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SIP over TCP

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As we know, most SIP devices use SIP over UDP by default,  however, some SIP servers or communication servers only support SIP over TCP. We have to say that’s really strange. Our customers need work with such devices and we upgrade MSS to V10.5 to support SIP over TCP to fit this requirement.

It is very easy to configure MSS to support this feature. If you are interesting in it, please refer to following document for more details about it:

http://www.myvoipapp.com/docs/mss_services/sip-over-tcp/index.html


miniSipServer for ubuntu 12.10

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Yesterday we got and installed latest Xubuntu/Kubuntu/Ubuntu 12.10, then we installed latest miniSipServer V10.5 on these platforms. The result is perfect! It is no problem to run MSS without any modification. In another way, we would like to suggest Xubuntu … Continue reading

Gtalk to SIP

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We are very exciting to release miniSipServer new version 11! This is a great version and can support ‘Gtalk to SIP’ feature now.

What that means? It means you can connect your VoIP network with Gtalk network now. Gtalk users can call to your extensions directly, use your VoIP services, and so on. We bring Gtalk into your enterprise communication system. Funny!

Please refer to our online document for more details:

http://www.myvoipapp.com/docs/mss_services/gtalk/index.html

New versions, same experience!

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We are pleasure to announce that local-MSS V11.2 is released and cloud-MSS is upgraded at the same time.

The most important is that all these versions have almost same user experience! As we know, local-MSS has quite different UI with its own web UI and cloud-MSS. Some customers are often  confused when they migrate local-MSS to cloud-MSS since they have quite different UI. From now, they have almost same UI and same operations.  Please take a look at following figures:

GUI of local MSS

GUI of local MSS

web UI of local MSS

web UI of local-MSS

main UI of cloud-MSS

main UI of cloud-MSS

They are almost same, right? Of course, some details are different. We have tried to make them as same as possible to provide unified user experience. Hope you can enjoy it. Take a try please! It is interesting.

 

 

 

 

miniSIPServer LTS upgraded to V12

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The latest LTS is upgraded from V6.6 to V12 which has been tested and deployed around world by our customers.

V12 has much more features than previous V6.6.x and stable enough to be a new LTS version. Of course, we provide at least one-year supporting service for this new version.

Previous LTS version (V6.6.x) will be kept in supporting list until end of this year (2013).

At the same time, stable version is upgraded to V13. New features will be merged into this version and V12 will not include any new feature or service.

SIP over TLS

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With latest V13.1, MSS can support “SIP over TLS” now. In some enterprise communication systems, they often requre that SIP/Call signals or messages must be crypted to protect users information. Now it is very easy to do that with this new version.

Please refer to our online document for more details about this feature:

http://www.myvoipapp.com/docs/mss_services/sip-over-tls/index.html

Kubuntu/Ubuntu 13.04 released

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Yesterday, new Kubuntu and Ubuntu 13.04 were released. We download all these versions and try to run miniSIPServer V13.2. Everything is perfect, it is no problem to run MSS on the latest Kubuntu and Ubuntu 13.04. Since MSS is built … Continue reading

miniSIPServer for Debian 7

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We update and rebuild miniSIPServer V13.2 to fit Debian 7 system, now it is no problem to run MSS on Debian 7 and Ubuntu 12.04 systems with same DEB packages. If you are Debian 6 users, it is time to … Continue reading

How to run miniSIPServer on Ubuntu?

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If you are using Ubuntu (desktop), it is very easy to run MSS. Just download .deb file from our website and double click such .deb file to install it. Then you can find MSS icon in ‘internet’ or ‘network’ menu group and run it to start MSS.

If you are using Ubuntu (server), something is different. You need do more steps to install and run MSS. We describe details below.

First, you must download .deb file from our website, for example ‘mss_v13_u20.deb’. Then, you can run following command:

sudo dpkg -i mss_v13_u20.deb

At this step, Ubuntu will prompt error that some dependencies must be installed. Don’t warry, please continue following command:

sudo apt-get -f install

Then Ubuntu will try to get and install all necessary dependencies from Ubuntu software center.

In Ubuntu server, you cannot run MSS GUI since X windows is not started. You can run MSS CLI version whose file name is ‘msscli’ and can be found in directory ‘/opt/sipserver’.

So it is very easy to start MSS now:

/opt/sipserver/msscli

To visit and manage MSS system, you need visit its web system whose TCP port is 8080 by default. Please refer to following document:

http://www.myvoipapp.com/docs/mss_services/web_management/index.html

Reports in MSS

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Most SIP or PBX devices may have some reports to help administrator to check system’s status.

MSS has some reports too. Please click menu “Reports” to show them. MSS has two reports at this time:

(1) Basic call report

(2) Local user report

Both these reports are generated per hour. If there is not any value for each item, then the report will not be generated.

“Local user report” is simple. It only calculates how many local users are online per hour.

“Basic call report” is used to calculate how many calls MSS has been processed in one hour. There are some important items, such as “attempt”, “alert”, and “answer” and so on. It includes total durations and it can help administrator to understand how many calls have been answered and the avarage duration in one hour. If there are too many calls to make MSS heavy work-load, administrator can decide to upgrade MSS.

How to change TCP port?

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By default, MSS starts 5060 TCP port to receive or send SIP over TCP messages. 5060 is default TCP port defined by SIP standard. In MSS GUI or web UI, we can only change SIP UDP port.

If we want to change its TCP port, for example, we want to change it to 1234, we need do as following:

(1) In the MSS install directory, please create a text file whose name is ‘mss_var_param.ini’ file.

(2) In this INI file, please add following content:

[sip]
gVarSipTCPPort=1234

(3) Please restart MSS to enable it.

Trigger hunt-group for external lines incoming calls

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Some customers use external lines to connect VoIP providers’ servers. So it is required to trigger hunt-group for all incoming calls from such external lines.

Lets’ assume the external line number/account is ’123456′, so we just need trigger HG service according to this number.

Please click menu “services / multi-line hunting group / Detection configuration” to add a new record as following:

dial plan = default
called number = 123456 <-- this is the external line number
... ...

Others are same with HG service document:

http://www.myvoipapp.com/docs/mss_services/hunting_group/index.html

If the external line is configured with auto-attendant or routed to another destination number, what is the result? Don’t worry. HG detection has higher priority than these services, that means HG will be triggered even you configured AA or destination number in external line.

Call waiting service

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Most SIP devices, such SIP phones, can support “call waiting” service themselves. But for some old SIP devices, such as some simple SIP ATA devices, it is required PBX to provide “call waiting” service.

So we upgrade MSS to V13.6 to provide this service. In MSS, we name it as “simple call waiting” service because MSS’ call waiting service only support

(1) put original call into waiting queue if current called party is busy.

(2) reconnect previous calls if current called party is available.

So user cannot press ‘R’ key to switch calls. It is simple and we hope it can fit most customers requirements.

Please refer to our service document for more details:

http://www.myvoipapp.com/docs/mss_services/call-waiting/index.html

miniSIPServer V14

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miniSIPServer is upgraded to V14.2 now. With latest versions, following key features are included:

(1) For raspberry Pi. As we know, raspberry Pi is a very small PC run on Debian system, MSS can work on this hardware now. It is very helpful for some small business who dont want to maintain a PC hardware and save many cost.

(2) Provide open API. MSS can start web server and provide open HTTP API . 3rd parties can use this interface to intergrate with their own system. At this time, open API set can support: query MSS version; configure or query details of local users.

(3) Routing group. It is a very helpful feature for customers who have lots of external lines or SIP trunks. They can manage these lines or trunks into different groups for different users or for different calls. The workload of group is in round-robin policy.

Hope you can enjoy our new versions! If you have any questions or suggestions, please feel free to contact us. We are always here for you.

miniSIPServer & Ubuntu/Kubuntu 13.10

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The latest Ubuntu/Kubuntu V13.10 is released. We download and install it immediately in our lab. It is no problem to run miniSIPServer on these systems. But according to our experience and test result, we dont suggest customers to upgrade systems to V13.10 because it is not stable enough for business deployment now.

So if you are running miniSIPServer on Ubuntu/Kubuntu, it is better to keep your current versions and wait for next LTS version V14.04.


Two external lines, how to use specific one by dialing different called number prefix?

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Description

One of our customers has two different VoIP accounts, for example (1) 1234 and (2) 5678. It is required to select account “1234″ if users dial “9xxxx” numbers and select account “5678″ if users dial “8xxxx” numbers. The final numbers should delete these prefix “9″ or “8″ and “xxxx” should be sent to VoIP providers.

Solution

We can use MSS powerful “dial plan” features to fit this requirement.

By default, MSS uses called number prefix “9″ to distinguish outgoing calls to outsides. If there are several external lines and without any special configuration, MSS will select one of them in round-robin for each call. Now what we need do is to configure different called number prefix and select different external line for them.

Step 1: configure number transition

In this step, we need configure a record to delete number prefix “8″ or “9″ from called numbers. Please click menu “Dial plan / Transition” to add a record illustrated below.

Transition ID = 1
Transition type = delete
Start position = 0
Length = 1

Step 2: add new “Analyze called number” records

According to requirement, we need indicate MSS to analyze called number prefix “8″ and “9″ to use different specific external line. Please click menu “Dial plan / Analyze called number” to add two records.

Record 1: analyze called number prefix “9″

Dial plan = default
Called number prefix = 9
Route type = external line
Specific external line = 1234 <== use specific external line
Change called number = yes
Transition ID = 1 <== configured in step 1
Re-analyze after transition = no

Record 2: analyze called number prefix “8″

Dial plan = default
Called number prefix = 8
Route type = external line
Specific external line = 5678 <== use specific external line 
Change called number = yes
Transition ID = 1 <== configured in step 1
Re-analyze after transition = no

Refine called number

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V14.4 is updated to support a new feature in “dial plan” process. This feature is “refine called number”.

“Refine called number” can be used to refine called number before calls are routed to external lines or SIP trunks. It is the last chance to change called number to fit different requirements from peer VoIP servers.

For example, one of our customer has two VOIP accounts. One is from local provider, another is from international provider. These two VoIP providers have different number format requirements, and our customer only want to has one kind of dial plan for both of them. So we can configure “refine called number” to refine the final destination number to fit it. This scenario is illustrated below.

Scenario

As described above, there are two VoIP accounts, and users need dial “90xxxx” to make outbound calls. “9″ is MSS default outgoing call prefix. “0″ is required by local VoIP provider. At the same time, the international VoIP provider requires that the number format should be “0086xxxx”.

After compare these number formats, we can find that we only need change prefix “0″ to “0086″ for international VoIP account.

Step 1: configure an independent “outgoing group ID” for international VoIP account

Please click menu “Data / External line” and select the account to edit, then please click “Outgoing call” tab and configure following item:

Outgoing group ID = 1

Step 2: configure “number transition”

We need configure a new record to change preifx “0″ to “0086″. Please click menu “Dial plan / Transition” to add a new record:

Transition ID = 1
Transition type = Replace
Start position = 0
Length = 1
Replace string = 0086

Step 3: refine called number for specific outgoing group

Please click menu “Dial plan / Refine called number” to add a new record:

Outgoing group ID = 1 <== defined in step 1
Called number prefix = 0
Transition ID = 1 <== defined in step 2

Here we maybe have a problem: the called number prefix is “0″, why? why not analyze “9″ prefix? It is because that “9″ has been deleted in “analyze called number” procedure and the number has been changed to “0xxxx” before it is sent to external line or SIP trunk, so we should analyze prefix “0″ to refine final called number.

Happy new year! 64 bits miniSIPServer!

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Today we upgrade miniSIPServer V14 to support 64 bits Debian and Ubuntu. That’s a great news for some of our customers.

As we know, in some networks, customers have deployed 64 bits Debian or Ubuntu servers. So when we want to install previous 32 bits miniSIPServer in their servers, it is required to install some additional 32 bits libraries. Of course, it is no problem to do that, but some customers often think it could be unstable for their servers.

So we decide to migrate MSS to 64 bits to fit this requirement. Now it is unnecessary to care about 32 bits problem.

For customers who are using 64 bits Windows, it is unnecessary to care about 32 bits problem. Windows can process them in perfect way, so please keep current versions.

New service engine

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Today we release latest V15 for miniSIPServer. This version is focus on providing a new service engine which is written in Python script. That means almost all services are written in Python script files. New service engine is more flexible … Continue reading

miniSIPServer can run on Ubuntu/Kubuntu 14.04

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Today we download the latest Ubuntu 14.04 and install it in our lab to make some test. It is no problem to run miniSIPServer on this system. And it seem V14.04 is better than its previous version. So if you are … Continue reading
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