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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.

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.

miniSIPServer on Ubuntu 18.04

miniSIPServer on Ubuntu 18.04

It is perfect to run miniSIPServer on Ubuntu 18.04 which is the latest LTS version.

We have tested some scenarios with miniSIPServer on Ubuntu 18.04, everything is OK. We strongly suggest you to upgrade your Ubuntu system to this version.

miniSIPServer on Ubuntu 18.04
miniSIPServer on Ubuntu 18.04
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.

miniSIPServer on Debian 9

miniSIPServer on Debian 9

It is a good news to see that the latest Debian 9 is released. We have downloaded and tested it in our lab.

Debian 9 is very interesting. Since it is a stable version, it is important for us to run miniSIPServer on this system. We have to find that so many libraries and softwares have been changed or upgraded. Previous MSS versions cannot work on it by default.

We did lots of work to fix these conflict and upgrade MSS to V31 (build 20170621). And we are exciting to announce that the latest versions can still work on previous Debian systems, such as Debian 7 and Debian 8. Everything is perfect now!

If you want to try Debian 9, please upgrade MSS to the latest V31. And please refresh the document for more details about libraries.

Concurrent calls of SIP trunk

Concurrent calls of SIP trunk

By default, MSS previous versions don’t limit concurrent calls of SIP trunk. That means you can make or receive calls as much as you can. If peer sides don’t have enough resources, they will reject calls by themselves. But now in some scenarios, customers hope MSS can handle concurrent calls and limit them automatically.

To fit this requirement, we upgrade MSS to provide concurrent calls configurations in SIP trunk. Too much calls will be rejected by MSS itself. Please refer to following figure for more details about these items.

Concurrent calls of SIP trunk
Concurrent calls of SIP trunk

Please pay attention to these.

(1) These items are independent. You can configure different values for them to limit different concurrent calls for outgoing calls and incoming calls.

(2) If one of them is zero, in fact all them can be zero, that means only incoming calls can be received, or can only make outgoing calls outsides.

Modification of “one number” service

Modification of “one number” service

We upgraded miniSIPServer V30 today to change “one number, multi-devices” service in local user’s configuration. In previous versions, we don’t need configure anything to enable this feature in local user since it was enabled by default. Customers think it is good idea to reduce configuraiton workload, but it brings new management problem. In fact, they hope to be able to control which local users can have this feature. In most scenarios, only some local users have several phones with same number, others are not permit to do that.

To fit this requirement, we add a new optional item in local user’s configuration. Please refer to following figure for more details. By default, this service is not enabled now until you configure it obviously.

One number service right in local user's configuration.
One number service right in local user’s configuration.

This modification is applied to cloud MSS too.

Citel Technologies, Inc. Announces Interoperability with MyVOIPApp miniSIPServer

Citel Technologies, Inc. Announces Interoperability with MyVOIPApp miniSIPServer

AMHERST, NY and ShenZhen, P.R. China — August 4, 2016 — Citel Technologies, Inc., is pleased to announce that it has successfully completed interoperability testing with MyVoIPApp’s miniSIPServer, a software-based SIP PBX, designed for small and middle size companies, miniSIPServer is very easy to use with rich features and can work on multi-platforms, such as Windows and Linux whilst also fitting both IPv4 and IPv6 networks.

The Portico™ TVA™ offers companies the means to migrate their customers to VoIP without the unnecessary burden of ripping and replacing existing cabling infrastructure, purchasing new IP phones and installing Power over Ethernet switches. Customers can retain their existing digital, analog and Centrex phones without removing the existing switches by SIP enabling those phones through the use of the Portico™ TVA™.  This is a quick and cost-effective means of VoIP migration.

“Used together, the two innovative solutions provide companies with the fastest and most cost-efficient means to upgrade from legacy systems to modern Unified Communications,” said Ian Gomm, VP of Sales & Marketing for Citel.

Citel and MyVoIPApp kicked off interop testing with miniSIPServer at the request of a high-profile customer who was looking to use this Windows-platformed softswitch for a cross-country network of customer service training centres. “We have undertaken interoperability with many IP PBXs over the years, and know that sometimes some systems are easier than others to complete. MiniSIPServer was new to us, but I was really pleased to hear engineers’ initial feedback that the system was easy to work with and interop had gone smoothly” said Andrew Davies, VP, Engineering at Citel. He continued “Then they showed me some quite sophisticated presence monitoring and Busy Lamp Field functionality and I saw the product was a great fit with the Portico™ TVA™, wherever digital business phones were the preferred handset. We look forward to taking the relationship further.”

About Citel Technologies, Inc. (citel.com)

Citel enables SMBs, large enterprises and service providers to realize the cost and productivity benefits of IP telephony while at the same time leveraging their existing PBX infrastructure. Businesses with single or distributed locations and PBX vendors can now deploy next-generation IP applications and services at their own pace, with minimal business disruption. Service providers can deploy Hosted IP telephony services quickly, without having to “rip and replace” existing enterprise PBX handsets and LAN cabling. Citel is based in Amherst, New York with offices in Loughborough, England (UK) and Toronto, Canada.

About MyVoIPApp (myvoipapp.com)

MyVoiPApp was founded in 2007, in ShenZhen, P.R. China. Although a small company its employees all have more than ten years experience in the field of communications and its focus on communication technology has facilitated strong growth and penetration in the VoIP marketplace.

Refined SMTP library

Refined SMTP library

In voice mail feature, MSS need use SMTP library to send emails. Since MSS can embed Python script functions, it is easy to use Python-smtplib to send email. That’s what we done and it works well, we are satisfied with it.

But smtplib is too old ( in python 2.7) to fit some modem SMTP servers’ requirements. It also has a shortage. It is synchronous. That means it can block thread when sending a email, then its performance is poor and cannot fit our requirements in cloud system.

Something is changed, we want MSS to be better, so we develop a new SMTP library to send voice mails. This SMTP library is asynchronous and can work perfectly with most SMTP servers. And, it is written in C/C++ language.

We upgraded MSS V23 and cloud-MSS to replace python-smtplib with this new SMTP library.

Hope you can enjoy latest versions.

By the way, since MSS V23 has been released for several months and we got very better result, we think it is time to release new LTS version which is V24  and new stable version which is V25 in the end of this year or in the beginning of next year.

miniSIPServer on Ubuntu 15.10

miniSIPServer on Ubuntu 15.10

It is no problem to run miniSIPServer on latest Ubuntu 15.10 system. Please refer to attached figure.

miniSIPServer on Ubuntu 15.10
miniSIPServer on Ubuntu 15.10

As you know, it is better to stay with previous Ubuntu LTS versions, such as 12.04 and 14.04.