Product release messages, including LTS, stable, development versions and patches.
miniSipServer is upgraded today to support “call recording” feature. This feature can be used to record local user’s audio stream.
Please refer to following service document:
http://www.myvoipapp.com/docs/mss_services/call-recording/index.html
2018-04-30 updated:
With the latest V32, miniSIPServer will not support this feature. Please try to run miniSIPServer in windows auto-start methods.
If you have tried miniSipServer on Windows platform, you can find it is so easy to configure or manage a IP-PBX because miniSipServer provides a perfect GUI interface.
You can find it is very easy to do that on Ubuntu/Linux because miniSipServer can support GUI interface on Ubuntu/Linux now! WINE is unnecessary now!
Please visit our download page to download miniSipServer .deb packages according to your requirements:
http://www.myvoipapp.com/download
After you download .deb file from our website, please click (or double-click) it to install MSS. No more actions! So easy, so funny.
After you finish your install, you will find miniSipServer at “Applications / Internet” group.
Following figure is a screenshot of MSS running on Kubuntu:
In another way, if you just want to run a command line sip server, you can also go to directory “/opt/sipserver” and find “msscli” which is a command-line miniSipServer and has the same core with GUI-miniSipServer. You can manage and configure it through web interface.
One of our customer has two office branch in different cities and two MSS have been deployed. Following figure describe the network topology:
The extensions of MSS1 are 1xx, such as 100,101, and so on.
The extensions of MSS2 are 2xx, such as 200, 201, and so on.
With previous MSS versions, it is no problem to send/receive instant messages between local users. But it cannot send instant messages to the extensions of another SIP server.
So we upgrade MSS to V6.1.5 to support instant messages between SIP servers.
To do that, we need establish SIP trunk between these SIP servers. Once you can make calls to the extensions of another SIP server (MSS), it will be no problem to send instant messages to them.
That means we need configure MSS with (1) peer server configurations, (2) Dial plan configurations which we have described in “how to use SIP trunk” document. Please refer to following document for details of SIP trunk:
http://www.myvoipapp.com/blog/2011/05/02/sip-trunk/
Local miniSipServer (stable) is upgraded to V6.1.4.
In this version, trial duration is changed from 20 days to 30 days. Some customers need more time to test MSS full features. We think it is very reasonable, so V6.1.4 or higher can support more trial duration.
In another way, this version is updated to support “direction” parameter in PRESENCE NOTIFY message which is very necessary for call center applications. This parameter is used to indicate the operator current call is outgoing or incoming.
Cloud-MSS is upgraded to support uploading customized audio file.
In auto-attendant or voice-mail features, customers often hope to use their own audio files. It is very easy to replace system audio files in local-MSS. But in cloud-MSS, it has little problem, such as file transferred through internet, etc.
With the new version, customers can upload audio file through HTTP. Of course, the audio file should have the same format same with local-MSS required.
After sign into your cloud-MSS account, please follow the steps described in below figure:
We need mention that please be paitent because cloud-MSS will try to download the audio file from your HTTP URL immediately. It could be slow if the network is not very good.
Gianfranco, one of our partners, released a new softphone SDK. This SDK provides full SIP features, such as HOLD, Transfer, and so on. It works as .net component or .com component. It is easy to build a new softphone base on this SDK.
Most important, it has been tested with lots of SIP servers or proxy, such as our SIP server (miniSipServer), Asterisk, OpenSER and so on.
If you are interesting in building your own SIP softphone, this SDK could be a very good choice for you.
The official website is http://www.lullabysdk.com/
miniSipServer V6 is upgraded to V6.1.3. This version can support SIP INFO message to transport DTMF signals.
Some VoIP carriers’ servers or VoIP gateway can only support this INFO message to transport DTMF signals which are very necessary for IVR services, such as auto-attendant.
There are several content type in INFO message to transfer DTMF. MSS can support “application/dtmf-relay” content type.
Now, MSS can support INFO and RFC2833 for DTMF signals.
MSS IVR-XML document is released. This document describes details about MSS IVR-XML, includes structure, XML elements, action definitions, and so on.
We hope this document can help you to understand IVR-XML and write your own IVR process.
Please visit following URL to read the document:
http://www.myvoipapp.com/docs/mss_services/IVR-XML/index.html
MSS stable version is upgraded from V6.0 to V6.1.
In this new version, IVR-XML is supported and we can use XML to write IVR services. It is very easy and funny. We re-write auto-attendant service with IVR-XML, you can see it is very simple. The latest auto-attendant service file can be found in sub-directory ‘xml’ and file name is ‘aa.xml’.
The latest AA document is updated together, please refer to :
http://www.myvoipapp.com/docs/mss_services/auto_attendant/index.html
Most important, it is very easy to setup multi-levels menus in IVR services.
Please take a try and you will like it.