MTG 3000 VoIP Trunk Gateway User Manual v1.0
MTG 3000 VoIP Trunk Gateway User Manual v1.0
MTG 3000 VoIP Trunk Gateway User Manual v1.0
Address: 9th Floor, Guoxing Building, Changxing Road, Nanshan District, Shenzhen, China
Website: www.dinstar.com
Welcome
Thanks for choosing MTG3000 Trunk Gateway! We hope you will make optimum use of this flexible,
rich-feature trunk gateway. Please read this document carefully before install the 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 this document are aimed at users who will actually use the gateway.
Revision Records
Document Name MTG3000 Trunk Gateway User Manual v1.0
Document version V1.0
Firmware version 2.05.03.03
Revised by Ivanka Yuan
Date March 3, 2016
Table of Contents
1.1 Overview..................................................................................................................................................... 1
5 Abbreviation................................................................................................................................................ 83
6 Commands................................................................................................................................................... 84
MTG3000T supports the conversion of multiple coding methods such as G.711A/U, G.723.1, G.729A/B,
iLBC and AMR. It has good compatibility with Huawei SoftX3000, and other softswitches and IMS
systems from ZTE, Cisco, VOS and Langxun.
Compared to other similar products, MTG3000 has more advantages in terms of performance, system
reliability and compatibility. It is widely used by call centers, operators and large-size enterprises for the
entry into IMS network based on VoIP.
MTG3000 Enterprise
IPPBX
SDH PRI/SS7
SIP
Internet Users
MTG1000 PBX
Optical Transceiver
E1/T1
Telephone
Receptionist
PSTN
Call Center
DMCloud Softswitch
Front View
Back View
MCU Board:
PWR On The MCU board has been inserted and connected properly to the
MTG3000 box.
Off The MCU board is faulty or it is not connected properly to the
MTG3000 box.
RUN Flash slowly The MCU board runs normally.
Flash quickly The MCU board is faulty or it is not connected properly to the
MTG3000 box.
CONSOLE / The console port used to carry out maintenance-related
configurations, with a baud rate of 115200bps
The gigabit network port for services, which is used to realize
the data transmission of signal or voice. Its default IP address is
192.168.1.111, and default netmask is 255.255.255.0.
DTU Board:
Indicator/Port Status Description
PWR On The DTU board has been inserted and connected properly to the
MTG3000 box.
Off The DTU board is faulty or it is not connected properly to the
MTG3000 box.
RUN Flash slowly The DTU board runs normally
Flash quickly The DTU board is faulty or it is not connected properly to the
MTG3000 box.
On The DTU board is properly connected to the MCU board
Link 0 Off The DTU board is not or improperly connected to the MCU
board, or the DTU board is faulty.
Link 1 Reserve Reserve
SDH Board:
Indicator/Port Status Description
PWR On The SDH board has been inserted and connected properly to the
MTG3000 box.
Off The SDH board is faulty or it is not connected properly to the
MTG3000 box.
RUN Flash slowly The SDH board runs normally
Flash quickly The SDH board is improperly connected to the MTG3000 box
On The SFP optical module has been inserted into the SFP port 0.
Light 0 Off The SFP optical module has not been yet inserted into the SFP
port 0.
Ethernet Interface
GE1: 10/100/1000 BaseT Adaptive Ethernet
GE0: 10/100/1000 BaseT Adaptive Ethernet
Console Port
1* RS232, 115200bps
Both GE1 and GE0 can be used to carry out management on MTG3000, but only GE1 is put in use
generally. GE0 is used when there is a need to separate the management on MTG3000 from the service
processing of the MTG3000.
Optical Transceiver
TX RX
MTG3000
0 1
PWR 1 SDH
TX RX
RUN 0
SFP0
2.3 Troubleshooting
When the MTG has been connected to the optical transceiver, but light 0 on the SDH board is still dull or it
flashes, please check according to the following steps.
a. Check whether the MTG3000 gateway is properly connected to the optical fiber.
b. Switch the Tx port with the Rx port of MTG3000
c. Check whether the numbers of the two ends of the optical fiber are the same.
d. Carry out a loopback test.
2. On the PC, open the TCP/IP Settings interface, click Advanced, and then click Add to add an IP
whose format is 192.168.1.XXX. Or you can open the Internet Protocol (TCP/IP) interface to modify an
existing IP into 192.168.1.XXX.
Step 2: Connect the F port of the serial cable to the COM port of PC.
If the PC does not have a COM port, please use a USB-to-COM converting tool to connect the serial cable
to the PC.
Step 3: Connect the M port of the serial cable to the console port of MTG3000.
Enter username and password, which are the same with the username and password of the Web of
MTG3000. And then you will see a linux platform where you can carry out maintanance-related
configurations.
Note: For commands to query MTG3000 information, make reference to Chapter 6 of this manual.
1. Use a serial line to connect the console port of MTG3000 with a PC;
3. Click OK, and then enter ‘ifconfig’, and the IP address of GE1 or GE0 of MTG3000 will be displayed.
Internet
Router
Switch PC
MTG3000
Click Start Run on the PC and enter cmd to execute ‘ping 192.168.1.111’ to check whether the IP
address of the MTG3000 runs normally.
It is suggested that you should modify the username and password for security consideration on the
Maintenance Password Modification interface.
No
No
No
Channel status:
Select DTU card number to check the channel status of each DTU board.
Parameter Explanation
Trunk Name This trunk name is the name used to register the SIP trunk. If the
SIP trunk is not registered, the trunk name is displayed as “---”.
Trunk Mode There are two trunk modes: peer (peer-to-peer) and access.
ASR (Answer-seizure Ratio): is a call success rate, which reflects the percentage of answered telephone
calls with respect to the total call volume. ASR = answered call/total attempts of calls.
ACD (Average Call Duration): is a measurement in telecommunication, which reflects an average length
of telephone calls transmitted on telecommunication networks. ACD = total call duration/total connected
calls.
4.5 Network
Generally, it’s necessary to modify the default IP address of GE1 according to actual network conditions,
and then modify the IP address of PC to make it at the same network segment with the IP address of GE1.
After completing the configurations, you need to restart the MTG3000 device for the changes to take effect.
Note: The IP address of GE1 and that of GE0 cannot be at the same network segment.
Note:
Please ensure parameters related to J0, C2, J1, V5 and J2 are consistent with those configured on optical
transceiver.
LOS Alarm
SFP There are two SFP optical modules, namely SFP 0 and SFP 1.
Onboard Display the status of SFP optical modules. Online means SFP optical module
has been inserted. Offline means SFP optical module is not inserted.
Used Display SFP port is enabled or disabled.
LOS Alarm Green: signal is received.
Red: signal is lost (the reason may be the optical fiber is not connected
properly).
CDR Alarm Green: CDR successfully parses data signal and clock
Red: CDR fail to parses data signal and clock
Clock Green: Clock is parsed and locked
Red: Clock fails to be parsed
Parameter Options
Calling Party Numbering Plan Include ‘ISDN/Telephony Numbering Plan’, ‘Data Numbering
Plan’, ‘Telex Numbering Plan’, ‘National Standard Numbering
Plan’, ‘Private Numbering Plan’ and ‘Unknown’.
Calling Party Number Type Include ‘International Number’, ‘National Number’, ‘Network
Specific Number’, ‘Subscriber Number’, ‘Abbreviated Number’
and ‘Unknown’.
Screening Indicator for Displaying Include ‘User-provided, not screened’, ‘User-provided, verified
Caller Number and passed’, ‘User-provided, verified and failed’,
‘Network-provided’
Screening Indicator for No Include ‘User-provided, not screened’, ‘User-provided,
Displaying Caller Number verified and passed’, ‘User-provided, verified and failed’,
‘Network-provided’
Called Party Numbering Plan Include ‘ISDN/Telephony Numbering Plan’, ‘Data Numbering
Plan’, ‘Telex Numbering Plan’, ‘National Standard Numbering
Plan’, ‘Private Numbering Plan’ and ‘Unknown’.
Called Party Number Type Include ‘International Number’, ‘National Number’, ‘Network
Specific Number’, ‘Subscriber Number’, ‘Abbreviated Number’
and ‘Unknown’.
Information Transfer Capability Include ‘Speech’ and ‘3.1 kHz audio’
Click the Add button, and you can add a PRI trunk. If you want to delete or modify the information of a
PRI trunk, select the checkbox on the left of the trunk, and then click the Delete button or the Modify
button.
Parameter Explanation
Trunk No. Trunk No. starts from 0 to 19, it means you can establish 20
PRI trunks at most.
The trunk No. is decided by the No. of the E1/T1 port linked
to the trunk. But if D-channel is not enabled for a trunk, the
No. of the trunk must be the same with a trunk under which
D-channel has been enabled.
Trunk Name The trunk name is used to distinguish the trunk from other
trunks.
Channel ID The ID of the channel selected for the PRI trunk. The
channel ID is used for the switch to identify a PRI trunk in
case that the Trunk No. of two trunks are the same.
D-Channel The channel used to carry control information and signaling
(Delta Channel) information
E1/T1 Port No. The No. of E1/T1 port linked to the PRI trunk
Protocol Support two protocols: ISDN and QSIG. Default value is
ISDN.
Switch Side The EI/T1 port of the PRI trunk is taken as User Side or
Network Side.
Alerting Indication Include Alerting and Progress
Alerting: Play ring-back tone when receiving alerting signal
Progress: Play ring-back tone when receiving progress signal
Auto Reset Circuit indicates that the gateway send “GRS” or “CGU” message to the switch side to initiate
circuits when the MTP links up. Disable this message in case of the switch side doesn’t response to “GRS”
properly.
Select “enable/disable” through drop box.
Parameter Explanation
Trunk No. The No. of the SS7 trunk. Generally, one SS7 trunk is for one DPC.
Trunk Name The trunk name is used to distinguish the trunk from other trunks.
Protocol SPC types: ITU-T (14 bit), ANSI (24 bit), ITU-CHINA (24 bit)
SPC: Signaling Point Code
Protocol Type ISUP (ISDN User Part) and TUP (Telephone User Part)
Parameter Explanation
Link No. Each SS7 trunk supports two links which share the loading equally. If
one link malfunctions, the other link will automatically bear all the
loading until the faulty link is restored.
Signaling Link Code If the Link No. of the trunk cannot match with that of the peer device,
the SS7 trunk will be linked to the peer device according to signaling
E1/T1 Port No. The No. of E1/T1 port linked to the SS7 trunk
Channel No. The No. of the channel under which signal is transmitted. Default value
is 16.
Caller Type The type of the caller number. Options include ‘Not Configured’,
‘Subscriber’, ‘International” and “National’.
Callee Type The type of the called number. Options include ‘Not Configured’,
‘Subscriber’, ‘International” and ‘National’.
OrgCallee Type The type of the original called number in case of number manipulation.
Options include ‘Not Configured’, ‘Subscriber’, ‘International’ and
‘National’.
Step 2: Select a trunk and an E1/T1 port. (Trunk 0 and Port 1 are taken as example in the following figure
Parameter Explanation
Trunk No. The No. of the SS7 trunk
Start E1/T1 Port No. The No. of the start E1/T1 port
End E1/T1 Port No. The No. of the end E1/T1 port
Start Channel When the start E1/T1 port is also the end E1/T1 port, it’s required to set
the start channel, and the channels starting from the set channel to the
No.31 channel of the E1/T1 port will be used by the SS7 trunk.
Start CIC No. CIC: Circuit Identification Code
The CIC No. of the start channel, which is generally 0, 32, 64, 96, 128,
160, 192, 224, 256, 288, 320, 352, 384, 416, 448…
Count The total number of the channels used by the SS7 trunk
Step3: Click OK. And then you can see the following data on the SS7 CIC interface.
Step 2: Select a trunk and E1/T1 ports. (Trunk 1, Port 0, Port1 and Port 2 are taken as example in the
following figure.
Step3: Click OK. And then you can see the following data on the SS7 CIC interface.
Status The E1/T1 ports have 16 statuses, including Activated, Disabled, Fault, RAI
Alarm, AIS Alarm, ISDN/SS7 Signal Alarm, Frame-Sync, Idle, Signal, Busy,
L-block, R-blocked, B-blocked, Blocking, Unblocking and Resetting.
Meanwhile, you can carry out maintenance on the E1/T1 ports through the following buttons: Select All,
Invert, Clear, Block, Unblock, Reset and Cancel.
Select Channel on the right of Operation Mode, and then select an E1/T1 port, the channels of the E1/T1
port and their statuses are displayed.
Meanwhile, you can carry out maintenance on the channels of E1/T1 ports through the following buttons:
Select All, Invert, Clear, Block, Unblock, Reset and Cancel.
Parameter Explanation
Select Clock Source Mode If Remote is selected, clock source is produced by crystal chip;
if local is selected, clock source is obtained from the data
received by E1/T1 port.
Select Remote Clock Source Port The No. of the E1/T1 port from which clock source is
obtained.
Automatic Clock Protect Clock source is protected automatically.
Parameter Explanation
Parameter Explanation
Packetization Time (ms) The minimum packetization time of voice codec. For example, if
packetization time is 20ms, voice will be packetized every 30ms.
Rate (kbps) Transmission rate of voice
Silence Suppression If silence suppression is enabled, the bandwidth occupied by voice
transmission will be released automatically for the silence party or
when talking is paused.
Default value is ‘Disable’.
Step1. Enter into the Codec Group interface and select codec group ID 1 to create new codec group
Step2. Select preferred voice codec (G711A and G729) in this example, as below:
Step5. Enter into the PSTN Group interface to establish a PSTN group.
Step6. Enter into the PSTN Group Management interface to associate the PSTN profile and PSTN
group to an E1/T1 port or multiple E1/T1 ports.
can be divided into 5 groups with dial plan IDs. The setting in dial plan 0 is the default setting.
Click the Add button, and you can add a new dial plan in the following interface.
Parameter Explanation
Dial Plan ID The ID of the dial plan
Index Each dial plan has a unique index. Greater index value, higher priority for
the dial plan.
Prefix The prefix matching received numbers, through which the MTG3000 can
judge how many digits the received number includes.
Note:
1. Dial plans can be backed up and restored at the Maintenance Data Backup interface and the
Maintenance Data Restore interface respectively.
2. ‘Min Length’ and ‘Max Length’ does not include the length of prefix.
3. For overlapping dialing, it’d better to set ‘Min Length’ and ‘Max Length’ to a same value in order to
accelerate connection rate, since the length of the called number has been known.
The setting in Dial Timeout 0 is default setting, which can be modified but cannot be deleted.
Max Time for Collecting Prefix The maximum time for receiving all the digits of a prefix
Time to Reach Min Length After receiving the prefix, the maximum time before receiving the
(after Prefix) set minimum number of digits included in a telephone number.
Time to Reach Max Length After receiving the set minimum number of digits, the maximum
(after Min Length) time before receiving the set maximum number of digits included in
a telephone number.
Parameter Explanation
group.
Channel Selection Ascending: to search idle channels starting from channel 0 to channel
31;
Control Mode Options include Master Odd, Master Even and None.
Master Odd: it means channels with odd ID will be searched first, and
channels with even ID will not be searched until all channels with odd
ID have been searched.
Click the Add button, and you will see the following configuration interface.
In the above figure, as start E1 is the same with end E1, only one E1 port is used in the PSTN group and
you need to set start channel and end channel.
When there is a need to set several E1 ports, it defaults that all the 32 channels of each E1 port are used by
the PSTN group.
PSTN Profile ID The ID of the PSTN profile in this PSTN group (the PSTN profile needs to
be created at the PSTN Profile interface first.
Note: When the start E1/T1 port is different from the end E1/T1 port, the start channel is channel 0 by
default and the end channel is channel 31 by default (it means there is no need to choose a start channel and
an end channel).
2. Configure parameters on the SIP Trunk Add interface according to related explanations in the table.
As it is Peer mode, you should select No for the Register to Remote parameter, and enter the IP address of
the peer device.
Outbound Proxy Protocol Type Options include UDP, TCP and Auto
If Auto is selected, the protocol type is determined by the peer
device.
Outbound Proxy Port (UDP) The default outbound proxy port is 5060.
Local Domain The local domain set in the SIP Parameter interface
Get Callee from Get the called number from ‘Request-line’ or ‘To Header Field’
Get Caller from Get the caller number from ‘User Name’ or ‘Display Name’
Outgoing Calls Registration Whether to limit the number of the calls from PSTN to IP
network.
The default value is ‘No’.
If ‘Yes’ is selected, then input the number of concurrent calls
that are allowed to go out. The range is 0 to 65535.
Incoming Calls Registration Whether to limit the number of the calls from IP network to
PSTN.
The default value is ‘No’.
If ‘Yes’ is selected, then input the number of concurrent calls
that are allowed to come in. The range is 0 to 65535.
Incoming Time Registration The default setting is ‘Disabled’.
If ‘Enabled’ is selected, user can edit the start and stop time of a
prohibition period. During this period, all calls from IP network
to PSTN are prohibited. (Calls from PSTN to IP network are not
limited)
Detect Trunk Status Whether to detect the status of the SIP trunk. If ‘Yes’ is selected,
MTG3000 will send Heartbeat message to the peer device to
confirm whether the link status is OK.
Heartbeat Username The name of the Heartbeat message
Enable SIP Trunk Whether to enable the SIP trunk.
If ‘Yes’ is selected, the SIP trunk is available;
If ‘No’ is selected, the SIP Trunk is invalid.
3. Click OK.
4. Click SIP Account in the navigation tree on the left, and then click Add to add a SIP account.
Parameter Explanation
SIP Account ID The ID of SIP Account, from 0 to 127
Description Description of the SIP account
Binding PSTN Group Choose a PSTN group that is bound to the SIP account
SIP Trunk No. The No. of the SIP trunk bound to the SIP account
Username The username of the SIP account, which is used to register the SIP
account to softswitch
Authenticate ID The authentication ID to authenticate the SIP account for the
softswitch connected to MTG3000
Password The password of SIP account, which is used when the SIP account
is registered to softswitch
Confirm Password Enter the password again
Expire Time The interval to register the SIP account; Default value is 1800s.
6. Click OK. And you can click Status & Statistics IP Trunk Status to check the SIP trunk that has
been established.
4.11.1 IP Profile
On the IP Profile interface, you can configure the parameters about IP calls, such as whether to support
Parameter Explanation
IP Profile ID The ID of the IP profile, from 1 to 15.
Description Description of the IP profile
Ringback Tone to PSTN Originated Where the ringback tone to PSTN side is originated from
from If ‘Local’ is selected, the ringback tone is played from MTG3000.
If ‘IP’ is selected, the ringback tone is played from the IP network
Ringback Tone to IP Originated Where the ringback tone to IP network l is originated from
from If ‘Local’ is selected, the ringback tone is played from MTG3000.
If ‘PSTN’ is selected, the ringback tone is played from the PSTN
4.11.2 IP Group
On the IP Group interface, you can add IP groups and choose a strategy for selecting IP trunks.
Parameter Explanation
IP Group ID The ID of the IP group
If you want to add more IP trunks to the IP group, do not change the IP
group ID.
Index The index of the IP trunk added to the IP group
Trunk Type SIP
Trunk No. Select an IP trunk that has been established on SIP Config SIP Trunk
interface.
IP Profile ID The ID of the IP profile that will be used by the IP trunk.
Caller White List: Calls from the numbers on the Caller White List will be allowed to pass. If a caller
number cannot match with one of the numbers on the Caller White List, calls from the caller number will
be rejected.
Callee White List: Calls to the numbers on the Callee White List will be allowed to pass. If a callee
number cannot match with one of one of the numbers on the Caller White List, calls to the callee number
will be rejected.
Callee Black List: Calls to the numbers on the Callee Black List will be rejected to pass. If a callee number
match with one of the numbers on the Callee Black List, calls to the callee number will be rejected.
2. Click Add to enter into the following interface to add a caller number on the Caller White List
3. Choose an ID for the caller white list and an index for the caller number, and then enter the caller
number
4. Click OK.
Note:
You can add 8 white or black lists at most, with ID from 0 to 7. And each white or black list can contain
1024 numbers at most.
Note:
If ‘Starting Caller Number’ is 80080000 and ‘Number Count’ is 100, it means numbers from 80080000 to
80080099 are all in the caller pool.
Each caller poor can contain 512 numbers at most, and if there are multiple caller pools, the caller pools
can contain up to 1024 numbers in total.
If you select 255<None>, it means no while lists or black lists are set in filter profile, and no numbers will
be filtered.
Parameter Explanation
Index The Index of the PSTN IP route, from 0 to 255. Greater index
value, higher priority for the route.
Description The description of the PSTN IP route,
Number Filter Profile ID The ID of filter profile. The white lists and black lists set in the filter
profile will apply to this PSTN IP route.
Destination IP Group If source is PSTN group, please select a specific PSTN group.
IP Trunk No. If source is PRI/SS7 trunk, please select a specific PRI/SS7 trunk.
Number Filter Profile ID The ID of filter profile. The white lists and black lists set in the filter
profile will apply to this PSTN PSTN route.
Destination IP Group If source is PSTN group, please select a specific PSTN group.
IP Trunk No. If source is PRI/SS7 trunk, please select a specific PRI/SS7 trunk.
Number Filter Profile ID The ID of filter profile. The white lists and black lists set in the filter
profile will apply to this PSTN PSTN route.
Parameter Explanation
Index The Index of the IP IP route, from 0 to 255. Greater index value,
higher priority for the route.
Description The description of the IP IP route,
Number Filter Profile ID The ID of filter profile. The white lists and black lists set in the filter
profile will apply to this IP IP route.
Suffix to be added The suffix added to the callee number after its digits are lessened.
Number of Digits to Reserve The number of the retained digits which. are counted from the right
from Right of the callee number
For example:
If the called number is 25026531014, how do you change it into 026531014 ?
You can enter ‘3’ in the value box for the ‘Number of Digits to Strip from Left’ parameter.
Parameter Explanation
Index The index of this PSTN IP caller number manipulation, from 0 to
127. Each index cannot be used repeatedly.
Description The description of this PSTN IP caller number manipulation
PSTN Group Select a PSTN group. The caller number will be manipulated when a
call uses a trunk of this PSTN group, actual callee prefix matches the
set callee prefix, and actual caller prefix matches the set caller prefix.
Suffix to be added The suffix added to the caller number after its digits are lessened.
Number of Digits to Reserve The number of the retained digits which. are counted from the right
from Right of the caller number
1st Number Type If the caller number belongs to 1 st number type, the set prefix will be
added to the caller number.
st
Add Prefix for 1 Number Type The prefix that will be added to those numbers that belong to 1 st
number type
nd
2 Number Type If the caller number belongs to 2nd number type, the set prefix will be
added to the caller number.
Add Prefix for 2nd Number Type The prefix that will be added to those numbers that belong to 2 nd
number type
Parameter Explanation
Index The index of this PSTN PSTN callee number manipulation, from
0 to 127. Each index cannot be used repeatedly.
Suffix to be added The suffix added to the callee number after its digits are lessened.
Number of Digits to Reserve The number of the retained digits which. are counted from the right
from Right of the callee number
Number Type The type of the callee number. Options include ‘Not Config’,
‘International’, ‘National’, ‘Unknown’, ‘Network Specific’,
‘Subscriber’ and ‘Abbreviated’
Number Type The type of the caller number. Options include ‘Not Config’,
‘International’, ‘National’, ‘Unknown’, ‘Network Specific’,
‘Subscriber’ and ‘Abbreviated’
Parameter Explanation
Index The index of this IP PSTN callee number manipulation, from
0 to 127. Each index cannot be used repeatedly.
Description The description of this IP PSTN callee number manipulation
IP Group Select an IP group. The callee number will be manipulated when
a call uses a trunk of this IP group, actual callee prefix matches
the set callee prefix, and actual caller prefix matches the set
caller prefix.
4.14.7 IP IP Callee
On the IP IP Callee interface, you can set rules to change the actual callee number during IP IP
calling process.
Parameter Explanation
Index The index of this IP IP callee number manipulation, from 0 to
127. Each index cannot be used repeatedly.
Description The description of this IP IP callee number manipulation
4.14.8 IP IP Caller
On the IP IP Caller interface, you can set rules to change the actual caller number during IP IP
calling process.
Caller Prefix Set a prefix for the caller number. If the actual caller prefix matches
the set prefix, the caller number will be manipulated.
Number of Digits to Strip from The number of digits which are lessened from the left of the caller
Left number
Number of Digits to Strip from The number of digits which are lessened from the right of the caller
Right number
Prefix to be added The prefix added to the caller number after its digits are lessened.
Suffix to be added The suffix added to the caller number after its digits are lessened.
Number of Digits to Reserve from The number of the retained digits which. are counted from the right
Right of the caller number
Voice Parameter Period without RTP The set maximum time without receiving RTP
packet packets.
Default value is 60 seconds.
Echo Cancel Time The interval to remove echo from a voice
communication.
Options include 32ms, 64ms and 128ms.
Gain from PSTN The voice gain from PSTN to IP direction
Default value is -1dB
Gain to PSTN The voice gain from IP to PSTN direction
Default value is 2dB
Ringback Tone Type Local ringback tone
Recognition Mode Whether to recognize voice when prompt tone
is played.
Call from PSTN The maximum time of no answer for calls from
Timeout of No Answer PSTN
Call from IP The maximum time of no answer for calls from
IP Network
Fax Mode Options include T.38, Pass-through and
Adaptive.
Default value is T.38.
Adaptive means auto negotiate with peer side.
Fax Parameter Fax Tx Gain Gain of sending a fax
Fax Rx Gain Gain of receiving a fax
Parameter Explanation
SIP Trunk No. The No. of the SIP trunk that transmits the SIP message to be
encrypted.
Encrypt Mode Only support VOS RC4 at present
Device ID The ID of the SIP account to which the SIP trunk belongs
Database herein refers to the database where configuration data are placed.
The abovementioned password is also used to log in Web Interface, Telnet and SSH.
DND Do-not-Disturb
6.1.13 Query Packet Statistics of HDLC Channel and Related Error Codes
Enter the command show mcc x (x refers to the port No. of HDLC channel), and the packet statistics and
error codes (if there are any) of the HDLC channel are displayed.