Hybertone GoIP User manual

GoIP User Manual
http://www.hybertone.com Sales: [e-mail] Support：[email protected]
GoIP User Manual
VoIP GSM Gateways
Models:
GoIP
GoIP-4
GoIP-8
GoIP-16
HYBERTONE
Revision: 1.2
2012/6/22

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Content
1General .......................................................................................................................................................2
1.1 Introduction.........................................................................................................................................2
1.2 Protocols .............................................................................................................................................3
1.3 Hardware Features ............................................................................................................................3
1.4 Software Features ..............................................................................................................................3
1.5 Package Content ................................................................................................................................4
1.6 LED Indicators ...................................................................................................................................5
2Installation..................................................................................................................................................6
3Configuration.............................................................................................................................................8
3.1 HTTP WEB Server Login...................................................................................................................8
3.2 Status...................................................................................................................................................9
3.3 Configuration....................................................................................................................................12
3.3.1 Preference .........................................................................................................................................13
3.3.2 Network ............................................................................................................................................15
3.3.3 VoIP...................................................................................................................................................18
3.3.3.1 SIP.....................................................................................................................................................18
3.3.4 Call Management..............................................................................................................................27
3.3.5 GSM Service Provider......................................................................................................................35
3.3.6 GSM Base Station Settings...............................................................................................................36
3.4 Tools..................................................................................................................................................36
AppendixA. Special SMS Commands...................................................................................................42
Appendix B. SMS Modes.......................................................................................................................43
Appendix C. Custom Network Tones.....................................................................................................47
Appendix D. GSM Group Mode ............................................................................................................48
Appendix E. CID Call Forward..............................................................................................................49
Appendix F. Volume Adjustment...........................................................................................................50
Appendix G. Device Characteristics.......................................................................................................51

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1 General
1.1 Introduction
GoIP is the abbreviated from GSM over IP. It is a new type of VoIP gateway that allows call terminations
from a VoIP network to a GSM network and vice versa. Call connections between IP networks and GSM
networks are now bridged seamlessly to extend the voice communication coverage significantly. As the
traditional PSTN lines are starting to disappear in developed countries and are not going to be built
extensively in under-developed countries, GSM phones are getting more and more popular all over the world
with lower and lower service charges, the emergence of GoIP bridges the gap between the traditional
telephone networks and VoIP networks as shown in the diagram below. As a result, local and worldwide
voice communications are more convenience, lower cost, and broader coverage.
Internet /
Intranet
Local
GSM Network
Local
Telephone
Network
GoIP
World
Telephone
Network
VoIP
Service
Provider
Cellphone Telephone
Telephone
VoIP
Phone
VoIP
ATA
GoIP4 GoIP8
You can now make a call from anywhere in the world via a VoIP network and then terminate the call via a
GoIP to the local telephone network (PSTN). On the other hand, you can also make a call from the local
telephone network to a GoIP (the GSM phone number) and then dial another number via a VoIP network to
anywhere in the world. In these two cases, a VoIP Service provider is required for one side of the call
termination. For two fixed locations, it is possible to setup GoIPs at both ends for call terminations without
subscribing to a VoIP Service provider.
GoIP can also be used to achieve GSM roaming via VoIP. The idea is to route all your incoming GSM calls to
a GoIP via call forward or simply insert your SIM card to a GoIP. You can then setup the GoIP to forward all
incoming calls to another GSM number in the world via a VoIP service provider. The charge per call from a
VoIP service provider is significantly lower than the roaming charge.
For office environment, GoIP offers a quick way to replace the traditional PSTN lines or T1/E1 lines to your IP
PBX. There is no initial installation/reallocation charge and no need to wait for installation. Depending on
our usage, you can add or remove lines as per your requirement. You can even configure the system so that
everybody calls the same number regardless the number of lines available.

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1.2 Protocols
TCP/IP V4 (IP V6 automatic adaptive)
Dual VoIP protocols: ITU-T H.323 V4, IETF SIP V2.0
Multiple Codecs: ITU-T G.711 Alaw/ULaw, G.729A, G.729AB, G.723.1 and GSM
H.2250 V4
H.245 V7
H.235 (MD5, HMAC-SHA1)
RFC1889 real-time digital transmission protocol
NAT
STUN
Network Management Protocol (NMP)
PPPoE Dial Up
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP
Hypertext Transfer Protocol (HTTP)
Dynamic Host Configuration Protocol (DHCP)
Domain Name System (DNS)
User Account Authentication (via MD5)
Proprietary Relay Protocol (Avoiding VoIP Blockings)
1.3 Hardware Features
ARM9E processor
DSP for voice signal processing
Two 10/100MB Ethernet ports (IEEE 802.3 standard) with status LEDs
Quadband GSM module (850M 900M, 1800M and 1900M)
Internal Antenna (optional)
1.4 Software Features
LINUX OS
Built-in Web Server for device configuration
Built-in SIP Proxy (Simplified)
PPPoE Dial Up
Router function
DHCP client & Server
QoS (VLAN)
VPN (PPTP)
Online firmware upgrade
Remote Control Mechanism for remote technical support
Proprietary Auto Provisioning Mechanism
Remote SIM function
Short Messages (SMS) support (standalone and server based)
Call Management and Routing

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1.5 Package Content
Use care when unpacking the device package in order to avoid damage to the main unit and the packing
materials. Retain the packing materials in case the unit is to be transported in the future.
Please inspect the shipping container and the contents for any damages. If visible damages are present,
please contact your vendor. Keep the shipping materials for the carrier inspection.
The package should contain the items listed in the table below
Item
Appearance
Description
1.
GoIP (1-Channel)
GoIP-4 (4-Channel)
GoIP-8 (8-Channel)
GoIP-16 (16-Channel)
1 x Main Unit
2.
AC/DC Power Adapter:
GoIP1: 12V/500mA
GoIP4: 12V/2A
GoIP8: 12V/3A (Adapter + external
power cord)

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3.
1 x Ethernet CAT5 Cable (2M)
1.6 LED Indicators
LED indicators (shown above for GoIP-8) are used to show the current status of the device. They are often used
to determine if the GoIP is working normally or not.
LED Label
Description
Power
This LED is red and illuminates when power is connected.
LAN
This LED is red and illuminates when the LAN port is connected and blinks when data
transmission occurs.
PC
This LED is red and illuminates when the PC port is connected and blinks when data
transmission occurs.
RUN
This LED is green and blinks at a rate of every 100ms when VoIP is not ready for
making calls. (Fast Blink)
It blinks at a rate of every second when VoIP is ready for making calls (Slow Blink).
Channel “x”
Each GSM channel has its own status LED and its color is green.
1. It blinks at a rate of every 100ms (Fast blink) when the corresponding GSM
channel is not yet registered to a GSM network.
2. It blinks at a rate of every second (Slow blink) when the corresponding GSM
channel is ready for making or receiving calls (registered to a GSM network).
3. It illuminates when GSM call activities occurs (in use, ringing).

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2 Installation
The installation procedures for GoIP, GoIP-4 and GoIP-8 are the same. The only different is on the number
SIM cards to be installed.
1. SIM card slots are located either at the bottom (for old hardware) or at the back (for new hardware) of
the main unit.
For the models with the SIM card slots located at the bottom, you need to open the bottom SIM cover in
order to install SIM cards. First slide the metal clip to the direction as indicated on the top of the clip.
Insert a SIM card to each slot carefully and then place the metal clip back in place.
For the models with the SIM card slots located at the back, just insert a SIM card to each slot as shown in
the drawing on the right. Please make sure that the orientation of the SIM Card is correct before
inserting the card.
For GoIP (1-channel), the SIM card insertion orientation is shown in
the figure on the right. The metal contacts must face down and
the cut corner is inserted first.
For GoIP-4 and GoIP-8, the SIM card insertion orientation is
shown in the figure on the right. The metal contacts must
face up and the cut corner is inserted first.
2. The LAN port is intended for intranet or internet connection. Depending
on your network environment, it can be connected various type of network
equipment, such as network router, network switch / Hub, xDSL/Cable
modem, etc.
3. The PC port is intended for network sharing and it supports both bridge and
router modes. In Bridge mode, the PC port is connected to the same
network segment as the LAN port. In Router mode, the PC port is set to a
different network segment. In this case, please make sure that the PC
network segment IP (192.168.x.) is different from the one in the LAN port
network.
4. The DC port is for power connection. Please
only use the AC/DC adapter provided. Adapter
with different rating or vendor may damage the
device or affect its performance.

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5. The Reset button is recessed inside the GoIP cabinet. You need to use a sharp pointer to access the
reset button. Press it momentarily to reboot the device. Press it for 15 seconds or more to reset the
device settings including login password to its factory defaults.

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3 Configuration
The device can be configured via its built-in http web server or via an Auto Provision Server. Auto Provision
Server is a free utility supporting both Window and Linux OS. This utility is developed by HYBERTONE
Technology for the sole purpose of automating the configuration of our products. It is available in our
website for free download. This user manual only focuses on the device configuration via its built-in http
web server.
Please note that only window based Web browsers, such as IE, Chrome, Firefox, Mozilla, are supported. Do
not use Linux based Web browsers. If you are having problems in configuring your device with your existing
Web browser, please try one with lower version or a different Web browser and report the problem to us.
3.1 HTTP WEB Server Login
There are two methods to access the built-in web server.
1. Method 1 is to access the built-in web server via the LAN port. The LAN port is set to DHCP mode as a
factory default. When you connect it to a network with a DHCP host, it will obtain an IP address from
the DHCP host automatically. Via the GoIP’s GSM channel(s), there are two ways to find out the IP
address that is assigned to this port.
i. Dial the SIM number of anyone of the GSM channels available. Once the call is answered, dial
“*01”to hear a voice prompt reporting the LAN port IP address.
ii. Send the “###INFO###”SMS command to one of the GSM channels available. The GoIP will then
return back the LAN port IP address. Please refer to Appendix A Special SMS Commands for more
information.
Once the LAN IP address is known, you are now ready to access its built-in http web server by typing its
IP address in the address field of a web browser.
2. Method 2 is to access the built-in we server via the PC port. As a factory default, the PC port IP is
preset to 192.168.8.1. Connect a computer to the LAN port of the device and configure its IP to
192.168.8.x (x = 2 to 254). Type the IP address 192.168.8.1 in the address field of a web browser.
Once the IP address is entered, the login window
shown on the right pops up. Enter the user name
and password. There are three level of access via
three different user names.
1. Administrative Level -This offers a full access
right to all parameters available in the built-in
webpage. The user name and password for
the administrative level are "admin" and
"admin" respectively.
2. User Level - This level restricts user from
accessing the Call Setting page. User will not
be able to change any VoIP related settings.
The user name and password for the user level
are "user" and "1234" respectively.
3. SMS Level - This level only allows user to access
the Send SMS and SMS Box functions under the
Tool menu. The user name and password for

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the SMS level is "sms" and "1234".
3.2 Status
This is the default page when you first login to the built-in HTTP Web server. It consists of three columns as
shown in the diagram below. All three models use the same status page design. It is important to
understand the information shown in this page in order to debug or report problems encountered.

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1. The left hand column shows the VoIP registration and call status by line.
a) SIP Number –This shows SIP number of the Line. For Single Server mode, all lines use the same SIP
number. For Config. By Line mode, each line should have its own SIP number. For Config. By Group
mode, lines belongs to the same group use the same SIP number. For SIP Trunk Gateway mode, this
field is optional.
b) Login –This parameter shows the VoIP registration status. The status "OK" means that the SIP
registration for the line is successful. "FAIL" means that the SIP registration fails and no calls can be
routed to and from the GSM channels. For SIP Trunk Gateway mode, the Login status is always "OK"
c) Status –This shows the operating status of the SIP line. It is important to understand how call
routing works between SIP and GSM in order to understand the meaning of the status. In general,
the line status can be classified as follows:
Calls from SIP to GSM or Calls from GSM to SIP
i. IDLE - The SIP line is not engaged in any call activities.
ii. IN USE - This status is active when a call connection between SIP and GSM is established.
Calls from GSM to SIP
iii. READY - This status only applies to 2nd dial operation. It is active when a SIP call is answered
and a second dial tone is generated to prompt the SIP caller to dial a number out.
iv. DIALING - This status is active when the GSM channel is dialing a number out,增加了 NUMBER
如dialing 10086
v. RINGING - This status is active when ringback tone occurs and the call is not yet answered.
vi. CONNECTED—介绍
2. The middle column shows information on the hardware and network .
Hardware
a) S/N –This field shows the serial number of the device.
b) Firmware –This field shows the current firmware version.
c) Model –This field shows the model number of the device
Network
LOCAL TIME

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a) LAN Port –This field shows the IP address of the LAN Port.
b) LAN MAC - This field shows the physical hardware address (MAC) of the LAN port.
c) PC Port –This field shows the IP address of the PC Port.
d) PPPoE –This field shows the PPPoE dial up status. It is only meaningful when PPPoE is enabled.
e) Gateway –This field shows the default gateway IP for data traffics routing.
f) DNS Server –This field shows the current DNS server for domain name interpretation. It is possible
that some domain names are blocked by local DNS servers. Changing this to an
overseas DNS server may solve the problem.
g) VPN - This field shows the VPN status. If it is connected, the local IP is also shown. (Not yet available)
3. The right hand column shows the parameters and settings for each GSM channel. They are summarized
as follows: 增加了 REMOTE SIM
a) Channel x - This field shows the GSM module status of the channel x. "ON" means that the power to
the GSM module is shut down. You can then process to check other channel information. "OFF"
means that the power to the GSM module is switched off. You must first switch on the power to the
GSM module via the web interface. This setting is under the GSM Channel Shut Down which is
located under the Tools menu.
b) SIM –This field shows "INSERTED" if the SIM card inserted in the SIM slot "x" is detected. It displays
"NOT INSERTED" if no SIM card is detected. Please make sure that the SIM card is inserted properly
as described in Section 2. It is also possible that SIM card is damaged or incompatible. Please test
the SIM card with a cell phone to confirm if the SIM card is working properly. Please contact
technical support for further assistance.
c) IMEI - This field shows the IMEI number in use.
d) Signal –This field shows the signal strength of Channel x. Voice call quality with a signal strength
above 15 is good and below 15 is poor. For poor signal reception, it is recommended that the GoIP
should be re-located or an antenna with external cable should be used for better signal reception.
Signal strength of 99 means that the GSM channel is logout (not registered to any Carrier).
e) Login –This fields shows whether the GSM channel "x" is registered to a Carrier or not. "OK" means
the GSM channel is ready for making and receiving calls. If "Fail" is displayed, please check if
corresponding SIM and Signal are showing the correct status. Please insert the SIM card to a cell
phone to test if it is working properly. Contact technical support if the SIM card is valid but
registration to the carrier fails.
f) Details –Click on this expands the window to show additional channel information.
i. Module –This field shows the model number of the GSM module used.
ii. Number –This displays the phone number of the SIM card. User needs to entered this manually.
iii. Operator –This field shows the name of the GSM Service provider registered.
iv. Base Station - This field shows the reference number of the base station connected.
v. Remain Time –When the Total Talk Time limit of the channel is set, this field displays the
remaining talk time. When it is zero, the channel is disabled for all outgoing calls. Click on the
[Reset] button reset the Remain Time to the Total Talk Time limit. To enable the channel, Each
channel can be activated to operate for a certain period of time. The channel is then enabled for
outgoing calls.
Detail<<
--Module: G610

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--Service Provider: CHINA MOBILE
--Number:
--IMSI: 460026009838591
--ICCID: 898600
--Remain Time: NO LIMIT
--Module Ver: G610_V0C.58.0D_T17
g) Call Settings –Click on this expands the window to show additional parameters.
i. Call Waiting –This shows the make of the GSM module used.
ii. Call Forward –This shows the name of the Carrier registered.
Always - When it is "On", all incoming calls are forwarded unconditionally.
Busy - When it is "On", incoming calls are forwarded when the channel is in use.
No Answer - When it is "On", incoming calls are forwarded when they are not answered after x
rings.
No Service - When it is "On", incoming calls are forwarded when the channel fails to register to
the carrier.
3.3 Configuration
Click “Configuration”on the left hand column to display the Configuration page and the following submenu.
1. Preference
2. Network
3. VoIP
4. Call Management
5. GSM Service Provider
6. GSM Base Station Setting

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3.3.1 Preference
The preference page shown above consists of the following system level parameters and options as shown in
the table below.
Item
Parameter
Description
Default Value
1
Language
This sets the webpage and voice prompts language. Currently, only
English and Simplified Chinese (Mandarin for voice prompt) are
supported.
2
Time Zone
This specifies the offset of the local time zone with respect to GMT.
The syntax should be “GMTx”where x is the offset.
3
Time Server
This specifies IP address or the domain name of a network time server
for computer clock synchronization. The default is “pool.ntp.org”.
4
Auto-provision
Provision
Server
Provision
The auto provision is optional. When this option is enabled, the device
downloads its configuration from the Auto Provision Server at start up
or at the time interval specified by the Provision Interval. The
configuration file name is <Serial Number>.cfg which is just a text file
(not encrypted). If encrypted format is required, please contact
technical support for further assistance.
Please note that Auto Provision Server is a free utility supporting both
Linux and Window environment. Please visit our website or contact
technical support for more information.
The specifies the Provision Sever address (IP or Domain name)
This specifies the interval in performing an auto provisioning event.

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Interval
5
Remote Control
Remote Server
Remote Server
Port
Remote Server
ID
Remote Server
Password
This is a unique feature that allows remote access to the device's built-in
Web server even when it is installed behind NAT. To achieve this
function, a Remote Control Server is required to be installed. This
server is a free Linux based utility and is available for download via our
website. Please contact technical support for further assistance if
required. Once installed, please make sure that the Remote Server
Port and Password are set properly.
This specifies the IP address or the domain of the Remote Control
Server.
Check with your Remote Server administrator for the communication
port.
This specifies the name to be appeared in the Remote Control Server.
It is used as a reference for the device.
This specifies the login password to the Remote Control Server. This is
not the password to login to the built-in webpage. Please ask your
Remote Server Administrator if it is not available.
1920
6
Network Tones
Network tones are the tones associated with the traditional (PSTN)
telephone network, such as dial tone, ring back tone, busy tone, call
waiting tones, etc. These tones will only be used when the device
answers an incoming call and the call is not forwarded to a SIP server
automatically. Predefined Network Tones are classified by country
name. If the country desired is not found in the list, the “Custom”
selection allows users to define the network tones individually. Please
refer to Appendix B for more information.
7
DDNS
DDNS Address
DDNS Port
Update
Interval
This is a proprietary DDNS service offered by HYBERTONE. It allows
HYBERTONE's products to identify each other via this DDNS service.
When this service is activated, the domain name of the device is its
<serial number>.com. This feature is useful to support peer-to-peer
configuration.
The default DDNS Address is “voipddns.net”which a free service offered
by HYBERTONE. Please contact your vendor if you want to install your
own DDNS server.
The default communication port number is 39800.
This specifies the interval between registrations to the DDNS.
8
Auto Reboot
Reboot Time
This option allows the device to reboot itself at the time defined by
Reboot Time.
This parameter specifies the time to reboot the device and t is defined
in 24-Hour format (HH:MM).
9
IVR
The device is equipped with a simple voice prompt. When this option
is enabled and a call is answered, the device plays a voice prompt
instead of a dial tone to the caller.

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10
Remote SIM
Server
ID
Password
Only the GoIPs with the serial number xxxx support the Remote SIM
feature.
Enabling this feature allows the SIM Cards to be installed in a SIM Bank
rather than in the on-board SIM slots. GoIP can either register to a
SIM Bank or a SIM Server. Please refer to the SIM Bank User Manual
for more information.
This specifies the IP address of the SIM Bank or the SIM Server.
This specifies the name to be appeared in the SIM Bank or the SIM
Server.
This specifies the login password to the SIM Bank or the SIM Server.
11
DTMF Tone
Dropout (200 -
400)
This parameter specifies the maximum dropout time for a DTMF tone.
When making a call from SIP to GSM
or from GSM to SIP by using the
second dial method, the device
needs to detect the dialing digits
from the DTMF tones received via
the voice data stream. Depending on the network conditions, short
dropouts may occur due to packet jitter / loss. Therefore, DTMF digit
may be detected more than once if these dropouts are not taken into
account. Consequently, the call is dialed to an incorrect number. To
avoid this problem, a dropout window is used to avoid false detection
when dropouts occur. During this window, the same DTMF digit is not
recognized more than once.
The range of the dropout window is specified in terms of packet
timestamp value. The smaller the value is, the smaller the dropout
window is. This increase the chance of detecting the same digit twice
or more. However, if the value is set too large, there is a possibility
that the next digit is missed
3.3.2 Network
Proper network environment is the key to insure the voice call performance of the device. In general,
Intranet offers a more stable network environment than Internet and it is the preferred network to be used.
If Internet is going to be used, please make sure that the network can offer low packet loss, small packet
jitter and low packet delay. Each voice channel requires less than 90 kbps when A-law or -law voice codec
is used. GoIP-8 will require 8 times this bandwidth. Therefore, it is very important to make that both
upsteam and downsteam have enough bandwidth (+ 30% headroom) in order to accommodate the data
traffics for the device installed when all lines are used simultaneously.
In order to get external network access, the LAN port must be configured according to the network
environment to be connected.
LAN Port

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There are 3 access methods available to configure the LAN port.
1. Static IP –This mode applies to both public and private IP
network environment. In the LAN port configuration
shown on the left, select “Static IP” and then fill in the
parameters as provided by your network administrator.
2. DHCP (default setting) –When the device is installed behind
NAT and a DHCP host is available, select “DHCP”to enable
the device to obtain LAN IP address and other network information automatically.
3. PPPoE –ADSL and Cable modems very often use PPPoE dial
up to obtan network IPs. If this is the case, select “PPPoE”
and then enter the information as provided by your ISP.
PC Port
The PC Port allows other network devices to be attached to the device in order get network connection. It
offers both Router and Bridge modes to meet your requirements.
1. Static IP (default setting) –This mode enables the device
to create another network segment and it then
functions as a router/gateway for this new network
segment. Select “Static IP”for this new segment and
then enter the PC port IP address and Subnet Mask
accordingly.
It also has a built-in DHCP server to assign IPs to the
devices attached to this network segment. Enable it
and then enter the “Starting Addrss”, “Ending Address”,
and “Static DNS”as required.
As a factory default, the PC port is set to “Static IP”(Router mode) with IP Address set to 192.168.8.1 and
Subnet Mask to 255.255.255.0.
2. Bridge Mode –Select this mode if your network topology
requires the network devices attached to the PC port to
be in the same network segment as the LAN port.
Advanced Features
1. VLAN –This is a type QoS service and is intended to give
higher transmission priority to real time packets.
However, your router switch and ISP network need to
support this feature as well.
2. PPTP VPN –This option allows the device to create a VPN
tunnel with the designated VPN Server. The VPN
protocol supported is PPTP with no encryption or 40-bit
encryption which is defined on the VPN server. In

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general, this option is used to avoid VoIP blockings.

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3.3.3 VoIP
This section defines all VoIP related settings. The device supports both SIP and H.323 protocols. For GoIP
(1 channel) and earlier version of GoIP-4 / 8, both protocols are embedded in a single firmware version.
User must select the desired protocol via the parameter “End Point Type”as shown below in the VoIP
Settings section.
As more features are added, SIP and H.323 VoIP protocols are supported in two different firmware versions.
GoIP-4 and GoIP-8 are now shipped with the SIP protocol firmware as a factory default. If H.323 protocol is
required, the firmware of the device can be changed to the one that supports H.323 protocol. Please visit
our website for the latest firmware versions or contact your supplier for more information.
To find out the protocol installed in the device, please check the firmware version as described below.
1. SIP firmware version is in the format GS-4.01-xx…
2. H.323 firmware version is in the format GH-4.01-xx…
Please contact your vendor or visit our website for the upgrade link to the latest version. The upgrade
method is discussed in Section x.x. You can always switch the firmware from SIP to H.323 or vice versa.
In general, it is important to understand your VoIP application with the device before proceeding to device
configuration. If the device is going to work with a IP PBX, please make sure that you know how to
configure your IP PBX. It is very important that you send us your application requirements in full details
when seeking for technical support in configuring the device.
3.3.3.1 SIP
SIP settings are categorized into three groups of parameters in order to simplify the configuration. Click on
“VoIP”selection on the left column menu to access the SIP Settings page.
Depending on the models, a total of four configuration modes are supported.

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1. Single Server Mode (for models GoIP, GoIP-4, and GoIP-8)
In this mode, only one SIP registration is used for single or multiple-line operation. Please make sure
that your SIP server supports multiple-line operation and the SIP account is configured for the number of
lines that matches the number of lines available in the device. When receiving a call from the SIP server
(outgoing call), it picks the idle GSM channel that is used the least (in terms the number of calls dialed) to
dial out the call. When a GSM channel receives an incoming call, the call can either be answered by the
device or forwarded to a SIP extension or IVR. Please see more details in the Section x.x for Call
Management.
Parameter
(Single Server Mode)
Description
Default Value
1. Phone Number
2. Authentication ID
3. Display Name
4. Password
5. SIP Proxy
6. SIP Registrar Server
7. Re-register Period
(s)
8. Outbound Proxy
9. Home Domain
10. Backup Server
SIP Proxy
SIP Registrar Server
Home Domain
This is the SIP number used by the device.
The name of the device used in Caller Identification is defined here.
The Authentication ID used for SIP registration is specified here.
The password used for SIP registration is specified here.
The address of the SIP Proxy or Server is specified here.
The address of the SIP Registrar Server is specified here.
Register to the SIP Server at an interval specified by this parameter.
The address of the Outbound Proxy used for VoIP communication is specified here.
Home Domain is used in SIP identification. It should be specified as required.
Backup Server improve service reliability and is used only when the primary server
fails.
This specifies the backup SIP Server address.
This specifies the backup SIP Registrar Server address.
This specifies the backup Home Domain address.