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In this new version, following features are updated:
1. For FXO external lines, the max simultaneous call is limited to 1 by default.
2. The install package is updated to remove previous installed version automatically when installing the new version in the same directory.
MSS V4.0 is coming….It is a very important version for our customers. The most important feature of this version is to support Linux!
Yes! It is true! MSS V4.0 can run on Ubuntu/Kubuntu system without WINE now! MSS will be a cross-platform SIP|VOIP server.
The development work is finished and we are still working on system test. We hope to release this version in the end of this month or the beginning of next month.
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