (SIP
If this parameter is enabled, the gateway only accepts incoming call
server or IP Trunk)
from SIP server only. Default value is ‘not enable’.
Anonymous Call
If this parameter is enabled,
‘anonymous’ will be included in SIP
message.
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If this parameter is enabled, all anonymous calls will be rejected.
Reject Anonymous Call
Default value is ‘not disable’.
# as ending Dial Key
‘# ‘ is used as the end mark for dialing.
If this parameter is enabled, ‘#’ is considered as a digit of the number
# Escape
Value
that is dialed.
of
“Refer
To“
refers
If this parameter is enabled, ‘contract header’ needs to be filled in in
to ”Contact”
the ‘refer to’ field of a SIP message.
Third Party Do Not Send 18x
If this parameter is enabled, the third party will not send 18x
Response
response during a attended transfer.
Send
BYE
when
Recv
REFER
Response (unattended)
Send New REGISTER when
If this parameter is enabled, the third party will send BYE to release
session after receiving REFER during a blind transfer.
Recv
423 Response
Implicit Subscribe
CSeq Start with 1
Forbid Invilad m=line in reINVITE
If this parameter is enabled, the value of ‘expires’ header will be
automatically updated and REGISTER will be re-sent after receiving of
423 response.
If this parameter is enabled, the gateway will accept implicit
subscription.
If this parameter is enabled, the value of CSeq starts with ‘1’.
If this parameter is enabled, the gateway will prevent
‘invilad
m=line’ from being carried in the SDP of re-INVITE.
RTP Mode in SDP when Call Holding
Use ‘sendonly ‘ or ‘inactive’ as RTP mode during call holding.
Support Call Waiting of Huawei
If this parameter is enabled, the gateway will support call waiting of
IPPBX
Huawei IPPBX.
Accept Orphan 200 OK
If this parameter is enabled, the gateway will support different ‘to-tag
200 OK’ in a INVITE session
Domain Query Type
There are two modes: A QUERY and SRV QUERY. Default is ‘A QUERY’.
Domain Re-resolution Interval
Default 0: forbidden
DNS cache
Early Media
If this parameter is enabled, the gateway will cache the DNS query
results.
Support the receiving of Early Media.
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PRACK(RFC3262)
Support reliable transmission of provisional response
PRACK Only for 18x with SDP
Send PRACK only when there’s SDP in 18x response
Early Answer
If this parameter is enabled, SDP will be contained in 18x
Session Timer (RFC4028)
Whether to enable ‘session timer’, default value is ‘ no’.
Session-Expires
Min-SE
The Session-Expires header field conveys the session interval for a SIP
session.
Min-SE header field indicates the minimum value for the session
interval.
T1
T1 timer of SIP protocol, default is 500ms
T2
T2 timer of SIP protocol, default is 400ms
T4
T4 timer of SIP protocol, default is 500ms
Max Timeout
The max timeout of sending or receiving, default is 32s
Heartbeat Interval
Default is 10s.
Heartbeat Timeout
Default to 16s
Username of OPTION(Heartbeat)
The user ID part of OPTION SIP message in the heartbeat request for
for “SIP Server”
SIP server
Username of OPTION(Heartbeat)
The user ID part of OPTION SIP message in the heartbeat request for
for “IP TRUNK”
IP trunk
Voicemail instructions:
Here takes the IAD-16S gateway together with Elastix as the example to introduce how voicemail
works in the gateway.
1)After the gateway registers to Elastix server, enable the voicemail function in Elastix for the
corresponding extension number and then set password. As below:
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Elastix Voicemail Configuration Interface
2)Check feature code in Elastix and change it if necessary. Its default feature code setting is as follows:
Elastix Voicemail Setting
On the Web interface of IAD-16S, click Advanced  SIP Parameter in the navigation tree and then
enter voicemail User ID.
VoiceMail Setting in SIP Parameter
3)Set ringing time in Elastix. Elastix will prompt user to leave a message after the corresponding
extension rings 15 seconds (by default). Then the Elastix sever will record the message. Related setting
is shown as follows:
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Voicemail Setting
4)Dial *200# on the extension which is connected to IAD-16S, then dial voicemail user ID and enter
password for authentication. After that user will hear voice message.
3.10.4 Fax Parameter
Figure 3.10-4 Configuration Interface for Fax Parameter
Explanation of fax parameters:
There are four fax modes: T.38, T.30(Pass-through),Modem and
Fax Mode
Adaptive.
Include “a=X-fax” Attribute
If this parameter is enabled, “a=X-fax” attribute will be carried in SDP
Include “a=fax” Attribute
If this parameter is enabled, “a=fax” attribute will be carried in SDP
Include
If this parameter is enabled, “a=X-modem” attribute will be carried in
“a=X-modem”
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Attribute
SDP
Include “a=modem” Attribute
If this parameter is enabled, “a=modem” attribute will be carried in
SDP
ECM
Whether to enable ‘Error Correction Mode’.
Rate
The rate of sending or receiving fax
Tone Detection by
Fax sound is detected by caller, callee or automatically
3.10.5 Digit Map
Figure 3.10-5 Digit Map
Digit Map Syntax
Supported
Digit
0-9
objects
T
Timer
DTMF
A digit, a timer, or one of the symbols of A, B, C, D, #, or *.
[]
One or more DTMF symbols enclosed in the [], but only one DTMF
Range
symbol can be selected.
Range
()
One or more expressions enclosed the
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(), but only one can be selected.
Separator
|
Separated expressions or DTMF symbols.
Subrange
-
Two digits separated by hyphen (-) which matches any digit between a
nd
including the two.
Wildcard
x
Matches any digit of 0 to 9
Modifiers
.
Matches 0 or more times of the preceding element
Modifiers
?
Matches 0 or 1 times of the preceding element
Examples:
(13 | 15 | 18)xxxxxxxxx
Matches the phone numbers with stating digits as 13, 15 or 18 and the
left nine digits as any of 0 to 9.
3.10.6 Feature Codes
Please make reference to 2.7 Description of Feature Codes and the following table.
Inquiry LAN port IP address
Dial*158# to obtain device’s LAN port IP address
Inquiry Phone Number
Dial*114# to obtain port account
Inquiry PortGroup Number
Dial *115# to obtain port group number
Setting IP Mode
Network Work Mode
*150*0#, means pppmodem, *150*1#, means static IP, *150*2#,
means obtain IP address by DHCP, *150*3#, means pppoe.
*157*0#, set network work mode to routing mode; *157*1#, set
network work mode to bridge mode
Configure IP Address
*152*+IP, set gateway IP address
Network subnet mask configure
*153*+subnet mask, set gateway subnet mask
Network Gateway Configure
*156*+gateway IP, set gateway
Renew DHCP
*193#, set dynamic IP again
Reset Basic Configuration
Dial *165*000000# to restore default username/password and network
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configuration
Reset Factory Configuration
*166*000000#, reset factory
Restart Device
*111#, restart device
During a call, dial*# into call hold.(Recovery the call through hook flash
Call holding
or *#)
Call by IP
Directly dial the end user IP to call
Call Waiting Activate
*51#, enable call waiting function
Call Waiting Deactivate
*50#, forbid call waiting function
Blind Transfer
If the call transfer to 801, first hook flash and then dial the * 87 * 801#
Call
Forward
Unconditional
Activate
Call
Forward
Unconditional
Deactivate
*72*+ phone number#, transfer the call from the phone number
*73#, forbid call forward unconditional
Call Forward Busy Activate
*90*+ forward busy number#
Call Forward Busy Deactivate
*91#, forbid call forward busy
Call Forward No Reply Activate
*92*+ forward no reply number#
Call
Forward
No
Reply
Deactivate
*93#, close this function
Do Not Disturb Activate
*78#, enable DND function
Do Not Disturb Deactivate
*79#, close DND function
Dial Voicemail
*200#, visit voice mail box
3.10.7 System Parameter
System parameters include: STUN, NTP, Provision, EB parameter and Telnet.
1)STUN:STUN(Simple Traversal of UDP over NATs)is a lightweight protocol that allows applications to discover
the presence and types of NATs and firewalls between them and the public Internet. It also provides the ability
for applications to determine the IP addresses allocated to them by the NAT. STUN works with many existing
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NATs, and does not require any special behavior from them. STUN doesn’t support TCP connection and
H.323.
2)NTP:Network Time Protocol(NTP)is a computer time synchronization protocol.
3)Provision:provision is used to make the gateway automatically upgrade with the latest firmware stored on an
http server an ftp server or a tftp server.
Figure 3.10-7 Configuration Interface for System Parameters
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Explanation for related parameters:
Hint Language
IVR language of the gateway
NAT Traversal
User can choose ‘Disable’, ‘ STUN’, ‘static NAT’ and ‘dynamic NAT’.
NTP
To Enable or disable NTP
Primary NTP server address
The IP address of primary NTP server; default IP address is us.pool.ntp.org.
Primary NTP server port
The service port of primary NTP server; Default port is 123.
Secondary NTP server address
The IP address of secondary NTP server ; Default IP address is 18.145.0.30
Secondary NTP server port
The service port of secondary NTP server; Default port is 123
SYN Interval
The interval to synchronize the time of the IAD-16S. Default value is 3600s.
Time Zone
The time zone of the gateway;
Default configuration is United States
central time, Chicago.
Daylight Saving Time
Enable or disable daylight saving time
Daily Reboot
Whether to enable daily reboot
Reboot time
The time to reboot the gateway daily
WEB Port
The web port of the gateway;
Telnet port
Listening port of telnet service; Default port is 23
Default port is 80
3.10.8 Action URL
Action URL can be used as a means to allow the VoIP platform to learn about the IAD’s status. It
transmits data via GET request over the HTTP protocol. The IAD is an HTTP client. At HTTP server side,
GET request must be processed by the VoIP platform. Thus, the purpose is achieved.
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3.11 Call & Routing
3.11.1 Wildcard Group
Figure 3.11-1 Wildcard Group
3.11.2 Port Group
On the Port Group interface, user can group several ports together and then set a strategy for port
selection of the group. Parameters of port group include registration, primary display name, primary
SIP user id, primary authentication ID and password, secondary display name, secondary SIP user id,
secondary authentication ID and password, off-hook auto dial, auto dial delay time, port select and so
on.
Figure 3.11-2 Configuration Interface for Port group
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Explanation of related parameters
Index
The NO. of the port group ; It uniquely identifies a route, range from 0-7
Description
The description of the port group; it is used to identify the port group
Port group display, which will be used in SIP message, for example:
INVITE sip:bob@biloxi.com SIP/2.0
Via:SIP/2.0/UDPpc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70
Primary/Secondary Display Name
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Here Bob and Alice is the display
User account information, provided by VoIP service provider (ITSP). Usually in the
Primary/Secondary SIP User ID
form of digit similar to phone number or actually a phone number.
Primary/Secondary
Authenticate
ID
Primary/Secondary
SIP service subscriber’s authentication ID, it can be identical to or different from
SIP User ID.
Authenticate
Password of SIP user ID
Password
Offhook Auto-Dial
To enter offhook auto-dial number
Auto-dial Delay time
How long auto-dialing will be delayed
It specifies the policy for selecting a port for ringing in the port group
 Ascending: the gateway always selects a port from the minimum number.
 Cyclic ascending: the gateway always selects a port from a number next to the
number selected last time. If the maximum number was selected last time, the
next selected number is the minimum number. The sequence moves in cycles like
Port Select
this.
 Descending: the gateway always selects a port from the maximum number.
 Cyclic descending: the gateway always selects a port from a number next to
the number selected last time. If the minimum number was selected last time,
the next selected number is the maximum number. The sequence moves in cycles
like this.
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 Group ring: all ports ring at the same time
When one port rings, user can dial ‘*#’ to pick up the call from other ports under
Pickup UP on group
the same port group.
Port
Select ports for this port group
3.11.3 IP Trunk
A peer-to-peer VoIP call occurs when two VoIP phones communicate directly over IP network without
IP PBXs between them. IP trunk helps establish peer-to-peer call between gateway and VoIP phones.
IP trunk will be used in routing configuration.
Figure 3.11-3 IP Trunk Configuration Interface
Explanation of related parameters:
Index
The No. of the IP trunk; from 0 to 127
Description
The description of the IP trunk; It is used to n identify the IP trunk
Remote Address
IP address or domain name of peer device
Remote Port
SIP port of peer device
Heartbeat
Whether to enable the ‘Heartbeat’ function for the IP trunk. Default value is ‘ not
enable’.
If heartbeat is enabled, the gateway will send “OPTION” to peer device.
3.11.4 Routing Parameter
This parameter determines a call is routed before or after manipulation.
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Figure 3.11-4 Configuration Interface for Routing Parameter
3.11.5 IP -> Tel Routing
Figure 3.11-5 Configuration Interface for IP-Tel Routing
Explanation of related parameters:
Index
IP Routing priority: from 0 to127; 0 is the highest priority.
Description
It is used to identify the IP  routing
Calls from
IP Trunk or SIP Server; ‘any’ means any IP addresses
The prefix of the caller number, which helps match routing exactly. its length is less
Caller Prefix
than or equal to the caller number. For example, if caller number is 2001, the caller
prefix can be 200 or 2. ‘any’ means the prefix matches any caller number.
Callee Prefix
The prefix of the called number, which helps match routing exactly. Its length is less
than or equal to the called number. If the called number is 008675526456659, the
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called prefix can be 0086755 or 00.,“any” means the prefix matches any called
number
Calls to
Which port or port group to which calls are routed
3.11.6 Tel-IP/Tel Routing
Figure 3.11-6 Configuration Interface for Tel-IP/Tel Routing
Explanation of related parameters:
The index of this Tel IP/Tel routing, from 0 to 127. Each index cannot be used repeatedly.
Index
Routing priority: 0 is the highest priority.
Description
It is used to identify the routing
Calls From
Tel IP calls are from a port or a port group
The prefix of the caller number, which helps match routing exactly. its length is less than or
Caller Prefix
equal to the caller number. For example, if caller number is 2001, the caller prefix can be 200
or 2. ‘any’ means the prefix matches any caller number.
Callee Prefix
The prefix of the called number, which helps match routing exactly. Its length is less than or
equal to the called number. If the called number is 008675526456659, the called prefix can
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be 0086755 or 00.,“any” means the prefix matches any called number.
Calls to
Calls are routed to a port, port group, IP trunk or SIP server
3.11.7 IP – IP Routing
Figure 3.11-7 Configuration Interface for IP->IP Routing
Explanation of related parameters:
Index
The index of this IP IP routing, from 0 to 127. Each index cannot be used repeatedly.
Routing priority: 0 is the highest priority.
Description
It is used to identify the routing
Calls From
Calls are from IP trunk.
The prefix of the caller number, which helps match routing exactly. its length is less than or
Caller Prefix
equal to the caller number. For example, if caller number is 2001, the caller prefix can be 200
or 2. ‘any’ means the prefix matches any caller number.
The prefix of the called number, which helps match routing exactly. Its length is less than or
Callee Prefix
equal to the called number. If the called number is 008675526456659, the called prefix can
be 0086755 or 00.,“any” means the prefix matches any called number.
Calls to
Calls are routed to IP trunk
3.12 Manipulation Configuration
Number manipulation refers to the change of a called number or a caller number during calling
process when the called number or the caller number matches the preset rules.
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3.12.1 IP -> Tel Callee
Figure 3.12-1 Add IP -> IP Callee
Index
The index of this manipulation, from 0 to 127. Each index cannot be used
repeatedly. 0 is the highest priority
Description
Name of this IP ->Tel manipulation name
Calls From
Determine the calls come from IP trunk or SIP server
Set a prefix for caller number. The prefix’s length is less than or equal to
Caller Prefix
that of the caller number, which helps to match routing. If caller number
is 2001, the caller prefix can be 200 or 2. “any” means match any caller
number.
Set a prefix for called number. The prefix’s length is less than or equal to
Callee Prefix
called number, which helps to match routing. If called number is
008675526456659, the called prefix can be 0086755 or 00., “any” means
match any called number
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Calls to
Determine the port or port group to which the call is routed.
Stripped Digits from Left
The number of digits which are lessened from the left of the callee
number
Stripped Digits from Right
The number of digits which are lessened from the right of the callee
number
Prefix to Add
The prefix added to the callee number after its digits are lessened.
Suffix to Add
The suffix added to the callee number after its digits are lessened.
Number of Digits to Leave from
The number of the retained digits which. are counted from the right of
Right
the callee number
3.12.2 Tel -> IP/Tel Caller
Figure 3.12-2 Add Tel -> IP Caller
Configuration parameters are the same with those of ‘IP->Tel Callee’.
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3.12.3 Tel-IP/Tel Callee
Figure 3.12-3 Add Tel-IP Callee
Configuration parameters are the same with those of ‘Tel->IP Caller’.
3.13 Routing rule examples
3.13.1 Route any calls from any IP to specific port
After enter the Web interface, click Call & Routing  IP-Tel Routing in the navigation tree on the left,
and then click Add to create a new routing rule.
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In the example above, all calls will be routed to port 0 when the routing rule is matched.
3.13.2 Route any calls from any IP to specified port group
Create port group
Before we can route calls to a port group, create the port group first as below. On the Call & Routing
 Port Group, click Add to create a new port group.
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Port 12, port 18, port 27 and port 631 are assigned to port group 7.
Route any calls to the port group
On the Call & Routing  IP-Tel Routing interface, click Add to create a new routing rule.
As shown above, if the routing rule is matched, calls will be routed to port group 7.
3.13.3 Route any calls from any port to specific SIP IP trunk
Create IP Trunk on the Call & Routing  IP Trunk interface:
After IP Trunk is created, check the following configuration:
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As shown above, the IP trunk is created, and the remote end IP address is 172.16.125.125, the SIP port
is 5060.
Create Tel -> IP routing rule
On the Call & Routing Tel-IP Routing interface, click “Add” to create a new Tel  IP routing rule.
All Tel calls from any caller number to any called number will be routed to IP trunk 127.
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3.14 Maintenance
3.14.1 TR069
ACS URL (auto-configuration server URL address) is provided by service provider. The ACS URL
generally starts with http:// or https://
Username and password are used for ACS authentication.
Figure 3.14-1 TR069 Parameters
3.14.2 SNMP (Simple Network Management Protocol)
SNMP Parameters:
• SNMP enable: to disable or enable the SNMP feature
• SNMP version: the IAD-16S gateway supports SNMP v1 and v2
• Community: the community name used to read through SNMP protocol
• Source: the IP address of SNMP server
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Figure 3.14-2 SNMP Parameters
User configuration is only available on SNMP v3.
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Group configuration
Group: community group name which consist of character string.
Community: let community join the community group which configured above
Trap configuration
Trap configuration enable to configure Trap server IP and port. This setting available for SNMP v2c and
v1.
3.14.3 Syslog
Syslog is a standard for network device data logging. It allows separation of the software that generates
messages from the system that stores them and the software that reports and analyzes them. It also provides
devices which would otherwise be unable to communicate a means to notify administrators of problems or
performance. There are 5 levels of syslog, Including NONE, DEBUG, NOTICE, WARNING and ERROR.
The Signal Log is include following traces which defined in system by default
- SD, hardware debug
- SIP, SIP signaling trace
- STUN, STUN logs
- ECC, detail information of call control module
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- RE, the common communication module for SCP and SIM
- SCP, the communication protocol between gateway and cloud server
The media log is include following traces which defined in system by default
- RTP, RTP stream info collection
- SIM, to output traces between gateway and remote SIM cards
The System Log is include following traces which mainly used by developer
- SYS, system log
- TIMER, system process
- TASK, system task process
- CFM, system process
- NTP
The Management Log is include following traces which defined in system by default
- CLI, command line
- TEL,
- LOAD, firmware upload
- SNMP
- WEBS, embedded web server
- PROV, provisioning
Server Syslog:
When the gateway register to SIM Cloud server, the option will be changed to un-configurable and all
logs to be storage on server.
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Figure 3.14-3 Syslog Parameter
Enable send CDR, and then send communication information to syslog server.
3.14.4 Provision
Provision is used to make the IAD-16S automatically upgrade with the latest firmware stored on an
http server an ftp server or a tftp server.
Figure 3.14-4 Provision
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URL
Provisioning server URL, support HTTP, TFTP, FTP
Check Interval
The interval to check the changes on the provisioning server
Account
Account for login provisioning server
Password
Account for login provisioning server
3.14.5 Cloud server
User can register the gateway to cloud server, and then the gateway will be managed by cloud server.
Figure 3.14-5 Cloud Server
Explanation of related parameters
Server Address
The IP address or domain of the cloud server
port
The listening port of the cloud server
Password
Password for register with cloud server
3.15 User Manage
On the following interface, user can choose whether to enable the ‘User Manage’ function.
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3.16 Remote Server
On the following interface, user can choose whether to enable remote server.
3.17 Record Parameter
On the following interface, user can choose whether to enable the record function.
3.18 Security
3.18.1 WEB ACL
ACL (Access Control List) for WEB is used to configure IP addresses (users) that are allowed to access
the WEB page of the gateway. The IP address list can’t be null once ACL is enabled.
Figure 3.15-1 ACL for WEB
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3.18.2 Telnet ACL
ACL (Access Control List) for WEB is used to configure IP addresses (users) that are allowed to access
the Telnet page of the gateway. The IP address list can’t be null once ACL is enabled.
Figure 3.15-2 ACL for Telnet
3.18.3 Passwords
On the following interface user can configure or modify the username and password for access the
WEB interface and the Telnet interface.
Note:Both the username and password of Web and Telnet are ‘admin’ and ‘admin’.
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Figure 3.15-3 Password Modification
3.18.4 Encrypt
3.19 Tools
3.19.1 Firmware upload
Firmware upload steps:
Step 1.
Check the current firmware version on the System Information page
Figure 3.16-1 Firmware Version
Step 2.
Prepare firmware package. The most important is that the package must match with the existing version.
Package version consists of the following parts:
1.18.xx.xx
01/02 is vendor name
18 is hardware version, xx.xx is version number
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Step 3.
Upload firmware, select the package from specific folder on the computer and click Upload button.
Figure 3.16-2 Firmware Upload
Step 4.
Keep waiting until it prompts ‘Software loaded successfully!’
Figure 3.16-3 Successful Firmware Upload
Step 5.
Reboot gateway. Refer to web page Maintenance-> Device Restart
Figure 3.16-4 Restart Gateway
3.19.2 Data Backup
The process data backup:
1)
Click “Data Backup”
2)
Click “Backup” to backup data to PC.
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Figure 3.16-5 Data Backup
3.19.3 Data Restore
The processes of data restore:
Click ‘Data Restore’;
Browse file, select data file.
Click ‘Restore”’and then import successfully, the device will restart automatically.
Figure 3.16-6 Data Restore
3.19.4 Ping Test
On the Tools  Ping Test interface, user can use Ping to check whether the network is working or not.
Ping instructions:
1) Click ‘Tools  Ping Test’ on the navigation tree on the left;
2) Fill in IP address or domain whose connection needs to be checked, click start.
If a message is received, it indicates that network connection is normal. Otherwise the network
connection is faulty.
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Figure 3.16-7 Ping Test
3.19.5 Tracert Test
Tracert is a trace router used to track routing.
Tracert sends a sequence of Internet Control Message Protocol (ICMP) echo request packets
addressed to a destination host. Determining the intermediate routers traversed involves adjusting the
time-to-live (TTL), aka hop limit, Internet Protocol parameter. Frequently starting with a value like 128
(Windows) or 64 (Linux), routers decrement this and discard a packet when the TTL value has reached
zero, returning the ICMP error message ICMP Time Exceeded.
Tracert works by increasing the TTL value of each successive set of packets sent. The first set of
packets sent have a hop limit value of 1, expecting that they are not forwarded by the first router. The
next set have a hop limit value of 2, so that the second router will send the error reply. This continues
until the destination host receives the packets and returns an ICMP Echo Reply message.
Trace route uses the returned ICMP messages to produce a list of hops (which usually consists of
routers and layer 3 switches) that the packets have traversed. The timestamp values returned for each
router along the path are the delay (aka latency) values, typically measured in milliseconds for each
packet.
Tracert introduce:
Click ‘ Tracert Test’ in the navigation tree;
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Fill in IP address or domain whose route needs to be tracked, and then click start.
Figure 3.16-8 Tracert Test
3.19.6 Outward Test
Outward test enable user to diagnose the physical phone lines which follow GR909 standards. To start
outward test, select the ports to be tested and click ‘start’. Testing costs a few minutes.
Figure 3.16-9 Outward Test
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Test results
OK: the analog phone set and phone line are working well
FAIL: analog phone doesn't connect to FXS port or there’s something wrong in phone set
3.19.7 Network Capture
Network capture is a very important diagnostic tool for maintenance. It can be used to capture data
packages of the available network ports.
Default Setting is PCM capture
PCM capture helps to analysis voice stream between analog phone and DSP chipset.
To enable PCM capture
 Select ‘PCM’ on Network Capture page
 Click “Start’ to enable PCM capture
 Dialing out through gateway, start talking a short while then hangup the call.
 Click ‘Stop’ to disable network capture
 Save the capture file to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The
sample of PCM capture as below:
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Getting start to Syslog capture
Syslog capture is another way to obtain syslog which the same as remote syslog server and filelog. The capture
file is save as pcap format so that it can be opened in some of capture software like Wireshark, Ethereal software
etc.
To enable syslog capture
 Select Syslog special only on Network Capture page
 Click “Start’ to enable syslog capture
 Dialing out through gateway, start talking a short while then hangup the call.
 Click ‘Stop’ to disable syslog capture
 Save the capture to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The
sample of syslog capture as below:
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Getting start to RTP capture
PCM capture is help to analysis voice stream between gateway and remote IPPBX/SIP Server.
To enable RTP capture:
 Select RTP special on Network Capture page
 Click Start to enable RTP capture
 Dialing out through gateway, start talking a short while then hangup the call.
 Click Stop to disable RTP capture
 Save the capture to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The
sample of RTP capture as below:
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Getting start to DSP capture
DSP capture is help to analysis voice stream inside DSP chipset. The DSP chipset will handle RTP from IP network
as well as voice stream from analog phone.
To enable DSP capture:
 Select DSP only on Network Capture page
 Click Start to enable DSP capture
 Dialing out through gateway, start talking a short while then hangup the call.
 Click Stop to disable DSP capture
 Save the capture to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The
sample of RTP capture as below:
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Configurable capture options
Getting start to custom capture
This menu provides more options to capture specific packets according to actually needs.
3.19.8 Factory Reset
Click ‘Apply’ to restore the factory settings.
Factory Reset
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3.19.9 Device Restart
After saving all the configurations or changes to the equipment, user can restart the IAD-16S gateway
for the changes to take effect.
Restart Gateway
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4 Glossary
•
DNS:Domain Name System
•
SIP:Session Initiation Protocol
•
TCP:Transmission Control Protocol
•
UDP:User Datagram Protocol
•
RTP:Real Time Protocol
•
PPPOE:point-to-point protocol over Ethernet
•
VLAN:Virtual Local Area Network
•
ARP:Address Resolution Protocol
•
CID:Caller Identity
•
DND:Do NOT Disturb
•
DTMF:Dual Tone Multi Frequency
•
NTP:Network Time Protocol
•
DMZ:Demilitarized Zone
•
STUN:Simple Traversal of UDP over NAT
•
PSTN:Public Switched Telephone Network
•
IMS: IP Multimedia Subsystem
•
ACL: access rule list
•
SNMP: Simple Network Management Protocol
•
FXS: Foreign Exchange Station
•
FXO: Foreign Exchange Office
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