Browsed by
Category: How-to

Tips of using MyVoipApp products

how to link fxo gateway?

how to link fxo gateway?

It is a often asked question. In MSS, you can use “external line” to connect to your fxo gateway. In our forum, we give a simple description about how to link to linksys 3102. Maybe you can refer to this document for help.

http://www.myvoipapp.com/forums/viewtopic.php?f=5&t=419

(Sorry, since our forums have been removed, above document cannot be visited. Please refer to online F.A.Q document. Thanks. )

Run miniSipServer on Ubuntu

Run miniSipServer on Ubuntu

2011-12-26 updated:

Please visit following blog for latest message:

http://www.myvoipapp.com/blog/2011/12/26/the-most-easy-to-use-ip-pbx-for-ubuntulinux/

======================================

Since miniSipServer V4.0 is released, it is very easy to deploy VOIP server on Ubuntu now.

The difference between MSS V4.0 (windows) and MSS V4.0 (Linux) is that MSS (linux) doesn’t provide GUI management. It only provide web management. It is still very easy to setup and manage the VOIP network.

First, please download xxx.tar.gz file from our website. In following section, we give example on how to use miniSipServer V4.0 (20 client) :

http://www.myvoipapp.com/download

It is a zip package file, once you have download it, please use following command to unzip it.

tar zxvf mss_v4_u20_i386.tar.gz

After that, you can find a “install.sh” file in the directory. Please run it to install MSS. During the process, some 3rd parties software, such as libmysqlclient, will be installed automatically from Ubuntu software center.

sh install.sh

Then, you can run miniSipServer now:

sudo msscli

When the MSS is running, you can find following information in the command window:

HTTP server is running at port 8080

It means MSS has started HTTP server at TCP port 8080, you can  visit it now to manage your SIP server by using firefox, chrome, IE,etc. For example, you can open local firefox and type following address to visit it:

http://localhost:8080

We have to mention here the default administrator password is blank. So once you login into the web management, the first thing you need do is to set administrator password. Please click “basic configuration / system configuration” to do that.

So easy, right? If you have any questions, please fell free to contact us. We hope you can enjoy it!

sip trunk

sip trunk

In VOIP depolyment, “SIP trunk” is often used to establish a connection with peer sip servers or gateways. For example, in most DID services deployments, SIP trunk is required to send or receive DID calls.

The difference between “SIP trunk” and “External lines” is that SIP trunk doesn’t require authorization during the call. That means, “external line” is server-to-users mode and “SIP trunk” is a server-to-server mode.

It is very easy to establish SIP trunk in MSS.

For example, we want to establish SIP trunk with peer server whose domain name is “sip.demo.com” and its SIP port is 5060 which is a default SIP UDP port.

step 1: add the server into MSS servers list

Please click menu “data / peer servers” and add a new record with following information:

peer server id=1
description = demo sip server
server address = sip.demo.com
server port  = 5060

step 2: process incoming call

Once we receive incoming calls from peer servers, we want to route them to local users. We can use “dial plan” to do that.

For example, we want the DID incoming calls whose called numbers prefix is “1234” to local users, such as 1234100 to local user 100, 1234101 to local user 101, etc.

Please click menu “dial plan / transition” to configure a number transition:

transition ID = 1
transition type = delete
start position = 0
length = 4

Please click menu “dial plan / analysis called number” to configure a record to route DID numbers to local users:

dial plan = default
called number prefix = 1234
route type = local user
change called number = yes
transition id = 1

step 3: process outgoing call

We want our outgoing calls to be routed to such peer SIP server/gateway. We still need configure “dial plan” to do that.

For example, we want all calls whose called number prefix is “00” should be routed to such SIP server, such as “008613800138000”, etc.

Please click menu “dial plan / analysis called number” to add a new record with following information:

dial plan = default
called number prefix = 00
route type = SIP trunk
peer server ID = 1

 

How to upgrade miniSipServer?

How to upgrade miniSipServer?

In different scenario, this question can be different. In fact, this question can be seperated to several questions:

Q1: What we need do when we upgrade MSS?

The most important thing is backup your previous configuration files.If there is any exception during the upgrade, we can roll-back to previous version.

All miniSipServer configuration files are stored in the ‘config’ sub-directory where MSS is installed.So it is very easy to backup the system, what we need do is just copy this sub-directory to anywhere your want.

We can uninstall previous version and install the new version in the same directory.Because the configuration files are kept in the ‘config’ directory, the new version will detect, use and upgrade previous configuration files automatically.

Q2: I am using V2.9, can I upgrade to V2.10 or above?

Yes, of course. By default, it is free to upgrade MSS from low version to higher version since they can share the same license. So just download the latest version and enjoy it.

Q3: I am using 20 clients version, can I upgrade to 50 clients and how?

Since different licenses are used, you need pay the price differentials between these version. Please visit our ‘buy now‘ page to get details.

Once you get the new license, you can uninstall previous version and install new version, then input new license to enable it. The upgrade process is almost same with above description.