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SIP trunk between MSSes

SIP trunk between MSSes

1. Description

Some customers have several office branches and hope to establish VoIP connections between them. There are several methods to do that. Here we give an example to describe how to establish voip connections between two MSSes with SIP trunk feature.

2. Network topology

The network topology is simple. There are two office branches. Please refer to below figure.

Demo network
Demo network

Before we setup voip network, it is better to assign extension numbers. Different office numbers will effect how to configure call routing in both MSSes. In above figure, we can see the extensions in office 1 are 1xx, and extension numbers in office 2 are 2xx.

Both MSSes are configured with public IP address. If your MSS is behind NAT or router and you want to provide connection for outsides users, please refer to another document firstly.

Now we give detail configurations for it.

3. Configuration

In below configurations, items should be kept their default values if we don’t configure them obviously.

3.1 MSS1

Please click menu “data – SIP trunk”, then add a record:

SIP trunk ID = 1
Description = to MSS2
Server address = 10.23.x.x

Please click menu “dial plan – analyzed called number” , then add a record for routing 2xx to such SIP trunk.

called number prefix = 2
route type = SIP trunk
SIP trunk ID = 1

3.2 MSS2

It is almost same with MSS1.

Please click menu “data – SIP trunk” to add a record.

SIP trunk ID = 1
Description = to MSS1
Server address = 41.32.x.x

Please click menu “dial plan – analyzed called number” for routing calls to above SIP trunk.

called number prefix = 1
route type = SIP trunk
SIP trunk ID = 1
Refine called number

Refine called number

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.