Browsed by
Category: miniSipServer Cloud

PBX in cloud for small business

Scheduled Maintenance

Scheduled Maintenance

Our data center updates us that maintenance is required for one or more of our servers’ physical hosts. The hosts will be rebooted, and a number of critical updates will be installed.

The maintenance schedule in UTC is as follows:

2020-08-07 03:00:00 AM UTC

Important to know:
(1) A 2-hour window is allocated, however, the actual downtime should be around 45 – 60 minutes.
(2) Your virtual server(s) will be cleanly shut down and will remain inaccessible during the maintenance window.
(3) You might reboot your SIP phones/devices to register to virtual server(s) again once the virtual server(s) is(are) back to work.

Please let us know if you have any questions or concerns. Thanks!

Personalized voice in VM service

Personalized voice in VM service

In normal, cloud miniSIPServer almost has the same services with local miniSIPServer. But for some limitations, there are some different features between them. For example, voice mail service is different.

With local miniSIPServer, each local user or extension can prompt their own audio in voice mail procedure. But with cloud miniSIPServer, each local user can only prompt the unified audio associated with their virtual PBX server. Of course, the default unified audio can be replaced with customer’s own audio file.

Now cloud miniSIPServer is upgraded. In a virtual server, each local user can has its own audio file now. Please refer to following figure. You can see a new item “Personalized voice ID” which can be different for different user.

Audio ID configuration in local user's configuration
Personalized audio configuration

Of course, the audio file cannot be uploaded to virtual server by customers themselves. If you want to upload audio files, please send them to our support team, we will upload them to your virtual server manually.

Once the audio files are uploaded, you can manage them by yourself. Please refer to following figure.

Audio files and ID management
Audio files management

In another way, you need follow some rules to create your own audio files, such as the file format and the audio codec, and so on. Please refer to online document for more details.

Clear zombie virtual servers

Clear zombie virtual servers

In the end of next month (2020-01-31), we will clear some zombie virtual servers from our cloud system.

If the virtual server has following features, we will define it to be a zombie node and will be cleared in this action.

(1) The virtual server has not be signed in or activated since 2 years ago. If you didn’t sign into your account or virtual server after 2017-01-01, please sign into your account at less one time to avoid that.

(2) And there isn’t any SIP clients register to the virtual server, or there isn’t any SIP calls after 2017-01-01.

Zombie virtual servers waste our resources and are not affair to other customers, so please pay attention to this action and thanks for understanding.

2020-02-13 updated: This task is finished now. In future, we will keep clearing zombie virtual servers without notification. If your virtual server is not signed in or activated in recent 1 year, please pay attention to this.

Monitor events in IVR call flow

Monitor events in IVR call flow

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.

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
Configure miniSIPServer to trigger IVR-XML
“SIP over TLS” enabled in cloud system

“SIP over TLS” enabled in cloud system

We upgraded cloud miniSIPServer system for some key features. The most important feature is “SIP over TLS”.

By default, cloud system opens TCP port 6060 to accept “SIP over TLS” messages. It is used to encrypt SIP messages. This feature is available for all virtual servers without any additional fee or configurations.

Now, SIP phones can connect to cloud miniSIPServer nodes with “SIP over TLS”, but “external line” and “SIP trunk” still can only use “SIP over UDP” to work with voip providers.

This feature can only encrypt SIP messages. If you want to encrypt media streams, such as audio stream and video stream, you need enable SRTP in your SIP phones. By default, media streams are bypass and processed by SIP phones themselves, cloud miniSIPServer will not process these media streams.

Please visit online document “SIP over TLS” for more details.

Emergency maintenance

Emergency maintenance

Our data center has detected an issue affecting the physical host on which our cloud system resides.

During this emergency maintenance:

  • Your virtual server may not be accessible.
  • No action is required from your end at this time.
  • We don’t have an ETA on when your virtual server will be brought back to its original state.

Once data center administrators have completed the investigation, we will update this blog with how we plan to proceed from here.

Your patience and understanding is greatly appreciated.

[Tue Apr 9 05:05:15 UTC 2019] updated: Everything is OK now and the service was broken for one hour this time. We will keep watching. If your SIP phones are out of services, please reboot them to take a try. If you still have problem, please update us. Thanks.

Refine “SIP over TLS”

Refine “SIP over TLS”

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.

Scheduled Cloud-MSS maintenance

Scheduled Cloud-MSS maintenance

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

Maximum concurrent calls of local user

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.

Maximum concurrent calls
Maximum concurrent calls

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.

External lines configurations

External lines configurations

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