Support Center

Backgrounds

Issabel is Asterisk based open-source Unified Communications PBX software, providing user-friendly web-based configuration, management, and reporting for telephony. Modules providing predictive dialing can be added. It is a fork of the open-source versions of Elastix.  

By default, the Issable is pure IP based software phone system, it has no any interface to connect with PSTN. If it wants to support PSTN interface, it has to install analog voice cards with a physical server or connect with extrenal SIP-PSTN gwteways. Nowdays, most of users use gateways to connect with Issabel. Dinstar analog gaeway is the first option to let Issabel work with PSTN. In this guideline, we will show you how to use Dinstar Anaylog gateway to connect with Issabel and make incoming and outgoing calls through Dinstar analog gateways. 

Preparations

One Issabel communication server           (IP:172.27.0.128,Port:5060)

One Dinstar Analog gateway                     (IP:172.27.0.100,port :5060)

One extension                                             (Softphone: 100 )

Operation

[Configure Issabel]

1.Create an extension 

Click PBX >PBX configuration>Extensions.Enter the extension account information like the picture shows below.

2.Create a trunk

Click PBX >PBX configuration>Trunks and Add SIP trunk.Enter the trunk information like the picture shows below.Enter the host,username,secret,type,quality and so on . After these, please click submit.

3.Create Inbound routes

Click PBX >PBX configuration>Inbound routes.Enter the route information like the picture shows below.

Note: Choose the correct destination

4.Create Outbound routes

Click PBX >PBX configuration>Outbound routes.Enter the route information like the picture shows below.

Note: Choose the correct Trunk Sequence for Matched Routes.Here we set dinstar_gateway as the highest 

5.Overview

The configured steps on Issabel have finished, users can check the extension and Trunk status in Dashboard.There shows a registered extension and a registered trunk.

[Configure Analog gateway] 

1.Overview

Check the device System information in Status&Statics >System information .

2.Add a sip server

Click SIP Server,enter the server IP and server port（default：5060 ,the remote listen port）.As is shown in the picture.

3.Add a port 

Click Port and Add，then users can add account on the port.Here we use Fxo port 4 and enter the following information on the port ,As is shown in the picture.

Note:The sip username is matched to pbx peer information

If you want to make incoming calls without prompting “please dial extension number”,Please enter Offhook Auto-Dial.When make incoming calls from FXO to SIP server，the call from sip server go to the corresponded extension.

4.Create an Inbound route

Click Call&Routing> Tel>IP routing.Enter description,call from,calls to and  “any” in caller prefix and called prefix without limiting.

Note : Call from is port and calls to is sip server. 

5.Create an outbound route

Click Call&Routing> IP>Tel routing.Enter description,call from,calls to and  “any” in caller prefix and called prefix without limiting.

Note : Call from is sip server and calls to is port. 

Tests and Results

[Make an outgoing call test]

A call from extension 100 to service number 10086.Users can observe the call status in the following picture.

[Make an incoming call test]

A call from mobile phone to extension 100.Users can observe the call status in the following pictures.

1. Introduction

What Is SNMP

SNMP:  Simple Network Management Protocol. it used for network equipment management. The administrator (snmp server side ) obtains data/information from the device through snmp read operation (GET) and configure device through snmp write operation (SET). The device notifies the device of important status change events to administrator through the snmp trap(alarm).

Why Need SNMP

 The basic idea of snmp is to define a unified interface and protocol for devices of different types, manufacturers, and models, so that administrators can manage these network management devices with a unified interface, improving the efficiency of network management and simplifying the work of network administrators. snmD is designed on the TCP/P protocol family, works based on TCP/IP protocol, and manages the devices that support snmp protocol in the network 

Target

1. we can learn how to configure SNMP on IAD device through the document

2. we can learn how to set/get data of IAD and IAD trap events to snmp server through the document.

Description

In the document, we used SNMP software "MG-SOFT MIB brower" and it is an extremely flexible, technically sophIisticated, powerful and user-friendly SNMP browser,It allows you to monitor and manage any SNMP on your network using the SNMPv1, SNMPv2c and SNMPv3 protocols over UDP and TCP over IPv4 and IPv6 networks equipmen.

 The IP of SNMP server is 172.27.0.155

  The IP of test IAD device is  172.27.0.201 (DA2000-16S, version is 34831121)

2. Configure On IAD 

2.1. Enable SNMP and choose SNMP version

IAD support all three version v1/v2c/v3 of snmp,  it is as below

Please enable SNMP and choose the  version same as SNMP server. we will using the version  V2C in the sample.

2.2 Configure Community Configuration

Description:

This configuration is applicable to v1 and v2c. You need to configure the community and source address. Community is a string equivalent to a password in authentication; The source IP address is the IP address of the snmp server. The default value is default(means any ip). This configuration is for configure snmp server information. the community must be same with SNMP server .

In the sample, we configure public as community and default as source. it means we also configure public community on SNMP server.

2.3 User Configuration

Description:

This configuration is applicable to V3 version. You need to set the user name, authentication type, authentication password, encryption type, and encryption password. Add authentication information to SNMP v3 messages and encrypt the packet content. those information is provided by SNMP server.

In the sample , we do not use V3 version, so keep this the user configuration to blank.

2.4 Configure Group Configuration

Description:

This configuration is applicable to v1/v2c/v3 version. The group name is a string and we can  set any string as group name. then select the community that has been configured.  The community is added to a group. The access configuration of step 5 configure  give access authority of red/write/trap based on the group name.

In the sample,we set publicgroup as group name and select public community.

2.5 Configure View Configuration

Description:

This configuration is applicable to v1/v2c/v3 version. The ViewName is a string and we can  set any string as View Name. and you need to configure ViewType/ViewSubtree/ViewMask.

ViewType: it have included and excluded. included means included that MIB node that we set on viewsubtree and excluded means excluded that MIB node that we set on viewsubtree.

ViewSubtree: MIB node 

ViewMask:it is mask for MIB OID

In the sample, we set viewagent as viewname and set included and .1 as viewsubtree. keep blank on viewmask.

2.6 Configure Access Configuration

Description:

This configuration is applicable to v1/v2c/v3 version.you need to select group and select the view for the group to read, write, and notify (trap).

the view of read/write/notify means give authority of read,write and trap 

In the sample, we select publicgroup that we added on step 3 and all read/write/notify both select viewagent view. it means give authority of read,write and trap to the group.

2.7 Configure Trap Configuration

Description:

This configuration is applicable to v1/v2c/v3 version.

Trap type: V1/V2C/V3. it is snmp version that snmp server side used

Trap ip:  it is ip of snmp server. 

Trap port: it is snmp port of snmp server. 162 is detaul trap port of snmp procotol.

Trap community : it is community that SNMP server side used

In the sample, the ip of snmp server is 172.27.0.155 and trap type of snmp server is v2c. the trap port of snmp server is 162 and trap community of snmp server is public.

3. Configure On SNMP Server

3.1. Upload MIB File Of IAD 

We need to upload MIB file of IAD to the SNMP server. If not, the SNMP server can't recongize the datastruct of IAD, different product using differet MIB file. 

Note: There is mib file of IAD.

VOIP-IAD-SIP(2).mib

When upload successfully, the mib brower will show mib tree of IAD

3.2 Configure Snmp Server

We need configure ip of IAD and read community and set community and SNMP port . We also select SNMP version .

In the sample, the IP of IAD is 172.27.0.201 and used SNMP version V2C. the community of read/set both are public.

3.3 Contact IAD On SNMP Server

Right click and it will show menu. then click contact and the SNMP server will connect IAD.

4. Apply the  SNMP function.

4.1 Read(Get) Information From IAD

Right click the node and it will show menu. Then click "Get" and it will get the information from IAD.

For example, we do it in node "iadDevSerialNumber", it  will get the SN from device. 

Get  SN successfully, it's of device side.

4.2 Write(Set) Configuration To IAD

Right click the node and it will show menu. Then click "Set" in menu. It will Write(Set) the informatin to IAD side.

For example, we do it in node "iadSIPServerAddresss", it  will Write(Set) the IP to  IAD. 

We set the IP  172.19.211.110 to SIP server of IAD.

Settting successfully！ the IAD side show the SIP server setted.

4.3 Trap Information 

Trap mean the IAD report the status automatic. 

Check trap message 

please note: sometimes the trap port will be occupied and we can change the default port 162 to other idle ports. it is as below

when we changed the trap port on snmp server side and we also need to change the trap port on IAD device.

When device stauts changed, IAD will send trap message to SNMP server.

it is slot trap as below. 

Note: There is detail description for all IAD trap.

IADSNMPTrap

Backgrouds

DSP is responsible for the voice processing of the entire device, so when the DSP problems, are directly manifested as voice problems, and all ports and calls are the same.

(1) No dial tone on off-hook

(2) No ringing on incoming calls

(3) No ringback on outgoing calls

(4) No voice in both directions

(5) Single-pass voice (telecommunications)

(6) Calling noise, etc.

And what is the dsp frequently questions,

(1) DSP firmware issues, including missing, corrupted and mismatched firmware.

(2) DSP license issues, including license authorization failure, unauthorized, insufficient license channels, corrupted license file.

(3) DSP hardware issue

In this chapter, it will be shown how to deal with DSP problems.

Steps

1. Upgrade the device and reload the dsp file 

Upgrade the device version to xx.xx.10.05 or above, and upgrade the corresponding DSP file (Version,DSP and DMS matching is very important)

If there is still a problem with the DSP on the device after the upgrade is completed, please do the following.

2. DSP authorization

Telnet into device and then enter these commands below,

ROS>en

ROS#^config

ROS(config)#sd dsp

(Password：*dinstar#） 

After dsp authorization, there will be two situations, the first reset DSP auth data succ, reboot the device and then check the dsp state.

The second reset DSP auth data failed. if, the authorization fails or there is still a problem with the DSP after the device is rebooted, then reauthorize.

3. Load a new authorization file and reauthorize it

3.1 Obtain authorization file

Enter the following command to get the flash UID

ROS>en

ROS#^ada

ROS(ada)#cmd 50 100

and send the device version information and flash UID to Dinstar Supports for a new license file.

3.2 Load the newly acquired authorization file

(1) Disable firewalls and antivirus software

(2) Place the new license file and TFTP server in the same path of your computer.

Ready the remote to Dinstar supports reload the dsp file .

Backgrouds

The FXO device is used to achieve the connection of the network to PSTN and then greatly helps reduce the cost of long-distance calls. Sometimes, you may face some issues with voice quality. 

In this guide, we will show you how to deal with FXO voice issues.

Operation

1. IP side hear voice issues 

Voice issues heard on the IP side may come from the following three sources:

PSTN side 

FXO device

The network from FXO to IP

So we need to locate the roots of the voice issues.

1.1 Check whether the Voice issues from PSTN Side

First, check what kind of voice we have received from the PSTN side, and whether the voice is the same as that we hear on the IP side.

Connect the pstn line with an normal analog phone directly. Make calls and check the voice situation.

If there is voice issue ,we can judge the bad voice quality from pstn line itself. Please contact the operator.

If there is no voice issue ,go next.

1.2 Check whether the Voice issues from FXO Device to IP Side

Do a PCM package on device.Click ‘Network Package’ > ‘PCM’, and then choose a port that you will use.Make a call and speak from both sides for about 60 seconds.Stop the call.Then you will get the capturing files .

As we have gotten the capture-eth , we can open it and check the voice which have been processed by the FXO device.We can open the “capture-eth” file by using Wireshark and then hear the voice that we sent out. 

If the voice that has been processed by the FXO device is of good quality, it means the problem occurs on the network from FXO device to IP side.

If the voice from FXO device is of bad quality, we need to adjust some parameters of the device.

Note: 

1. Wireshark can play the voice when using Codec G711 only.

2. You can try to open through OmniPeek if  Codec uses G729.

1.3 Solve Voice issues heard on the IP side

When the voice from PSTN line is of good quality but there is voice problem out from the FXO device, we can adjust the parameters of the device.

The PSTNs from different PBX switches have different features.Please use a PSTN line to connect the FXO port. 

And then start in two ways:Impedance and Gain

[Adjust the Impedance]

Different countries require different line impedances for the telephone network.If the impedance of the FXO device is inconsistent with that of the PSTN line, the voice quality will be greatly affected.

If you know the line impedance of your country, you can check device setting, modify it and then restart device to make the new setting take effect.

The impedance setting is on the FXO Parameter >Advance.

Note: After changing Impedance, need reboot to active.

[Test the Impedance]

If you do not know the impedance of your current line, our FXO device provides impedance test.

Click Tools>“impedance test” FXO Test, 

Note: It may take about 15 minutes for the test. You had better not use the port when impedance test is ongoing.

[Parameter settings]

(1)Choose an online port

(2)Dial Test Number: you can input a real number, delete the last three digits.

(3)Dial Timeout Time: the dialing timeout is 9 seconds generally, but you can click “Dail Test” and get the value.

(4)Match Mode: use “simple mode” which may take about 15 minutes.

[Result Settings]

The process takes about 15 minutes. When it is finished, you can get the Acim and Hybrid value.Record it and click save. This will apply to the tested port directly. 

If you want to apply those values to all ports, you need to set the impedance at the following page and save it, and then reboot device to make it take effect.

Choose 8 for Acim, set 1 for Hybrid, save and then restart the device.

[Adjust the Gain]

Why does the gain effect voice quality?

The receiving frequency of the FXO device has a range. If the wave form is too large, we will cut off the peak, which will cause sound distortion.

Test a call to check the voice quality.

If the voice quality is OK, you do not need to modify the gain.

If there is still noise or echo, and the voice is louder, you can add one more Tel Profile and decrease the RX gain.Every time you can reduce 2, save it and then test the call.

If the quality becomes better, you can continue to decrease the gain.

If the quality is worse, you can try to add the gain

After configuration is finished, it should be applied to the port.

2.Tel side hear voice issues

If the PSTN side hears bad voice, the voice problem may come from IP side,FXO device,PSTN side

2.1 Check the Voice from IP Side

We need capture a complete call first, and then analyze the voice from IP side.Choose the network and PCM, and then choose the port that you will test.Make a call, speak on both sides for about 30 seconds.Stop the call.Then you get the captured file

Open the ‘capture-eth’ file by using the wireshark, and then we can hear the voice from IP side and know the voice from IP side is of good quality or not.

If  the voice we heard is of good quality, it means the FXO device received good voice from the IP side, and we need to check the voice sent to PSTN.

2.2 Check the voice sent out from device

We can open the ”capture-pcm” by using Beethoven Tool; it can separate the received and sent voice.

Click open, choose the “capture pcm” file. If you can’t see the file, choose ‘ALL Files’ for ‘file type’, and then open the file.

Click “STAT”, you can get the RX and TX in tab “TDM Steams”

Choose the TDM RX Capture, then right click, you can save the RX voice file and TX voice file .

Open the voice file by using Cool Edit Pro, choose 8000HZ,MONO,8bit,8-bit A-law, we can hear the TX voice .

If the TX voice is the same as that of the IP side, it means the voice sent from the FXO device is of good quality.

If the TX voice is poor quality, it means the voice problem comes form the device and we need to adjust the device parameter.

2.3 Solve voice issues heard on the PSTN Side

Same as dealing with the voice problems on the IP side, we need to adjust the impedance first, and then adjust the gain.You can refer to Chapter 3 about the impedance setting

If the voice is still of poor quality, we need to adjust the TX gain .

If the sound peak is too high, it needs to be cut off and we need add one more Tel Profile and decrease the TX gain.

If the sound peak is too low, we need to increase the TX gain.

After configuring the gain, it should be applied to the port.

Backgrouds

Generally , the echo issue is caused by the sound on one side being too loud and transmitted back to the other side by the radio on the other side or by the line impedance and FXO mismatch. When we encounter this kind of problem, we can judge and debug by catching PCM packets on the device and doing telnet tracing on the device.

Operation

1. Detect if there is an echo on the PSTN line

Connect the PSTN line directly to the analog phone first to ensure that the line itself is echo-free when using the analog phone.

2.Impedance matching test

Before impedance matching, please select an online port and ensure that no call work will be performed on that port. The dialing test number can be configured by yourself, but it cannot be the same as the number of the * service. After selecting a test number such as 61919966.

2.1. Click Tools> FXO test. Select Impendece Test

2.2.Select test number 

The test number does not need to be entered completely, only the first part of the number can be entered, usually 6 digits. For example, if a phone number is 61919966, the test number only needs to be set to 619199.

2.3 Analysis of test values 

When the test is done and the ACIM option can see the normal value .Please save the current port configuration. (Normal range 0-15, 255 is the default value indicating that the correct result has not been obtained)

Note:The test is for the current port, not for all ports.

If the PSTN lines used for FXO ports originate from the same carrier and multiple ports have the same ACIM, you can modify the global impedance value based on the test results. For example, if the ACIM is 11, modify all ports as follows.

Don't forget to restart the device to take effect.

If the echo problem persists after completing impedance test, proceed to the next step.

3.Adjust gain

3.1 Install software which can enter telnet. (example: putty)

3.2 Create a session on putty

 Select telnet, enter the device ip.If you changed the telnet port on device ,please enter the corresponding one.Then open it .

3.3 login device 

Enter the username and password.

3.4.Adjust the dsp parameter 

Enter the commands to modify the gain 

ROS>en

ROS#dsp

If the echo is heard on the IP side, lower the value of FXO CODEC ----> PSTN.

If the echo is heard on the PSTN side, lower the value of FXO CODEC < ----PSTN.

ROS#^config

ROS（config）#save

ROS（config）#ex

Note: Don't forget to restart the device to take effect of the dsp parameter .

4.Call test

Dialing out through gateway, start talking if echo still exists. Please repeat step 3.

Descriptions

SIP registration and SIP trunk are the connections between two SIP-based entities, and are used to communicate over the internet in VOIP services. 

Registration mode is typically used when one has a private IP address and the other is a public IP address. In SIP registration, the FXO gateway acts as a registration client, and the softswitch acts as a registrar.

Using SIP trunk mode without registering, it is required to know the peer IP address and SIP listen port, and these IP addresses and ports must be reachable.Both devices would be in trusted network.  users can choose the mode.

 [Common Usage Scenarios for SIP trunk mode]

Both gateway and softswitch are at the same local network 

Gateway and softswitch connect to the same local network over VPN

NOTE: It is not recommended to use SIP trunk one is behind NAT and the other with public IP address. This usage’s behavior is unexpectable. 

Preparations

SIP server IP :112.74.215.213, Port:5060 

SIP account:111666, Password:123456

Operation

1. Registration mode

1.1 Add a sip server

Click SIP Server,enter the server  IP and server port（default：5060, the remote listen port）

The local sip port usually be set as 5060(SIP UDP /TCP  Local Port) 5061（SIP TLS Local port）

1.2 Add ports

Click Port or Call&Routing>Port group

Add a port or a port group(adding port is single port to register a account,but adding a group is many ports to register the account) we add a port group here.

If you want to make incoming calls without prompting “please dial extension number”,Please enter Offhook Auto-Dial.

Note: Offhook Auto-Dial number select the extension number or group number.When make incoming calls from FXO to SIP server，the call from sip server go to the corresponded extension or group

1.3. Check register status

Click Status&Statics to check register status.

1.4. Add routes

[Add an outbound (IP->Tel) route]

Enter description. Select call from : sip server, calls to: port &port group. Enter “any” in caller prefix and called prefix without limiting

[Add a （Inbound）Tel->IP route]

Enter description.Select call from :port &port group ,calls to :sip server. Enter “any” in caller prefix and called prefix without limiting

2. Sip trunk mode without registering 

2.1. Add a sip trunk 

Click Call&Routing>IP trunk. Enter the description,the server IP and port.

Set sip server (172.29.1.6) and port 5060 as a demo.

2.2. Add a port group

Click Calls&Routing>Port group. Enter SIP User ID,and the port select all ports.If you want to make incoming calls without prompting “please dial extension number”, Please enter Off-hook Auto-Dial

Note:Off-hook Auto-Dial number select the extension number or group number.When making incoming calls,the call form FXO to SIP server ,and then sip server to the corresponded extension or group.

2.3. Add routes

[Add a IP->Tel route]

Enter “any”in caller prefix and called prefix without limiting.

[Add a Tel->IP route]

Enter ’’any’ ’in caller prefix and called prefix without limiting.

NOTE : FXO SIP port as the picture shown below. The UDP port is 5060.when FXO connect to sip trunk.Don’t select random ports.

Conclusion

....

Backgrounds

Caller identification (Caller ID) is a telephone service, available in analog and digital telephone systems, including voice over IP (VoIP), that transmits a caller's telephone number to the called party's telephone equipment when the call is being set up.

What are the general main concerns of customers regarding FXO port analogue gateways? For this ,We did a research from our customers. 

According to the survey of our customers' top concerns about our analogue gateways, Beside voice quality issues and caller id display issues is also one of the most caring one. Obviously, caller id is very important when calling.  If the callerid missed, some main feastures will be broken, for example, call routing for IPPBX incomiong call control based on the callerid, banking service, health care service call center for VIP service based on callerid.  lucklly, Dinstar analog gateways offers smart tools to help the customer to solve the callerid issue. 

The following document describes how to solve with caller id that is not detected properly on the fxo port. There are six steps go through for basic checking. 

Tips 

More about the caller id information ,please refer to the link,https://en.wikipedia.org/wiki/Caller_ID

Operations

1. Ensure the pstn line with caller id 

Connecting the PSTN line directly to the analogue phone and doing an incoming call test check the pstn line whether have caller id. 

If the caller id is shown . It is important to observe the Caller id is displayed before ringing or after.

2. Check the device settings about caller id 

Login the device GUI.

Click Advanced>FXO parameter , Enable "Send Original CID when Call from PSTN ".

And check whether the "Detect CID" settings  is the same as the phone test results in the step 1 . ( before ring /after ring )

After confirming and adjusting , but issue still , please go to the next step.

3. Make Impendence Test

Click Tools > FXO Test > " Impedance Test ", do the impendence test

Choose a online port and enter a test number.  (The number you can set as one real phone number,without limit )

When the Acim value is shown after testing, please save it .

If you get 255 in the Acim ,it is failed .please change a phone number and do one more testing.

After the Impendence test done ,but CID showing still issued. Let's go to next step. 

4. Trace for the device 

4.1 Create a putty session

Open "Putty" . Enter the device ip ,and connection type choose "telnet ".Click Logging>All session output >save the log on the pc .And click open.

The following picture shows the steps and options, please follow them to create.

4.2 Login the telnet session

Enter username and password to login it .

4.3 Enter the commands 

Enter the following commands ,And then do step 5 .

5. Capture a pcm package from device 

After trace the device , please back to device GUI . 

Click Tools>network capture, and choose PCM and port. Then click start .

Do a incoming call for about 10 s, and then hang up the call .After these , stop the capture. 

After that , you will get a capture log .

6. Analysis the logs and packages

You can check the PCM packages wether device get the CID.If you are not clear about it. Please share the trace log and captures to Dinstar technicians .

Note : About parse pcm packages steps , please take the file as reference .

Here is the link, How to parse the PCM package from device

7. Conclusions

If your business is experiencing caller ID issues on its FXO ports, there are several steps you can take to diagnose and correct the problem. If neither of these solutions works,This is a more complex issue ,please ask the dinstar technician to diagnose and repair. 

Backgrounds

According to ITU-T,  the definition of Busy tone is defined as below:Busy Tone: A tone advising the caller that the telephone number caller dialed is busy. 

FXO is generally connected to a PSTN phone line provided by the carrier or  small switch of the companies, and the default busy tone standard is different for each country carrier.Please refer to our linked document for the specific ITU-T specification for busy tone.Usually, there are standards for the cadence ON and OFF in the busy format, for example, the Chinese standard is 500ms ON, 500ms OFF, 500ms ON, 500ms OFF. However, in practice, the subscriber gateway has been configured with the local standard, many operators still support a variety of busy cadences, such as 400ms ON, 400ms OFF, 400ms ON, 400ms OFF, 1000ms ON, 1000ms OFF. This will cause the FXO port gateway not recognizing the correct busy tone and then not releasing the line and hanging up in time.  

Dinstar analog gateways support a busy tone algorithm to learn through FXO busy tone and to improve the hangup detection. 

[issue]

One side is already hung up, but the remote side is still occupied. If users encounter such a problem, they need to open busy tone learning on FXO to match the current specific environment through the troubleshooting tool in Dinstar Analog Gateway. After intelligent learning process and matching the result,  the FXO can be successfully hooked up.

More about busy tone，please refer to the link：https://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf，

Operations

1.Detect busy tone on FXO

Click Tools> FXO test.Select Auto detected Busy Tone and one port which is online and idle.Input the test number .

1.The port must be online and idle

2.The test number : suggest users connect a PSTN line with analog phones directly to test a non- existing number but can be heard busy tone directly.Or an existing number but the call hangs up immediately after the connection to get busy tone.

2. Analysis and Application of busy tone learning results

2.1 Save results if busy tone learning successfully

"Original cadence" is FXO detected 8 busy cadence. And it will show the corresponding ON/OFF time. A complete busy tone is usually 4 cadences . To ensure that there are no false detections, we choose two busy cycles. so 8 busy cadences are displayed.And it will show the corresponded recommended cadence(ON/OFF).

As is shown in the picture,360,340,0,0,0,0,0 is a complete busy tone.The value 360,340,0,0,0,0,0 means 360ms ON, 340ms OFF, 360ms ON, 340ms OFF, 360ms ON, 340ms OFF, 360ms ON, 340ms OFF

Now, please save the result.

2.2 If busy tone learning failed

If the original cadence value are all 0 or irregular(identify the beep sound not busy tone) ,that can’t be saved.

[ Case 1]

If the test result values all are 0,mean the line may not play the busy tone that causes failed

[Suggestion 1]Change  test number 

[Suggestion 2]Connect the PSTN line to analog phone directly and dial the testing number to check if there is busy tone after called hang up 

[ Case 2 ]

If the Original Cadence and Recommended Cadence are irregular values,the line does not play the busy tone directly but the IVR of the operator.

[Suggestion 1]Change test number 

[Suggestion 2]Do a capture of pcm package and analyse the cadence from the package

[ Case 3 ]

There is busy tone but can’t be detected in real call

[Suggestion 1] Adjust the IP receive busy tone gain or FXO send gain.

[Suggestion 2] Adjust the energy threshold.Click advanced> FXO parameters, busy tone detection parameters interface, on->off energy threshold is adjusted down (-35 ~ -25), off->on energy threshold is adjusted down (-35 ~ -25) .The smaller the value of busy tone detection more sensitive.

NOTE:This setting needs to reboot to take effect.

3.Check the busy tone parameters

Click Advanced>FXO parameter.Check the busy tone detected parameters.FXO hook up condition is busy tone detection parameter enabled by default.

If the users know the cadence clearly,  please set it in the blank manually.

Here is the International specifications on busy tone ,please refer to the following links ,

https://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

https://audiojungle.net/item/china-standardized-telephone-tones/34045224https://en.wikipedia.org/wiki/Busy_signal

1.How to do the Impedance Test

Tips:

(1). The selected port has an online status; no calls can be made to the port during the test.

(2). "Dial Test Number" is a non-existent number, e.g. 123456789.

(3). After clicking save, the impedance value of the port is valid.

(4). It is recommended to perform an impedance test on each port and retest it after changing the PSTN line.

(5). For normal results refer to the chart below.

2.How to view the impedance value of each port

To access the FXO gateway through telnet, enter the following command to query.

(1). en

(2). sh fxo all

Tips:

(1). If the ACIM&HYBRID value is 255 it means that no impedance test has been done or the test result is wrong.

(2). If the ACIM&HYBRID value is 255, the impedance test will be performed with the web-configured impedance

(3). If web sets ACIM&HYBRID, save and reboot to take effect.

3. Other notes

(1). What is ACIM & Hybrid?

ACIM (Alternating Current Induced Modulation) and Hybrid are two concepts used in telephone lines.

ACIM refers to AC Induced Modulation, which is an interference suppression technique used in telephone lines. ACIM improves call quality by introducing a signal on the telephone line that is exactly the same as the interfering signal but in opposite phase to counteract the interference, which can occur when the telephone line is close to power lines or other electrical equipment.

Hybrid is another important component of the telephone line, which is used to separate the voice signals of the sender and receiver. Hybrid can effectively isolate the two signals, thus ensuring call quality and clarity.

In short, both ACIM and Hybrid are technologies and components used in telephone lines to improve call quality and reduce interference.

Overview

In today's digital age, the integration of traditional fax machines with modern VoIP systems has become a crucial challenge for businesses. One of the solutions to this challenge is the integration between 3CX, a popular VoIP communication system, and fax machines using Dinstar FXS gateway. This integration allows businesses to maintain the use of fax machines while leveraging the benefits of a modern communication system.

The Dinstar FXS gateway acts as a bridge between the 3CX IP PBX and the traditional fax machine. By converting the IP signals from 3CX into analog signals that the fax machine can understand, the Dinstar FXS gateway enables seamless communication between the two systems. 

This documentation outlines the steps for integrating 3CX, a popular IP PBX system, with a Fax machine using Dinstar FXS device. There are two different methods depending on what functionalities we need to achieve using FXS. 

Integration Methods

Method 1: Integrating 3CX with an FXS to enable the use of both fax machines and analog phones for faxes/calls

Method 2: Integration of 3CX with an FXS port specifically for connecting a fax machine

Prerequisites

3CX Phone System installed and configured.

FXS devices (analog phones, fax machines, etc.).

SIP / FXO lines added in 3CX as SIP trunk

Access to the 3CX Management Console and FXS gateway

Method 1___Integrating 3CX with an FXS to enable the use of both fax machines and analog phones for faxes/calls

Step 1: Addition of trunk in 3CX

Open 3CX Management Console. Navigate to "SIP Trunks" and click "Analog."  

Fill in the SIP Gateway details:

● Enter name for Trunk: [Trunk Name]

● Enter IP of FXS: 

● Main Trunk No: [Any random number like “00000” or “12345”]

● Authentication: IP Based

● Save and apply the changes

 Step 2: Extension creation at 3CX side

-Create an extension for fax in 3CX (Regular extension). Users -> Add -> Save

*Outbound Caller ID: This will be the DID of fax line

Step 3: Inbound rule and outbound rule

● Please note that the Fax line can either be in a SIP or FXO line from the provider. This trunk should be readily available in the 3CX SIP Trunk section. 

● SIP trunks -> Edit (SIP trunk where fax line is included) -> DID -> Add -> [FAX DID] -> Save

● Inbound Rules -> Add DID Rule -> Select Fax DID -> Select the in office and out office route as the fax extension

● Outbound Rules -> Add -> (You may make this outbound rule for fax extension to the top)

Step 4: Configuration of FXS gateway

● Login to Dinstar FXS gateway

● SIP Server -> [IP address of 3CX]

Heartbeat -> Enabled 

Local SIP Port -> Use Random Port -> Disable -> Port: 5060 & 5061 -> Save

● Then after, configure port under Port menu,

● Port -> Add -> Select Port -> (Authentication ID and Authentication Password will be from 3CX fax extension under phone provisioning)

● -> Save

● NB: “Authentication ID should be changed to number format in fax extension”. Please see the image below.

● IP -> Tel  [SIP server to FXS port]

NB: If only one port is active in FXS gateway, then both caller prefix and called prefix will be “any”

But,  If you have more than one port enabled in FXS, then you need to give fax DID in “called prefix” in the settings below.

● Tel  -> IP  [FXS port to SIP Server]

After the above steps the fax extension in 3CX will be shown green and it will be shown registered under Status & Statistics -> Port Status in Dinstar.

Now you can send and receive the fax if you connect the fax machine.

Likewise, you will be able to make and receive calls with the analog phone if connected in that FXS port. 

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Method 2: Integration of 3CX with an FXS port specifically for connecting a fax machine

Step 1: Addition of trunk in 3CX

Open 3CX Management Console. Navigate to "SIP Trunks" and click "Analog."

Fill in the SIP Gateway details:

● Enter name for Trunk: [Trunk Name]

● Enter IP of FXS: 

● Main Trunk No: [Fax DID]

● Authentication: IP Based

● Save and apply the changes

Step 2: Extension creation at 3CX side

Create an extension for fax in 3CX. Advanced -> Fax -> Add -> Save

*Default Caller ID: This will be the DID of fax line

Step 3: Inbound rule and outbound rule

● Please note that the Fax line can either be in a SIP or FXO line from the provider. This trunk should be readily available in the 3CX SIP Trunk section. 

● SIP trunks -> Edit (SIP trunk where fax line is included) -> DID -> Add -> [FAX DID] -> Save

● Inbound Rules -> Add DID Rule -> Select Fax DID -> Select the in office and out office route as the fax device

● Outbound Rules -> Add -> (You may make this outbound rule for fax extension to the top)

Step 4: Configuration of FXS gateway

● Login to Dinstar FXS gateway

● SIP Server -> [IP address of 3CX]

Heartbeat -> Enabled 

Local SIP Port -> Use Random Port -> Disable -> Port: 5060 & 5061 -> Save

● Then after, configure port under Port menu,

● Port -> Add -> Select Port -> (Authentication ID and Authentication Password will be from 3CX fax extension)

● -> Save

● NB: “Authentication ID should be changed to number format in fax extension”. Please see the image below.

● IP -> Tel  [SIP server to FXS port]

NB: If only one port is active in FXS gateway, then both caller prefix and called prefix will be “any”

But,  If you have more than one port enabled in FXS, then you need to give fax DID in “called prefix” in the settings below.

● Tel  -> IP  [FXS port to SIP Server]

After the above steps the fax extension in 3CX will be shown green and it will be shown registered under Status & Statistics -> Port Status in Dinstar.

Now you can send and receive the fax if you connect the fax machine.

—-------------------------------------------------------------------------------------------------------------------------------------

Conclusion

You have now successfully integrated 3CX with a fax machine using Dinstar FXS gateway. In summary, integrating a FAX machine into the 3CX system through the Dinstar FXS interface enhances communication efficiency, ensuring a reliable FAX solution.

This is a basic prototype, and you should adapt it to your specific hardware, network, and 3CX configurations. Additionally, refer to the comprehensive setup instructions in the documentation supplied by both Dinstar and 3CX for detailed guidance