MYVOIPAPP products blog – Page 4 – Software based SIP-PBX

A small thing: UPnP

2020/05/02 Gilson

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.

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.

miniSIPServer on Ubuntu 20.04

2020/04/24 Gilson

Ubuntu 20.04 is the latest LTS (Long Term Support) version, so it is very important for miniSIPServer to support this platform.

Unfortunately, Ubuntu 20.04 has cut Qt4 and all its libraries, so we have to upgrade miniSIPServer to V37 (build 20200424) with Qt5. Now it is perfect to run miniSIPServer on Ubuntu 20.04. Please refer to following figure.

By the way, miniSIPServer for Windows will stay with Qt4 since we have to support several old Windows systems, such as XP and 7, and so on. In the future, as planed, V38 will say goodbye to Qt4 finally on all platforms.

Personalized voice in VM service

2020/02/19 Gilson

In normal, cloud miniSIPServer almost has the same services with local miniSIPServer. But for some limitations, there are some different features between them. For example, voice mail service is different.

With local miniSIPServer, each local user or extension can prompt their own audio in voice mail procedure. But with cloud miniSIPServer, each local user can only prompt the unified audio associated with their virtual PBX server. Of course, the default unified audio can be replaced with customer’s own audio file.

Now cloud miniSIPServer is upgraded. In a virtual server, each local user can has its own audio file now. Please refer to following figure. You can see a new item “Personalized voice ID” which can be different for different user.

Audio ID configuration in local user's configuration — Personalized audio configuration

Of course, the audio file cannot be uploaded to virtual server by customers themselves. If you want to upload audio files, please send them to our support team, we will upload them to your virtual server manually.

Once the audio files are uploaded, you can manage them by yourself. Please refer to following figure.

Audio files and ID management — Audio files management

In another way, you need follow some rules to create your own audio files, such as the file format and the audio codec, and so on. Please refer to online document for more details.

miniSIPServer on Deepin 15.11

2020/01/13 Gilson

Deepin is a very popular Linux distributor system in China market. It is very beautiful and easy to use. More and more software have been migrated to this system in China. As we know it is based on Debian system, we think it should be no problem to run miniSIPServer on it directly.

And it is true! Follow the online document, we can install and run miniSIPServer as same as what we do in Debian system.

Yes, this system is very beautiful. After install miniSIPServer, you can find it in its software market.

And it is very easy to run miniSIPServer.

Please enjoy it!

Monitor events in IVR call flow

2019/12/02 Gilson

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.

IVR-XML example Download

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

miniSIPServer on Debian 10

2019/07/09 Gilson

Debian 10 (Buster) is released. It is a stable and important version and can be deployed in business environment, so we must pay enough attention to this version.

We make some test with miniSIPServer on Debian 10. Now we are proud to announce that it is perfect to run miniSIPServer on the latest Debian system. Please refer to following figure.

You can update Debian source list, then download and install miniSIPServer. No more action!

Debian organization, Congratulation!

T-MSS and L-MSS

2019/06/07 Gilson

Some customers deploy several miniSIPServer nodes to build their unified VoIP network around the world.

We give a simple document to describe this network and its configuration. There are two important concepts: T-MSS and L-MSS.

L-MSS means “Local miniSIPServer” which are deployed in local branches or offices to service their local SIP phones or gateways.

T-MSS means “Trunk miniSIPServer” which is used to bridge all L-MSS servers.

Please click here to get more details.