Support Center

1.Please telnet of MTG use putty.exe or other telnet software

For example in putty, should choose option “Telnet” and input the MTG IP address, keep the default port23.

2.Input following command, then test call till the issue reproduces, and then send the log to me.

ROS>en

ROS#sh version

ROS#^config

ROS(config)# deb SIP msg all   

ROS(config)#deb q931 detail

ROS(config)#deb cc detail all

ROS(config)#ex

ROS#^ada

ROS(ada)#[079-19:27:46:370]ADA CONNECTED ...,WELCOME!

ROS(ada)#turnon 71

ROS(ada)#turnon 27     

ROS(ada)#turnon 64                                   

1.Please telnet of MTG use putty.exe or other telnet software

For example in putty, should choose option “Telnet” and input the MTG IP address, keep the default port 23.

2.Input the following command, then test call till issue reproduces, and then send the log to me

ROS>en

ROS#sh version 

ROS#sh ss7 link                                 //will get the index of column LinkId, current is 0.

  linkId hdlcNo   mode    revErrs cc       rc    lsc             iac       poc         txc   aerm    suerm daedt daedr qlen ST   UseCnt

-----------------------------------------------------------------------------------------------------------------------------------------

  0      0      Default   0       free     open  iniLocate       notLocate free        open  free    free  open  open  0    butt 0

ROS#^config

ROS(config) #deb sip msg all                   //for  SIP trace

ROS(config )#deb cc detail all                 // for CC(MTG internal Control) trace     

ROS(config )#deb SS7 X  5                        // fro SS7 Trace, the X is the index value we got above. Current should: deb ss7 0 5  

ROS#^ada

ROS(ada)#[079-19:27:46:370]ADA CONNECTED ...,WELCOME!

ROS(ada)#turnon 71                               //for print the SIP log

ROS(ada)#turnon 27                              // for print the CC log

ROS(ada)#turnon 96                            // for print the SS7 log

1.Please telnet of MTG use putty.exe or other telnet software

For example in putty, should choose option “Telnet” and input the MTG IP address, keep the default port 23.

2.Input the following command, then test call till issue reproduces, and then send the log to me

ROS>en

ROS#^config

ROS(config)# deb SIP msg all   

ROS(config)#deb cc detail all

ROS(config)#deb r2 0 0 5    // format: deb r2 0 X 5,  X means the E1 port, current is port 0

ROS(config)#ex

ROS#^ada

ROS(ada)#[079-19:27:46:370]ADA CONNECTED ...,WELCOME!

ROS(ada)#turnon 71

ROS(ada)#turnon 27     

ROS(ada)#turnon 111

ROS(ada)#turnon 28  

ROS(ada)#turnon 48                           

Introduction

This guide is intended to show how to set the Caller ID for IP-to-E1 calls. In this guide, MTG200 is taken as an example to show the configurations, and the configurations will also apply to Dinstar’s other TG products. There are 2 modes to set Caller ID, named“Bind Port” and “Bind Route”. Users can choose one of them according to the actual application scenario. In the following, we will introduce how to use these two modes to set Caller ID respectively.

Note: 

1.MTG1000 does not have this function.

2.If the current firmware version of the device does not have this function, please upgrade it.

3.The two modes can not be applied to each E1 port at same time.

[Target]

During IP-to-E1 calls, use a Caller ID that customer want to replace the current one.

[Descriptions]

MTG200 port0 is connected with an E1 cable and has a Caller ID range of 22900600 – 22900615 and 22900617-22900699. When calls go from IP to PSTN, use any caller ID in the range to replace the original Caller ID.

Operation

1.Bind Port Mode

The “Bind Port” Mode is based on the E1 port. As long as calls go through this E1 port, all Caller IDs will be replaced.

1.1 Create the Caller Pool

Click Number Filter > Caller Pool.Select 0 as the Caller Pool ID.

The Starting Caller Number is configured as 22900600 and Number Count is set as 16, which means the Caller ID range is 22900600-22900615. MTG will choose a caller number in the range randomly to replace the original one.

1.2 Add the Port Number

Click PSTN Group Config > Port Number.

Add a rule for port number. That is, set Binding pool as “0” (which means bind the caller pool “0” to the E1 Port 0). And set Type of Outgoing Caller as “Replace”, which means the caller ID of IP-to-PSTN calls will be replaced. 

Note: The priority of Binding Number is higher than that of Binding Pool. That is to say, if Binding Number is set as a specified number, all caller IDs of IP-to-PSTN calls will be replaced by this binding number. In this case, there is no need to set binding pool.

For example, if you set 22900600 in the Binding Number field, all CID (caller IDs) will be replaced by 22900600 as follows:

2.Bind Route Mode

“Bind Route” Mode is based on the route rule in IP->PSTN, You can set a condition to determine which call’s CID will be replaced and which call’s CID will not.

2.1 Create the Caller Pool

Click Number Filter ->Caller Pool.Select “0” as Caller Pool ID.

The Starting Caller Number is configured as 22900600 and Number Count is set as 16, which means the Caller ID range is 22900600-22900615. MTG will choose a caller number in the range randomly to replace the original one.

2.2 Create the White List

Click Number Filter -> Caller White List.

Select 0 as Caller White List ID, and set Caller Number as “.”. The dot mark means all call numbers. 

2.3 Create the Filter Profile

Click Number Filter -> Filter Profile.

Create filter profiler 0, set Caller White List ID as 0 ( this white list has been configured in step 2.2), and set Caller Pool for White List  as 0 ( this caller pool has been configured in step 2.1).

2.4 Create the IP->PSTN route

Click Call Routing  ->  IP->PSTN Routing

Create routing rule for IP->PSTN calls. Choose 0 for Filter Profile ID (this filter profile has been configured in step 2.3).  You can also set Caller/Callee Prefix as conditions to filter out caller number.  The dot ‘.’ means match all call numbers.

Backgrouds

In some cases, an SS7 trunk may not have real Signal Link and therefore it needs to share Single Link from other SS7 Trunk. This instruction guide is intended to show how to use SS7 link set. In this guide, MTG2000 is taken as an example to show the configurations, and the configurations will also apply to Dinstar’s other TG products.

Note: MTG200/MTG600/MTG1000 do not have this function.

[Target] 

Both E1 trunks can be activated successfully.

Preparations

MTG Port 0 is connected with one normal SS7 trunk, the SL (Signal link) is in TS (timeslot) 16, OPC/DPC of port 0 is 9311/9315, and the CIC is 1-31, starting from TS1. 

MTG Port 1 is connected with another SS7 trunk. As port 1 does not have real independent SL, it needs to use the SL  (Signal link) of port 0. The OPC/DPC of port 1 is 9299/9294, and the CIC is 1-31, starting from TS1.

Operations

[Normal SS7 configuration]

1.Add the SS7 Trunk

Click SS7 --> SS7 Trunk, and add a normal SS7 trunk SS7_Normal for port 0 on the page.  

The OPC/DPC is 9311/9315 (which is in the format of DEC), and it needs to switch to the format of HEX first during configuration. 

You can use Calculator to switch the format.

DEC 9311   ->   HEX 245F

DEC 9315   ->   HEX 2463

If your SS7 provider supports the format of 14bit (3-8-3), you can set SS7 trunk directly as follows.

2.Add the SS7 MTP Link

Click SS7 --> SS7 MTP Link, and add a normal SS7 MTP Link No.0 for SS7_Normal on the page.

3. Add the SS7 Link Set

Click SS7 -->SS7 Link Set, and add one SS7 Link Set No.0, which is included in MTP Link No.0. 

4. Add the CIC

Click SS7--> SS7 Circuit, and add CIC information.

[Configuration without real Link SS7]

1 .Add the SS7 Trunk

Click SS7 --> SS7 Trunk, and add SS7 trunk SS7_Logic for port1, the Link Set No. is configured as Link Set No.0 (which has been set above steps). 

Since the SS7 trunk of port 1 has no real SL and it only has logical SL, it needs to share other port’s SL through link set. 

The OPC/DPC is 9299/9294 (in the format of Dec), and it needs to switch to the format of Hex. 

Dec 9299   ->   Hex 2453

Dec 9294   ->   Hex 244E

2 .Add the SS7 MTP Link

Click SS7 --> SS7 MTP Link, and add the SS7 MTP link No.1 for port.  Link Mode is set as Logical.

3. Add the CIC

Click SS7 --> SS7 Circuit, and add CIC information for both SS7 trunks.

Backgrouds

FreePBX is one of the world’s most popular open source IP PBX.It is an interface management system based on the open source Asterisk, which has become the most used open source IPPBX in the world through nearly 10 years of effort. Many of the world's leading open source IPPBXs are based on the FreePBX platform for secondary development, including the famous Elastix,Issabel, AsteriskNOW and Asterisk in Flash.

By default, the Freepbx is pure IP based software phone system, it has no any interface to connect with E1/T1. If it wants to support E1/T1 interface, it has to connect with extrenal gateways. Nowdays, most of users use Dinstar media gateways to connect with FreePbx. Dinstar media gateway is the first low cost choice for Freepbx to work with PSTN network. In this guideline, we will show you how to use Dinstar media gateway to connect with freepbx and make incoming and outgoing calls through Dinstar media gateways. Here we take the Dinstar media gateway200 as an example. Other models also apply the guideline .

Note: the chan_sip channel is used with FreePBX-13 and 14 versions, the  later version of SIP would use PJSIP. Therefore, the setup of SIP trunk would be quite different from the old version of chan_sip. 

Operations

[Configure FreePBX]

1.Add a SIP trunk

Click Connectivity>Trunks, Add Trunk to create a new sip trunk for the MTG200

Edit the parameters of the SIP trunk.Set a trunk name and other parameters, please keep the default.

Outgoing Settings: Define the outgoing settings, trunk name and peer details. 

Trunk Name: dinstar_out 

Peer Details: 

Bash

host=192.168.10.251 

type=friend 

fromuser=dinstar 

username=dinstar 

secret=qazwsx$# 

qualify=yes 

port=5060 

context=from-pstn

If the FeeePBX server and the MTG200 device are on different networks, please make host to be 

dynamic. 

Port 5060 is used to define the communication port between MTG200 and FreePBX server, 

user can define other ports too, but it must be same on both sides.

Incoming settings: Define the incoming settings, user context and user details.

User Context: dinstar_in

User Details: 

disallow=all 

allow=alaw&ulaw 

canreinvite=no 

context=from-pstn 

dtmfmode=rfc2833 

username=dinstar 

secret=qazwsx$# 

qualify=yes 

insecure=invite 

host=dynamic 

type=friend

After finishing the settings, click Submit Changes to save the configurations and click Apply to 

make them effective. 

2.Create an outbound route

Click Connectivity>Outbound Routes, click Add Route to add a new route, and configure 

the settings. 

Route Name: Local_Bucaramange, for example 

Dial Patterns that will use this Route: 6XXXXXX, this means all the dial pattern begins with 6 

and its length is equal to 7, which matches this outbound route to call out.

Trunk Sequence for Matched Routes: 

Select the trunk for this route to send out the calls. Users can set several ones, here we only use the one Dinstar which we defined for the MTG200.

After finishing these, click submit changes to save the settings and apply the configurations.

3.Create an inbound route

Click Connectivity-->Inbound Routes , click Add Incoming Route to add a new inbound  route.

Enter the name of this inbound route in Description option, Telefonica, for example. And 

configure the DID Number, make sure the DID number is same with the E1 line’s number.

Set the destination for this inbound route . It can be  an IVR or an extension or else.

Other parameters, please keep them as default. Submit the settings and apply. 

[Configure MTG200]

Login the MTG200 by entering its IP address in web browser, the default IP of 

MTG200’s FE1 port is 192.168.11.1, and default ip of MTG200’s FE0 port is 192.168.1.111, user 

can connect to either network port and use the corresponding IP to access the web of the device.

1.Set an R2 trunk 

Connect the E1 line then you get from the provider to MTG200’s E1 port, and access the web page.

Click R2 Config>R2 Trunk page, add a R2 trunk.

2.Check R2 status

Click System&Statistics>E1/T1 Status , check the R2 trunk’s status, make sure 

the E1 cable is connected and the protocol you setting is right.

3.Configure a sip trunk

Click SIP Config>SIP Trunk page, add the sip trunk on the MTG200

Trunk Name: Freepbx 

Remote Address: 192.168.10.252, the IP address should be the freepbx server’s. 

Register to Remote: Yes 

Outgoing Call Mode: Access 

SIP Username: dinstar_in

SIP password: qazwsx$# 

SIP Confirm Password: same with the sip password 

Detect Trunk Status: No.

Note: SIP username and password must be same as the one user created on FreePBX server. 

Outgoing call mode can be either access mode or peer mode.

4.Check trunk status

After finishing the settings,Click Status&Statistics>IP trunk Status to check if the trunk has 

been setup well.

1.  Add the caller white list

On page Number Filter / Caller white list, add the caller white list.

Tips

(1) on page number filter / Callee white list, you can add the callee white list.

(2) "." Represents any.

(3) caller white list is used to set the numbers allowed to call, if any caller is allowed to call, there is no need to set it.

2. Add the callee black list

On page Number Filter/Callee Black List, add the callee black list.

Tips

(1) On page Number Filter/Caller Black List, you can add the callee black list.

(2) "." Represents any, "00." is any number representing the "00" prefix. This rule is used to prohibit outgoing calls to international numbers with a 00 prefix.

(3) The black list number can be filled with the full number, such as 10086 here.

3．Set the desired filter item

On page Number Filter/Filter Profile, Set the desired filter item.

Tips：For unwanted lists, choose the default 255<None>.

4．Select the desired Filter Profile

On page Call Routing , In Routing, Filter Profile ID, select the desired Filter Profile.

Tips：If you don't need a Filter profile, use the default 255<None>.

IP -- > Call on the PSTN side

If the IP address does not hear any ring back tone, check whether there is a ring back signal and ring back tone

1. There is a ring on the PSTN, but no ring back tone

(1). If the ring back signal is detected, that is, the ring back tone is not played, that is, no sound is heard when the IP side answers the packet

The following configuration can be modified on the device, and after changing to local, MTG will send a compensation signal to the IP side

(2) After modifying the configuration, add a new ip group. After adding the ip group, you need to enter the group management, add the ip trunk to the group, and select the corresponding trunk and corresponding ip rules in the group management

(3) After modifying the ip packet, apply it to the route

2. No phone rings or ring back tone is played on the PSTN. 

In this case, the PSTN may directly return 200ok, but there is no ring back tone. The handling procedure in this case is the same as that in the case where there is a ringing signal but no ring back tone, change the source of the ring back tone for incoming IP calls to the local.

PSTN -- > Calls on the IP side

If no ring back tone is heard on the PSTN, check whether there is an 18X ring back tone

3. There is 18X on the IP side but no ring back tone

In this case, you need to change the source of the PSTN incoming ring back tone to a local one in the IP rules as follows:

(1) To create a new ip rule, ensure that the PSTN's incoming ring back tone is local

(2）Create a new ip group for managing the sip trunk

(3) Add the corresponding sip trunk to the IP group and add the corresponding IP rules

(4) Apply the ip packet to the corresponding route

4. There is neither 18X nor ring back tone on the IP side. 

For some strict carriers, if there is no 18X message, the call cannot be connected. In this case, you need to enable ringing compensation on the device. The configuration is as follows

(1) Enable ringing compensation. For a pri line, see the following figure

If it is SS7, it is enabled in SS7 parameter 

Upgrade Steps

1.Backup the data

Click Management>Data Download>Backup to download Database file to your computer to download the current config files.

2. Upload the packages

Click>Firmware Upload>Applications Upload and upload the package obtained from the Dinstar support team.

Wait for a while and it prompts for a successful upload!

Then click return .

3. Synchronize the packages

Click>Firmware Upload>Standby Software Transmit.

Wait for about one minute, and then click Firmware Upload to refresh the interface, 

If you see the word "...Synchronization Successful", it means the synchronization of the software package of the backup board is completed.

4. Restart the dual board 

Click Management>Device restart >Both RST

5. Check the upgrading results

After the device boots up, check if the current version of the master and backup boards is a new version.

Rollback to the old version steps

If you need to rollback the version, you can load the previous version directly to rollback, it is estimated that the whole rollback operation will take 15 minutes, the following are the detailed steps of rollback

1. Upload the old package

Click>Firmware Upload>Applications Upload and upload the old package

Wait for a while and it prompts for a successful upload!

Then click return .

2. Synchronize the packages

Click>Firmware Upload>Standby Software Transmit.

Wait for about one minute, and then click Firmware Upload to refresh the interface, 

If you see the word "...Synchronization Successful", it means the synchronization of the software package of the backup board is completed.

3. Restore the database

Restore the database which did before 

4. Restart the dual board 

Click Management>Device restart >Both RST

5. Confirm the rollback result

After the device starts up, check whether the current version of the master and backup boards is the previous version, and confirm whether the rollback is successful.

Backgrounds

When device meets voice issues,please go to do PCM capture on the device. And analyse it to verify that voice information is processed at the software layer.

Operations 

1.Obtain a PCM package

Log into device GUI.Clicking Diagnostic>Network capture,tick PCM as type and choose a testing port.And then click start.

1.It is recommended to use Chrome browser, the old IE browser does not have a download component and cannot download the data package.

2.UC2000____ Diagnostic>Network capture

MTG____Maintenance>Network capture

DAG____Tools>Network capture

2.UC2000____ Diagnostic>Network capture

MTG____Maintenance>Network capture

DAG____Tools>Network capture

Call testing to reproduce voice problems.Then hang up later.Last click stop.Now the PCM file will be obtained.

2.Analyse the file via Beethoven_1.73.exe

Open the file in Beethoven。

Click open ,select the file type as "All files" and choose the PCM package.

Click STAT and check at the bottom that the parsed COMplete lndCount is the same as the Diagnostic packets count.The analysis is done .

Select TDM Streams, you can see TDM Rx Capture (received voice data), TDM Tx Capture (outgoing voice data), select one of them,click Save Voice Data

The file is automatically saved to the same folder as the PCM capture; file name reference: capture-pcm_009_RX.pcm

3View audio data via Cool Edit

Choose capture-pcm_009_RX.pcm file,and open by Cool Edit.

Set Sample Rate as 8000, Chanel as Mono, Resolution as 8 bit.

Note:Cool Edit Pro opens PCM files in 8000Hz/8bit/mono, 8bit A law format.

 If the device is UC2000 H8 platform, please set 16 bit !!!

If the device is UC100/UC120/UC200 set 16 bit ,and choose 16-bit+Intel PCM(LSB,MSB)

The Raw Data (no header) choose 8-bit-A-LawCompressed

Then users can see the voice data in Cool Edit.It will be played and heard directly.