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.
miniSIPServer in Deepin market
Yes, this system is very beautiful. After install miniSIPServer, you can find it in its software market.
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.
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.
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.
miniSIPServer on Debian 10
You can update Debian source list, then download and install miniSIPServer. No more action!
Previous miniSIPServer versions only limit “maximum concurrent outgoing calls”, and didn’t limit the total concurrent calls. Normally, it can fit most requirements since we think SIP phones or SIP clients should be able to limit their incoming calls. In recent days, some customers response that their SIP phones don’t have enough functions and hope miniSIPServer to be able to limit total concurrent calls of each SIP phone. To fit this requirement, we upgrade miniSIPServer to V34. Please refer to following figure.
Maximum concurrent calls
You can configure “maximum concurrent calls” to be zero. In this strange scenario, the SIP phone will never receive call and cannot make any outgoing call. It is to be noted that “maximum concurrent outgoing calls” should be smaller than “maximum concurrent calls” because “maximum concurrent calls” limits both outgoing calls and incoming calls together.
In this document, we provide many interfaces almost cover all necessary interfaces for basic calls, such as SIP trunk, external lines, routing, and so on.
Hope it can be helpful for your solution. If you need us to provide more interfaces, please contact us. All suggestions are appreciated.
Sometime, we may fail to make outgoing calls through SIP trunk if its service provider has some problem, such as no enough resources, and so on. If customers configure several SIP trunks and they are from different service providers, we can configure miniSIPServer to try another SIP trunk to continue outgoing calls.
In the SIP trunk “outgoing call” configuration, please configure a chained SIP trunk described in below figure.
Previous miniSIPServer has a trace tool which is “trace all”. It can capture and trace all SIP calls which MSS receives or sends out. This tool is very useful when we build the VoIP network at the first step. But it is almost useless in an exist working environment.
It is dangerous to capture ALL SIP calls in a working system since there are too many SIP messages and inner information. By default, we can filter the call according to caller number or called number. In the recent V33 version, we disable “trace all” and replace it with “trace on IP address”. Please refer to following figure.
Trace on IP address
With this tool, we can capture a specific complete IP address, such as “10.0.0.101”. We can also set a part of IP address to capture some SIP calls from some IP addresses, such as “10.0.0”, in this scenario, all SIP calls from IP addresses begin with “10.0.0” will be captured. By the way, we can also set IPv6 address with this tool.
Now you can see this tool can be used in both lab environment and working environment.
Finally, we release the latest LTS version V32. It has been a long time since we released the first V32 for test. During these months, we have updated web UI, GUI, SIP core and lots of items. This is a very exciting version and we will provide FIVE years tech support for it.
In another way, the latest stable version is V33 now. From this version, Linux (X86-32) systems will not be supported. New features and services will be developed on this version.
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.
We rebuilt miniSIPServer V32 for the latest Raspbian.
That means only Raspbian (Stretch) will be supported now. If you are still running MSS on Jessie or older systems, please upgrade Pi before you prepare to upgrade MSS.
If you are still working on Pi1 or Pi2 and don’t want to upgrade Pi, you can still try MSS previous versions. For example:
