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.
Hope you can enjoy these versions!
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
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.
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:
V31 https://www.myvoipapp.com/download/backup/mss_v31/pi/mss_v31_pi_u20.deb
V30 https://www.myvoipapp.com/download/backup/mss_v30/pi/mss_v30_pi_u20.deb
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.
We have updated our website with the latest bootstrap v4. Please take a look at it.
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.
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.