Send or receive instant messages

Send or receive instant messages

The latest version of miniSIPPhone is released today to support two key features: (1) Contact, and (2) Instant messages.

It has a new window to create and manage contact list like belowing:

miniSIPServer contact list

In the contact window, you can select the target user and double click it to make a call out, or you can press ‘C’ key or click ‘Call’ button to do that.

If you want to send instant messages, you can select the target user and press ‘M’ key or click the ‘Message’ button, then you will get instant messages’ windows:

Instant message window on Windows system
Instant message window on Linux system

One instant message window is used for one user. Each window has three areas: (1) Display area. It displays both incoming messages and outgoing messages. (2) Input area. You can input the instant message content here, and press ‘Ctrl+Enter’ keys to send the message out. (3) ‘Send’ button. Click it to send the message out.

At this time, miniSIPPhone uses SIP-MESSAGE to send and receive instant messages, and can only support plain text messages, so you cannot insert images, files, audios and videos into the messages.

Of course, miniSIPPhone can run on Windows system and Linux system (including AMD64 and ARM64). In fact, the users in above figure run miniSIPPhone on different systems.

Hopy you can enjoy it. 🙂

the tel URI

the tel URI

As we known, the VoIP (SIP) domain always uses SIP URI to establish call sessions. To work with traditional PSTN networks, we need gateways (or SBC) to bridge two networks. Most of these gateways can support SIP URI, so we can always use SIP trunk to estanlish connections between VoIP and PSTN with SIP URIs which are as same as connections between VoIP domains.

But some gateways cannot support SIP URIs, they can only accept traditional telephone numbers which are the tel URIs defined in RFC3966. The URI is in “<tel: xxx>” format, not in “<sip:name@address>”format. Please refer to the figure below.

the tel URI network

miniSIPServer can always accept the tel URI from peer sides, but never send out the tel URI. In recent months, several customers ask us to support sending out the tel URI through SIP trunks to work with some PSTN gateways. So we upgrade miniSIPServer to V60 (build 20250208) to update the SIP trunk functions. In the “outgoing call” of SIP trunk, we can select “Use the tel URI” item, then miniSIPServer will use <tel> URI to make outgoing calls for the SIP trunk.

the tel URI configuration of SIP trunk in miniSIPServer.

For incoming calls of the SIP trunk, it is unnecessary to configure anything since miniSIPServer can accept both SIP URI and TEL URI.

miniSIPPhone for Linux (Debian/Ubuntu)

miniSIPPhone for Linux (Debian/Ubuntu)

Finally, miniSIPPhone is upgraded to V10. The most important thing is that it can support Linux system now. Of course, the distro must be Debian or Ubuntu. As same as miniSIPServer, Debian must be V10 (Buster) or higher versions, and Ubuntu must be V18.04 (Bionic Beaver) or higher versions.

Both X86_64 (amd64) and ARM64 (AArch64) are supported.

It is quite easy to run SIP phone on Linux system now. Please visit our website to download the latest version:

For example, you download “msp_v10_amd64.deb” and install it with following command:

sudo dpkg --install msp_v10_amd64.deb

Then you can click the linker to run miniSIPPhone:

If you want to uninstall miniSIPPhone, you can run following command directly to remove it:

sudo apt remove minisipphone
About Debian and Ubuntu versions

About Debian and Ubuntu versions

For miniSIPServer V60, the lowest requirement of Debian is changed to the “oldoldstable” version which is Debian V10 at this time. That means Debian V8 and V9 will not be supported now.

Of course, the lowest requirement of Ubuntu is also changed to Ubuntu V18.04.

Please refer to the document for more details about Linux systems.

Conference room and others

Conference room and others

miniSIPServer is upgraded to V60 which is the latest stable version for business development. The first big thing is “conference room” feature which provides conference calls for local users. At most 5 clients can join the same conference call. Please refer to the service document for more details. Cloud miniSIPServer is also upgraded to support this feature.

In another way, as we have posted in previous blog, several services are finally removed from local miniSIPServer, such as calling-card and call-shop. These features were important for some of our customers, but it is time to say good-bye now.

Refine miniSIPServer

Refine miniSIPServer

As we know, miniSIPServer was developed about 20 years ago. Lots of services and features are added into miniSIPServer to support more and more customers.

Recently we have reviewed all these services. Some services have so long history that we have to think whether they are suitable for current environments, for example call-shop, calling card, and so on.

Next version will focus on refining or clearing some services. miniSIPServer will step into next stage and be more faster, more stabler.

If you want to deploy FXS gateways,…

If you want to deploy FXS gateways,…

FXS gateways can help to connect traditional phones to VoIP deamon. The common topology is below. 

VoIP deamon <--> miniSIPServer <--> FXS gateway <--> traditional phone

In normal, one FXS gateway connects to one traditional phone, but some FXS gateways can connect to several phones at the same time. We need to pay attention to this scenario.

The FXS gateway need to bind several SIP users accounts since it can connect to several phones. In another way, the gateway uses one URI address (including IP address and port together) to register these SIP accounts to miniSIPServer. That meas several SIP users will connect to miniSIPServer with the same address.

If one of the users is configured with wrong information, and the gateway keeps sending SIP messages to miniSIPServer, it will trigger ‘fail to ban’ feature, then the gateway’s address will be blocked by miniSIPServer. As we said, since the gateway uses one address to register several SIP users to miniSIPServer, other SIP users will have to be blocked together.

In this scenario, we need to stop ‘fail to ban’ for the gateway. We can add the address of the gateway into white list. Please click menu ‘services – IP address black-white list’ to add a record to accept the IP address. For example:

IP address black-white list
Run miniSIPServer on Ubuntu 24.04 LTS (Noble Numbat)

Run miniSIPServer on Ubuntu 24.04 LTS (Noble Numbat)

Ubuntu 24.04 is the latest LTS (long-term support) version, so it will be deployed widely in business environment. We install miniSIPServer on this important version and make some tests. The result is perfect! Please refer to the figure below.

Run miniSIPServer on Ubuntu 24.04

If you want to deploy a new VoIP network on Linux system, Ubuntu 24.04 could be a good choice.

Please refer to online document for more details about how to run miniSIPServer on Linux system.

Support TLSv1.3

Support TLSv1.3

miniSIPServer recently is upgraded to support TLSv1.3. This modification doesn’t affect configuration, so you need to do nothing if you upgrade your miniSIPServer to the latest versions.

Two modules could use TLS transport: (1) SIP server, and (2) Embeded HTTP server. If your SIP phones can support TLSv1.3, it is better to use TLSv1.3 to protect communication. Please refer to “SIP over TLS” document for more details. Both local miniSIPServer and cloud miniSIPServer can support SIP over TLSv1.3 now.

By default, miniSIPServer starts an embeded HTTP server for web management. If you want to manage it through the pubilc network, you have to enable TLS transport to protect HTTP information. In another way, most navigators, such as Chrome, Edge, Firefox and so on, can support TLSv1.3 now. Please refer to “web management” document to enable encrypted HTTP.