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Category: Releases

Product release messages, including LTS, stable, development versions and patches.

call-back service is updated

call-back service is updated

By default, miniSIPServer opens UDP port 5080 to receive call-back message to invoke calls. If miniSIPServer is deployed in public network, it is possible to receive lots of other UDP packages. Of course, we can configure “application server address” for IP address authorization, but unfortunately its default value is blank and it could be dangerous.

So we update call-back service to protect the system and refine its service logic. Please refer to the configuration window firstly.

Call back service configuration
Configuration

(1) The default value of “Application server address” is changed to a local loop address “127.0.0.1”, so outsides UDP packages will be disabled. Of course, we can still keep it to be blank to accept all packages from any address, but we suggest not to do that.

(2) Local listen port can be zero which is used to close UDP socket. If the port is zero, miniSIPServer will disable the whole call-back service since it is impossible to receive any outsides UDP packages now. If you don’t use call-back service, it is better to set it to be zero.

(3) Cancel “external line mode” item. Some cutomers always ask us what it is and always confused with this item. It is just to add ‘out group prefix” automatically before the called numbers in two sessions. In fact, it is not flexible if we want to call local users and outsides users at the same time. So we discard it, if we want to call outsides users, we can add ‘out group prefix’ manually in the REQUEST messages. That means application server should be responsible for the numbers format and dial plan result.

Please refer to call-back service document for more details.

Refined ringing group

Refined ringing group

In normal, we configure ringing group information in local users’ profile. And one user can only be assigned to one ringing group. It works in most scenarios.

As you know, it is hard time now. Some companies reduce their human resources to save cost, so someone has to take more works. For example, it is possible that someone could be assigned to several ringing groups at the same time. In fact, some customers have requested us to fit this requirement.

We understand that completely, so miniSIPServer is upgraded to have a new method to provide ringing group service.

Two independent tables are added. One is used to define ringing groups and their users. Please refer to below figure.

ringing group users

Another table is used to detect ringing groups according to called numbers in different calls. Please refer to below figure.

ringing group detection

Service document has been updated. Please click here to get more details about this new feature.

Customized resource files

Customized resource files

Sometimes we need to use customized resource files, such as audio files, special IVR flows, certificate files and so on. With previous versions, all these files are stored in miniSIPServer install directory or its sub directories.

It could be a management problem when trying to uninstall or upgrade miniSIPServer. We have to be very careful to backup our own resource files.

V38 (build 20210108) is released to refine this. Customized resource files are separated to be stored in independent directories which are in application data directory.

For example, customized audio files will be stored in ‘custAnn’ directory. Once miniSIPServer is uninstalled or upgraded, these audio files will be kept without any affection.

Please refer to online document for more details.

A small thing: UPnP

A small thing: UPnP

When deploying a VoIP network, we often have one-way or no-way audio problems. It is caused by private network, for example, some SIP phones or miniSIPServer are behind routers and other SIP devices are in another different network which could be a private network or public network.

To resolve such problem, we often suggest to configure “forwarding ports” in routers manually. If you are familiar with routers, it is easy to do that.

But someone might not know how to do that, or someone might make mistake in router’s configuration. So we add a new feature in miniSIPServer to help that.

It is UPnP (Universal Plug and Play). UPnP can help miniSIPServer to map necessary ports automatically.

Firstly, you need confirm that your router can support UPnP and it has been enabled.

Then, you can click menu “Data – System” in miniSIPServer and enable the item “Enable UPnP to ask router to map ports”. Please refer to following figure.

UPnP configuration in miniSIPServer
UPnP configuration in miniSIPServer

By default, miniSIPServer will map SIP (over UDP) port and audio ports for relaying audio streams.

In another way, there is a limitation in routers. Most routers limit the number of UPnP ports, for example less than 30 ports. So if you are deploying a miniSIPServer for 50 clients or more, you will still have to configure “forwarding ports” manually.

Monitor events in IVR call flow

Monitor events in IVR call flow

In miniSIPServer, we can use IVR-XML script to enable our own services, such as automatic-attendant. With previous IVR-XML set, ‘callto’ action will invoke a call to destination and finish the whole IVR process.

But if we want to monitor some events in the call flow, such as we want to check ‘busy’ event and change the IVR flow to a new action, what should we do?

Now V37 is released and a key feature is updated in IVR-XML. We can use ‘monitor-events’ in ‘callto’ action to monitor some events and change the call flow if they are caused.

For example, the ‘callto’ action can be configured as below.

<action method="callto" name="mainAction">
    <destination>100<destination>
    <monitor-events>
        <monitor-event detection="busy" nextaction="callto101"/>
    </monitor-events>
</action>

In this example, if the call invoked by ‘callto’ action is busy, IVR procedure will be changed to next action ‘callto101’.

Please refer to IVR-XML document for more details about “monitor-events” element.

Above zip file is an example of new ‘callto’ action. You can save and unzip it into ‘xml’ sub-directory where miniSIPServer is installed and configure a new record to test it.

Configure miniSIPServer to trigger IVR-XML
Configure miniSIPServer to trigger IVR-XML
Refine “SIP over TLS”

Refine “SIP over TLS”

Some customers report a crash problem to us. All of them deploy “SIP over TLS” in their VoIP networks. We have upgraded miniSIPServer to latest V35 (build 20190313) with following key modifications.

(1) In the latest miniSIPServer, SSL library has been upgraded to the latest version.

(2) Only TLSv1.2 method is kept, that means SSLv2, SSLv3, TLSv1 and TLSv1.1 are cut. When we did research on customers’ problems, we found some bad guys were trying to use the bug of SSLv3 to hack into MSS. We have to move all these methods out to defend that. In future, we will add other methods, such as TLSv1.3. At this time, we need confirm SIP phones can support TLSv1.2 too if we want to deploy SIP over TLS.

In another way, we refine “SIP over TLS” document to provide a simple demo on how to create certificate files.

Public address of SIP server

Public address of SIP server

With the latest miniSIPServer version, it can be configured with several addresses, such as “local address” and “public address”. Please refer to following figure.

local address and public address

In normal, if miniSIPServer is deployed in public network with public address, it is unnecessary to configure independent “Public address” in above SIP configuration since the local address is public address itself. But in some network environment, for example, miniSIPServer is in a NAT and need serve outside users, we need configure “public address”. Outside users can use this “public address” to work with miniSIPServer.

If the network bridges several sub-networks, such as private network and public network, several different private networks, and so on, some SIP phones MUST use miniSIPServer private address, others use miniSIPServer public address, here we suggest to enable “relay media” in local users’ configuration which will make miniSIPServer to relay audio streams between these networks.

miniSIPServer on Ubuntu 18.04

miniSIPServer on Ubuntu 18.04

It is perfect to run miniSIPServer on Ubuntu 18.04 which is the latest LTS version.

We have tested some scenarios with miniSIPServer on Ubuntu 18.04, everything is OK. We strongly suggest you to upgrade your Ubuntu system to this version.

miniSIPServer on Ubuntu 18.04
miniSIPServer on Ubuntu 18.04

New web

New web

We have updated our website with the latest bootstrap v4. Please take a look at it.

https://www.myvoipapp.com/

The most important is that you can visit it by using PC, mobile and tablet.  Hope you can enjoy it!

In another way, we are tied to remove spam posts in forums, so we decide to say good bye to it. If you have any suggests or problems, please feel free to contact us directly.

https://www.myvoipapp.com/contact.html

V32 (stable) is pre-released

V32 (stable) is pre-released

V32 (stable) has been tested in most scenarios and we are proud to release it today!

As you can see, this V32 is in stable branch. When we finish all test scenarios and get enough response messages from customers, it will be upgraded to LTS branch and the new LTS will be released finally in the begin of next year.

Please download it from our website directly.

https://www.myvoipapp.com/download/

Hope you can enjoy it!