MTG1000 (B) Trunk Gateway User Manual V2.0: Dinstar Technologies Co., LTD
MTG1000 (B) Trunk Gateway User Manual V2.0: Dinstar Technologies Co., LTD
MTG1000 (B) Trunk Gateway User Manual V2.0: Dinstar Technologies Co., LTD
Website: www.dinstar.com
MTG1000(B) Trunk gateway User Manual
Revision Records
File Name MTG1000(B) Trunk Gateway User Manual
Document Version 2.0
Firmware Version 2.03.04.01/2.04.04.01
Date 03/10/2013
Revised by Technical Support Department
MTG1000(B) Trunk gateway User Manual
Content
1. Product Introduction ..................................................................................................................... 1
1.1 Overview ............................................................................................................................. 1
1.2 Equipment Structure ........................................................................................................... 1
1.2.1 Rear View ................................................................................................................. 1
1.2.2 Front View ................................................................................................................ 2
1.2.3 RJ-48c Line sequence ............................................................................................... 3
1.3 Functions and Features ....................................................................................................... 3
1.3.1 Protocol standard supported ................................................................................... 3
1.3.2 System Function ....................................................................................................... 4
1.3.3 Industrial standards supported ................................................................................ 4
1.3.4 General hardware specification ............................................................................... 4
2. Web Configuration Guide .............................................................................................................. 5
2.1 Login .................................................................................................................................... 5
2.2 Web interface structure and navigation tree ...................................................................... 6
2.3 Status & Statistics ................................................................................................................ 8
2.3.1 System Information .................................................................................................. 8
2.3.2 E1/T1 Status ............................................................................................................ 10
2.3.3 PSTN Trunk Status .................................................................................................. 11
2.3.4 IP Trunk Status ........................................................................................................ 11
2.3.5 PRI Call Statistics .................................................................................................... 12
2.3.6 SS7 Trunk Call Statistics .......................................................................................... 13
2.3.7 SIP Call Statistics ..................................................................................................... 13
2.3.8 H.323 Call Statistics ................................................................................................ 13
2.4 Network............................................................................................................................. 14
2.5 PRI Config .......................................................................................................................... 15
2.5.1 PRI Parameter......................................................................................................... 15
2.5.2 PRI Trunk ................................................................................................................ 16
2.6 SS7 Config .......................................................................................................................... 17
2.6.1 SS7 Trunk ................................................................................................................ 17
2.6.2 SS7 MTP Link .......................................................................................................... 19
2.6.3 SS7 Circuit............................................................................................................... 20
2.6.4 SS7 Circuit Maintain ............................................................................................... 21
2.6.5 Slave TG Management ........................................................................................... 22
2.7 PSTN Group Config ............................................................................................................ 23
2.7.1 E1/T1 Parameter ..................................................................................................... 23
2.7.2 Coder Group ........................................................................................................... 24
2.7.3 Dial Plan.................................................................................................................. 25
2.7.4 Dial Timeout ........................................................................................................... 27
2.7.5 PSTN Profile ............................................................................................................ 27
2.7.6 PSTN Group ............................................................................................................ 28
2.7.7 PSTN Group Management ...................................................................................... 29
2.8 SIP Config .......................................................................................................................... 30
MTG1000(B) Trunk gateway User Manual
1. Product Introduction
1.1 Overview
E1/T1 Trunk Gateway (Hereafter named ‘TG’ or ‘gateway’) aimed at operators and call center,
and used to help enterprise to realize the evolution from the traditional PBX to voice IP. On
the one hand, it supports PRI/SS7 protocol and adopts standard T1/E1 trunk interface to
realize docking with traditional PBX. On the other hand, adopt standard SIP protocol docking
with various soft switch to ensure PSTN seamless access to IP voice/NGN network, and
achieving VoIP/FoIP and more value-added service. The gateway supports intelligent multiple
trunk routing technology, makes the operator easy to manage trunk routing by price
optimum rule, and the automatic switch-over between multiple trunk routing makes the
E1/T1 Gateway has good call processing ability, and provides 1/2/4/8 T1/E1 interface. It is able
to handle a variety of signaling protocol and voice decoding. It supports the rich GUI
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trunk gateway adopts standard RJ-48C interface and iimpedance value is 120Ω. Connected end
device by cross lines sequence.
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2.1 Login
Firstly, connect network from pc to management port of gateway directly and input default IP
address in the browser. It will request customer to input user name and password. Default user
If customer modified the default IP or forgot the IP, that can’t enter the configuration page.
Please connect PC and device serial with the serial line. Enter the CLI to view or modify the
equipment IP. Here IP was set to 172.16.65.27. Enter the IP address of device in the browser
The default user name and password is "admin". To guarantee the system safety, when login for
the first time, please modify the default username and password Maintenance->Password
Modification.
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Users through to traverse the left navigation tree, and can complete view, edit and configuration
After entering configuration page, according to demand choose Chinese interface or English
Users through to traverse the left navigation tree, and can complete view, edit and configuration
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Open the operation of the navigation tree information node, and can view the device information
System information interface shows the general information and version information.
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5. ISDN/SS7 Signal Alarm: Means physical connection is normal, signaling link has
problem.
6. Active-OK: Means that physical connection and signaling link are normal.
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Remote blocked, means that communication can only be initiated from remote
9.B-blocked:
Both Sides blocked, means that the two sides cannot communication
E1/T1 Port No Indicate the E1/T1 line occupied by the SS7 trunk.
IP trunk status
SIP Trunk No The number of SIP trunk
Username When SIP trunk is under registered mode, change the value in the configuration
shown in the account registration, If SIP trunk is under non-registered mode, the
value is meaningless, as '---'
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This statistics page show the reasons for release of the call, including: Normal Call Clearing, Call
Rejected, User Busy, No User Response, No Circuit Available, Unassigned Number, Normal
Unspecified and others. Statistical information in an intuitive would be reflected on the pie char.
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The parameters of SS7 trunk call statistics are the same with PRI parameters. Please refer to PRI
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2.4 Network
Note:FE0 port IP and FE1 port IP should be set in different segments. After configure the
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Six optional types are provided for called party: Unknown, International number,
Called Party Number
National number, Network special number, User number, Short code dialing. The
Type
default option is Unknown.
Information Transfer
Support speech and 3.1khz audio. The default option is speech.
Capability
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Indicate PRI network property of E1/T1, it is divided into: “User side” and “Network
Switch Side side”. When PRI loopback is carried out, the network properties of E1/T1 port at both
receiving and sending sides must be different.
Alerting Indication The ring signal include Alerting and Progress
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Notes:
1. “Trunk No.” is a shared data, therefore, SS7 „Trunk No.‟ can't be the same as PRI “Trunk No.”
2. SPC length is 24bits when option “ANSI” or “ITU-CHINA” is selected in item “Standard Type”.
3. SPC length is 14bits when option “ITU” is selected in item “Standard Type”.
4. SPC Length represents the structure of OPC/DPC. SPC View Mode indicates which input
5. When SPC length is 24bits and 'Hex' are selected, the structure is like xyz, and x,y,z must be
6. When SPC length is 14bits and 'ITU Pointcode Structure' are selected, the structure is like
x-y-z, and x,z must be decimal number between 0-7, and y must be decimal number between
7. When SPC length is 14bits and 'Hex' are selected, the structure is like xyz, and x/z is a 3 bit hex
Note:
1. If protocol standard chose 'ANSI' or 'ITU-CHINA', and then the SPC length is 24 bits.
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3. SPC length performance on the OPC/DPC structure; SPC pattern instructions of the different
4. When the SPC length is 24 bits, and chosen ITU, OPC/DPC structure format is :x-y-z; x、y、z is
5. When the SPC length is 24 bits, and chosen Hex, OPC/DPC structure format is :xyz; x、y、z must
6. When the SPC length is 24 bits, and chosen ITU, OPC/DPC structure format is : x-y-z;x、z must
7. When the SPC length is 24 bits, and chosen Hex, OPC/DPC structure format is :xyz; x、z must
be three bitts hex value; y is 8 bitts hex value, such as:(202E)100 00000101 110
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CIC (circuit identification code) is an important parameter of SS7 circuit. It should be confirmed
with service provider. If the CIC is mismatched, it will result in one-way voice communication.
SS7 Circuit Add
Trunk No The “Trunk No.” here corresponds to the “Trunk No.” of SS7 trunk.
E1/T1 port No Fill in the port number of E1/T1. Assign E1/T1 to selected SS7 trunk.
Start Channel The start of SS7 channel trunk
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Start CIC No An initial circuit number to this E1/T1 matches by both parties
Count A total of 32 channels
According to the different operating modes, 7 circuit maintenance objects into two categories:
These ports can work in many ways: Select All, Invert, Clear, Block, Unblock, Reset and
Cancel.
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If user wants to manage the channel, please select operation mode to channel.
Select current port, use will see port status and protocol type. The following will show the slot
and channel status. There are 16 kinds of channel states and each state corresponds to a color
When need to share 7 signaling point, add slave TG, so as to realize the multiple TG sharing a link.
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Clock source of E1/T1can be selected “Remote” or “Local”. If selecting E1/T1 port to port0, when
user modified port0, port0-3 will be changed together with port0. Port4-7 changed following the
port4.
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these rules can be divided into 5 groups with a dial plan ID, where 0 is the default setting.
Notes:
1. In order to ensure each rule can take effect, long matching numbers (prefix) rule dial plan index
2. Maximum length is 30, this value is the number of the total length and including the prefix
length.
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special version:
1. Dial plan can be backup and restore in management configuration.
2. “Min Length” and “Max Length” are equal to the total number of possible length minus the prefix
length.
3. When overlap dialing, called number length sure, and then the “Min Length” and “Max Length” will
be set to the same value to accelerate connection rate.
4. Prefix configuration, compatible “digit map” mode.
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PSTN group configuration can be different E1/T1ports or the same port in different time slots to
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Adding PSTN group needs to fill three parameters: trunk group Numbers, trunk group Name.
Channel selection mode and at most, can add up to 16 set of data. Channel selection mode refers
to E1/T1 timeslot allocation strategy in a trunk group. There are four options: Ascending,
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IP to PSTN Calls Restriction IP to the side of the PSTN concurrent call the default without restriction. If select
Yes, and then fill in limited number of concurrent call in edit box. The max is
65535.
PSTN to IP Calls Restriction PSTN to the side of the IP concurrent call the default without restriction. If select
Yes, and then fill in limited number of concurrent call in edit box. The max is
65535.
IP to PSTN Time Restriction Default disables the function. If select enable, users will edit banning call of the
start time and end time. All call from IP to PSTN will be prohibited in this period
time.
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The user can group manage SIP/H.323 trunk through IP packet configuration.
2.10.1 IP Profile
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2.10.2 IP Group
Add the IP group including the IP group ID, IP group name, IP trunk selection. User can add a total
of 16 IP group. IP routing mod is to show in an IP group SIP time distribution strategy. There are
four options: Ascending, Descending, Cyclic ascending, Cyclic descending. (According to SIP trunk
number to choice)
This section is mainly introduce to how to configure white & black lists on the gateway.
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Caller White list: to create a batch of callers or prefixes which allow to pass the call to PSTN.
All calls coming from IP side will match this caller white list before deliver to PSTN
Caller Black list: the number or prefix which added in this list would not allow to pass the call
to PSTN. All calls coming from IP side will match this caller black list before deliver to PSTN.
Callee white list: the called number white list which allow to pass the call to PSTN. All calls
coming from IP side will match this called white list before deliver to PSTN
Callee black list: the called number black list which not allow to pass the call to PSTN. All
calls coming from IP side will match this called black list before deliver to PSTN
Caller Pool: sometimes, the caller ID from IP side are illegal which not allow to deliver to
PSTN. Caller Pool is to add a batch of number to replace the caller ID from IP side.
Example: add caller and callee white list
Add Caller white list, index 0:
Index 0 means allow the called number or called prefix 13788900 to call to PSTN
Index 1 means allow the called number with has 8 digits length to call to PSTN
Add filter profile to add both caller and callee white list works
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Example: add 100 caller nubmer to replace all caller ID from IP side
Start number 26452500, total count 100. All the caller number would be replace the number
range from 26452500 to 26452599
“Any” and "." are useful; suggesting configuration, to avoid cannot match the routing.
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IP->PSTN routing configuration and PSTN->PSTN routing configuration are similar, the only
IP->IP routing configuration and PSTN->IP configuration are similar. The only difference is that the
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Select “Number Manipulation” in navigation tree, the display interface is shown as below:
Figure 2-12-1 Number Manipulation
"Number Manipulation” is used to replace numbers. User can replace and remove the inbound
Notes:
2. Suggesting configure “Any” and “.”, avoid missing the call for the replace number。
3. When configuring data, it is suggested that index starts from large index value, to avoid adding
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PSTN->IP Callee configuration parameters and IP->PSTN Caller configuration parameters are the
same.
PSTN->PSTN Callee configuration parameters with the above is basically same, only more of a
“number type” parameter. Common number types are: Not Configured, Unknown, International,
The operation of the option the right are: Not configured, Allowed, Restricted.
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“IP->PSTN Caller”, “IP->IP Callee”, “IP->IP Caller” configuration parameters in the previous
number manipulation rules have been mentioned, please refer that section.
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DTMF Parameter
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2.15 Maintenance
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Simple Network Management Protocol (SNMP) is application layer protocol, and used to manage
communication line.
Users can manage and configure gateway on remote NM server through SNMP configuration. But
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This is a section for some customers who need more details of E1/T1 gateway with command
lines. To make sure the system runs successfully, we suggest customers setting E1/T1 gateway by
GUI. In this manual, some topics such as how to check the IP, signaling and call conversation are
covered.
Run system tool Telnet to login gateway, after entering username and password, and then run
command en to active the privileged commands.
run the command show int, the output shows FE0 and FE1 ports name, IP address and MAC
address.
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run the command show ss7 sta, the out should like this:
If the system connects with PRI, please run command show q931 sta
the system will show the status of each channel in each port.
Block entire e1
Example:
block port 2, run the command busy –cic 2, the system will disable port 2 into a locked status.
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Unblock entire e1
Example:
unblock port 2, to active the port 2, please run the command free –cic 2
Block specified ts
Example:
block ts 3 in port 2, to disable the ts 3 in port 2, run the command busy–cic 2 3
unBlock specified ts
Example:
unblock ts 3 in port 2, to enable the ts 3 in port 2, run the command free –cic 2 3
If the system connects with PRI, please run show q931 call
Customer also can run show cc call to list all the active calls with SS7/PRI (cc = call control)
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And then exit config mode, into ada mode and turnon port
(ex = exit)
Cancel debug cc, turn off the debug mode for cc all.
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a=rtpmap:107 BV32/16000
a=rtpmap:101
[074-14:18:49:710]CC: <5,Sip-t,2,65535,,idle> predispose start calling :987654321
called:1234567!
[075-14:18:49:710]CC: <5,Sip-t,2,65535,,idle> predispose end calling :987654321
called:1234567!
[076-14:18:49:710]CC: <5,Sip-t,2,65535,,idle> source user auth:0x6, is fxo call in auth pass:0
[077-14:18:49:710]CC: <5,Sip-t,2,65535,,idle> route type:3(Out route) -- before cc number
analysis.
[078-14:18:49:710]CC: <5,Sip-t,2,65535,,idle> [IP2tel]match route succ! srcIpGrpId:0,
dstTrkGrpId:0, ChnSelMode:0, callingProfId:0, srcIpGrpId:0.
(note : mach routing successful , if call failed, you can check this if source IP Group id and
destination Trunk Group Id is if the same with you expect, if not , please check the routing
configure)
(note: if configure number manipulation, can check the manipulate result here)
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t=0 0
a=ice-ufrag:2c37f5
a=ice-pwd:a0b4dc8cc787732f66e9e625e69dbbfd
m=audio 50832 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:10
[089-14:18:49:710]CC: <5,Sip-t,2,65535,,idle> ccb state change from 'idle' to 'proceeding', ccb
no:5
[090-14:18:49:710]CC: <Sip-t,2,65535>, <Ss7,65535,65535>, ====Processed: CC_ST_SETUP
[091-14:18:49:710]CCB: no:5, cr1:9, cr2:10, State:4(proceeding), SubState:0(idle), serv:0(normal),
serv_state:20(), route:3(rts_out), cause:0(CCS_NONE(无原因值))
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t=0 0
m=audio 5102 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
, priv sdp:
[114-14:18:49:770]CC: <5,Sip-t,2,65535,,wait ack> ccb state change from 'wait ack' to 'alerting',
ccb no:5
[115-14:18:49:770]CC: <5,Sip-t,2,65535,,alerting> route type:3(rts_out), called term type:8(Ss7)
[116-14:18:49:770]CC: <5,Ss7,65535,65535,,alerting> ==>> CC_ST_ALERTING, std sdp:v=0
o=call 10000 20000 IN IP4 172.16.51.10
s=-
c=IN IP4 172.16.51.10
t=0 0
m=audio 5102 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
, priv sdp:
[117-14:18:49:770]CC: <Sip-t,2,65535>, <Ss7,65535,65535>, ====Processed: CC_ST_ALERTING
[118-14:18:49:770]CCB: no:5, cr1:9, cr2:10, State:6(alerting), SubState:0(idle), serv:0(normal),
serv_state:20(), route:3(rts_out), cause:0(CCS_NONE(无原因值))
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a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
, priv sdp:
[121-14:18:49:770]ST: <Sip-t,2,65535,987654321> ====Processed: CC_ST_ALERTING
[122-14:18:49:770]ST: cr, no:9, ccb:5, State:5(deliver), cause:0(CCS_NONE(无原因值)), redirect:0
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s=-
c=IN IP4 172.16.51.15
t=0 0
m=audio 5120 RTP/AVP 4 18 8 0 101
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
, priv sdp:a=X-ACrypt
a=X-Tcp
a=X-P2PV2
a=X-P2PDst:67241984.104333337
[036-00:14:01:640]ST: <Ss7,2,65535,00000000> ====Processed: CC_SETUP_IND
[037-00:14:01:640]ST: cr, no:21, ccb:11, State:1(init), cause:0(CCS_NONE(无原因值)), redirect:0
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(note : mach routing successful , if call failed, you can check this if source IP Group id and
destination Trunk Group Id is if the same with you expect, if not , please check the routing
configure)
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