Our data center have scheduled maintenance windows during which the basic infrastructure will be updated.
During the maintenance window, all our Cloud-MSS nodes will be powered down and that means all virtual servers will stop working. While a two-hour window is allocated for the maintenance, the actual downtime should be much less.
The scheduled maintenance window is set for: Tuesday, January 8, 8:30 AM UTC
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.
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.
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.
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.
In some VoIP scenarios, we need configure “SIP trunk” to work with VoIP providers or gateways. When processing media streams, we hope (1) local users/phones should process their streams by themselves without MSS, and (2) MSS should help to relay media stream for all outgoing calls to peer SIP servers or gateways.
To fit these requirements, we update MSS V32 to be able to configure “relay media” item in “SIP trunk”. Please refer to following figure for more details.
By the way, MSS can only relay audio streams at this time, so video streams will be lost if you want to MSS to relay streams.
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.