MYVOIPAPP products blog – Page 5 – Software based SIP-PBX

Monitor events in IVR call flow

2019/12/02 Gilson

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.

<action method="callto" name="mainAction">
    <destination>100<destination>
    <monitor-events>
        <monitor-event detection="busy" nextaction="callto101"/>
    </monitor-events>
</action>

In this example, if the call invoked by ‘callto’ action is busy, IVR procedure will be changed to next action ‘callto101’.

Please refer to IVR-XML document for more details about “monitor-events” element.

IVR-XML example Download

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.

Configure miniSIPServer to trigger IVR-XML

miniSIPServer on Debian 10

2019/07/09 Gilson

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.

You can update Debian source list, then download and install miniSIPServer. No more action!

Debian organization, Congratulation!

T-MSS and L-MSS

2019/06/07 Gilson

Some customers deploy several miniSIPServer nodes to build their unified VoIP network around the world.

We give a simple document to describe this network and its configuration. There are two important concepts: T-MSS and L-MSS.

L-MSS means “Local miniSIPServer” which are deployed in local branches or offices to service their local SIP phones or gateways.

T-MSS means “Trunk miniSIPServer” which is used to bridge all L-MSS servers.

Please click here to get more details.

Refine “SIP over TLS”

2019/03/13 Gilson

Some customers report a crash problem to us. All of them deploy “SIP over TLS” in their VoIP networks. We have upgraded miniSIPServer to latest V35 (build 20190313) with following key modifications.

(1) In the latest miniSIPServer, SSL library has been upgraded to the latest version.

(2) Only TLSv1.2 method is kept, that means SSLv2, SSLv3, TLSv1 and TLSv1.1 are cut. When we did research on customers’ problems, we found some bad guys were trying to use the bug of SSLv3 to hack into MSS. We have to move all these methods out to defend that. In future, we will add other methods, such as TLSv1.3. At this time, we need confirm SIP phones can support TLSv1.2 too if we want to deploy SIP over TLS.

In another way, we refine “SIP over TLS” document to provide a simple demo on how to create certificate files.

Public address of SIP server

2018/12/30 Gilson

With the latest miniSIPServer version, it can be configured with several addresses, such as “local address” and “public address”. Please refer to following figure.

In normal, if miniSIPServer is deployed in public network with public address, it is unnecessary to configure independent “Public address” in above SIP configuration since the local address is public address itself. But in some network environment, for example, miniSIPServer is in a NAT and need serve outside users, we need configure “public address”. Outside users can use this “public address” to work with miniSIPServer.

If the network bridges several sub-networks, such as private network and public network, several different private networks, and so on, some SIP phones MUST use miniSIPServer private address, others use miniSIPServer public address, here we suggest to enable “relay media” in local users’ configuration which will make miniSIPServer to relay audio streams between these networks.

Scheduled Cloud-MSS maintenance

2018/12/22 Gilson

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

Maximum concurrent calls of local user

2018/09/27 Gilson

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.

Refined openAPI document

2018/08/16 Gilson

miniSIPServer provides openAPI interfaces for customers who want to operate or manage miniSIPServer through their own systems.

Previous openAPI was stored in GitBook and has few interfaces. Now we migrate it back to our official website. Please refer to following document.

https://www.myvoipapp.com/docs/mss_services/openapi/index.html

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.