SIP server for windows and Ubuntu system
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.
Some VoIP providers support “SIP trunking” services, most of them require user name and password for authorization.
To work with such SIP trunking, we need configure “external lines” in miniSipServer to establish connection with the VoIP carriers’ server.
In the external line configuration, we can configure peer server address (or domain name) , user name and password for authorization of register process and call process.
Both LTS and stable versions are upgraded. It is very important to our customers to upgrade your system to latest versions.
LTS version is upgraded to V4.0 to replace previous V2.x, more stable, more rich services.
Stable version is upgraded to V6.0. New features will be researched and developed based on this version.
V5.0 version is reserved for cloud-miniSipServer.
MSS V3 is updated to build 20110901 to fix a bug in SIP stack to support “Min-Expires” header in SIP messages.
This bug is also fixed in latest MSS V4.
MSS V4 is updated (build 20110831) to support following features:
(1) Store CDR into database if MySQL database is active.
(2) In prepaid services, such as calling card, caller prepaid and so on, “one-time fee” is supported. This means the fee will be reduced from balance immediately one time when the call is answered.
We are often asked this question about audio codec or video calls. Some customers want to confirm whether miniSIPServer can work well with their SIP phones/clients who can support several audio codecs or video calls.
We often answer that it depends on their own SIP phones/clients. miniSIPServer, not matter local-miniSIPServer or cloud-miniSIPServer, doesn’t care media codec by default.
Why we say that?
Please have a look at following figure which describes a basic media process model of miniSIPServer in a normal basic call.
From this figure, we can see:
(1) miniSIPServer only controls call signals.
(2) Media stream is processed by the clients themselves.
This model will reduce the work-load of the server since all media streams are bypass server, so it is unnecessary for the server to support several media codecs.
But in some service scenarios, miniSIPServer need play audio to the SIP phones. For example, in auto-attendant or calling card services, miniSIPServer need prompt audio announcement and collect user input information. Obviously, miniSIPServer is required to support audio codecs in these scenarios.
That’s right. If miniSIPServer need process audio stream, it can support two audio codecs: G711a (PCMA) and G711u (PCMU).
By the way, after playing audio stream, the final media stream will be processed still in end-to-end model. Please refer to following media model of miniSIPServer for these scenarios:
