Debian 11 (Bullseys) was released at August 14th, 2021. It is the latest stable version and very important for business deployment. So we run miniSIPServer on this system and make some tests. The result is perfect!
It is no problem to run miniSIPServer on Debian 11. You can refer to online document for more details about how to run miniSIPServer on Linux systems.
Today we add a new version (for 5 clients) into miniSIPServer lines. This version is FREE! That means you don’t need a license and don’t warry about expired problem.
This “5 clients” version is perfect for small VoIP delpoyment, such as family communication, testing and so on. You don’t need to pay a cent to get full VoIP functions. Of course, the clients are limited to be no more than 5 clients.
In another way, free versions are not available for commercial usage.
Hope you can enjoy it!
Sometimes, we want to know the details of real-time call status, such as who is calling, how many incoming calls, and so on. In fact, miniSIPServer has a real-time CDR window which can display CDR records just finished their calls. That means the real-time CDR window only has a half real-time function, and it cannot display the details of calls which are still in calling status.
Obviously, we think we need to upgrade such windows to display more details. And here we get:
In the new window, the call which has been released or disconnected is in gray. If the call is calling, it will be in black. When the called party is ringing, it will be changed to be in blue. Once the called party answers the call, it will be red immediately.
So you can get real-time status of all concurrent calls. That’s interesting.
Of course, you need to configure miniSIPServer to display real-time CDR and what kinds of CDR should be generated. Please click menu “data / system / call detail record” and get configuration below.
Most important, you need to configure “generate fail call CDR” and “show real-time CDR information” items. Please visit manual document for more details about other items.
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.
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!
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!
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.
We often configure miniSIPServer to connect VoIP carriers’ network with external lines. There are lots of VoIP carriers and someone always asks us how to configure external line.
In our step by step document, we give a demo to configure MSS to work with “call centric”. You can refer to this document for more details about VoIP networks and external lines. In another way, we give some more examples in chapter “External lines” of F.A.Q document. Please refer to these documents if you are interesting in it and hope they can be helpful to you.
https://www.myvoipapp.com/docs/faq/index.html
miniSIPServer can support customers’ own audio files to replace default files. With previous MSS, customers have to backup and restore these files once they want to upgrade MSS.
This is a little trouble. With the new V32, we can resolve it now.
When MSS starts up, it will create a sub directory ‘cust_ann’ in ‘mss_ann’ directory, now all your own audio files can be stored in this directory. When MSS is uninstalled or upgraded, this directory and its files will not be deleted or replaced by default files, and MSS can get audio files from this directory directly when it starts up.
In windows system, it could be “d:/myvoipapp/minisipserver/mss_ann/cust_ann” directory by default. In Linux system, it could be “/opt/sipserver/mss_ann/cust_ann/”.
Please refer to our online document for more details about how to record own audio files.
https://www.myvoipapp.com/docs/others/how_to_record_your_own_audio/index.html
Maintenance is required for all virtual servers in our cloud system. We will reboot all our servers in 2018-01-19 7:00:00 AM UTC. You can prepare your VoIP networks for this mantenance.
This action affects the underlying infrastructure that your virtual server resides on and will not affect the data stored within your virtual server
During the maintenance window, your virtual server will be cleanly shut down and will be unavailable while we perform the updates. A two-hour window is allocated, however the actual downtime should be much less.
We regret the short notice and the downtime required for this maintenance. However, due to the severity of these vulnerabilities, we have no choice but to take swift and immediate action to ensure the safety and security of our customers. For these reasons, we must adhere to a strict timetable, and will not be able to reschedule or defer this maintenance.
If you experience any issues following the maintenance, please feel free to reach out to us and we will be happy to assist.