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Category: miniSipServer ( local )

SIP server for windows and Ubuntu system

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.

Refined openAPI document

Refined openAPI document

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.

Chain to next SIP trunk

Chain to next SIP trunk

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.

Chain next SIP trunk
Configure chained SIP trunk for current SIP trunk
Trace on IP address

Trace on IP address

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.

Trace on IP address
Trace on IP address

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.

New LTS V32

New LTS V32

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!

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

 

miniSIPServer for Raspbian (Stretch)

miniSIPServer for Raspbian (Stretch)

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

Relay media streams of SIP trunk outgoing calls

Relay media streams of SIP trunk outgoing calls

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.

Configure "relay media stream" item in SIP trunk outgoing call
Configure “relay media stream” item in SIP trunk outgoing call

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.

Store your own audio files

Store your own audio files

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

Connect to sonetel

Connect to sonetel

“sonetel.com” is a VoIP carrier who can provide local phone numbers. We can add “external line” to work with their servers. According to its description, there are some special items need to be cared:

  • Sonetel uses email address as SIP account and
  • It deploys SBC or proxy to process all incoming SIP messages.

Here we give a simplel example to describe how to work with Sonetel. We assume the SIP account is “abc@gmail.com”.

In MSS, please cilck menu “data > external line” to add a record.

Configure Sonetel lline
Configure Sonetel lline

In “Basic” tab, the line type should be “Connect to peer VoIP server”, the Account should be “abc” and the Domain is “gmail.com”.

By the way, the Password is the password you sign up in Sonetel, not your own email password.

Since all SIP messages are processed by sonetel SBC/Proxy, we need configure “outbound” information in the “Outgoing call” tab. Please refer to following figure.

Sonetel SBC
Sonetel SBC

The Sonetel proxy address is “sip.sonetel.com” which should be described in the email sent by sonetel.