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Howto Asterisk Voicebluenext3

The document provides instructions for configuring a connection between a 2N VoiceBlue Next gateway and an Asterisk PBX via SIP trunk. It describes the network setup and parameters for each device, and provides steps for configuring the SIP proxy, LCR, GSM groups, and inbound/outbound call routing in both the gateway and Asterisk.

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Cesar Agustin Colin Guadarrama
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130 views

Howto Asterisk Voicebluenext3

The document provides instructions for configuring a connection between a 2N VoiceBlue Next gateway and an Asterisk PBX via SIP trunk. It describes the network setup and parameters for each device, and provides steps for configuring the SIP proxy, LCR, GSM groups, and inbound/outbound call routing in both the gateway and Asterisk.

Uploaded by

Cesar Agustin Colin Guadarrama
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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2N

VoiceBlue Next

2N VoiceBlue Next & Asterisk


connected via SIP trunk
Quick guide
www.2n.cz

Version 3.00

2N VoiceBlue Next has these parameters:

IP address 10.0.0.20

Incoming port: 5060

Asterisk parameters:

IP address 10.0.0.10

Incoming port: 5060

Scenario
If we have an IP network in which an Asterisk PBX, several SIP phones and 2N VoiceBlue
Next are connected, the configuration would be as shown in the figure below. Furthermore,
suppose that the network is addressed as shown in the figure and GSM numbers are all
numbers starting with 6, 7 and containing 9 digits.

SIP TRUNK INTERCONNECTION


1) For the setting of the trunk between the VoiceBlue Next and your Asterisk PBX, you need to
configure SIP proxy (GSMIP) for GSM incoming calls. SIP proxy (IPGSM) is designed
only for secure communication with the traffic from your Asterisk. You can specify the IP
address and port where the IP packets will be accepted.
If you leave there 0.0.0.0, the traffic will be unsecured.
To enable incoming calls to Asterisk, you can register the 2N VoiceBlue Next directly into
the Asterisk system. You can register it as Friend types in case you require registration on
based on username and password or peer type (on based of IP address and port).
-

SIP registrar...an Asterisk IP address which registers the gateway


Registration domain IP address where the gateway is going to be registered
Username...username under which the gateway shall be registered
Password...registration password

The IP address
where the traffic
is sent

The IP address and


port which the traffic
will come from

2) Configuration of the LCR (Least Cost Routing)


You have to specify prefixes for the operators in the country you are currently located. An
example of this would be that in Czech Republic prefix 6 and 7 have a 9 digits number.
The setting is displayed below.

3) You need to create specific guidelines connecting prefixes with the GSM group. In the
GSM group you will specify settings for SIM cards assigned to this specific group. In the
GSM group assignment you can assign the module for the appropriate GSM outgoing
group.

4) Configuration of GSM outgoing groups:


You are able to have different setting for each GSM group (CLIR, free minutes, Virtual ring
tone, roaming and others)

5) Incoming calls
For incoming calls you can define 2 groups with the different behaviors and assign them to
the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.

In GSM incoming groups you can specify the traits for each GSM incoming group. Choose the
mode to Reject, Ignore, Accept incoming calls or Callback.

You can define the list of numbers called. The number will be automatically dialed after the DTMF
dialing has timed out. This happens when the customer doesnt press any button until the specific
time. At this point, the number will be routed to the extension 100 to your Asterisk (if you set up
SIP proxy (GSM->IP) in VoIP parameters).

ASTERISK SETTING
Now add a few lines into the Asterisk configuration for proper routing of outgoing calls to the
2N VoiceBlue Next gateway and receiving calls coming from the GSM gateway to Asterisk.

1) Outgoing calls
The core of Asterisk connection is saved in the /etc/asterisk/extensions.conf file.
Open this file in your favorite editor and add the following lines:
exten=>_6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r)
exten=>_7XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r)
Once you have saved and closed the file, restart Asterisk. From this point forward, all calls
starting with 6 and 7 should be routed to the 2N VoiceBlue Next gateway.

2) Incoming calls
It is highly recommended to make a little restrictions for incoming calls to prevent
unauthorized people from calling over your system.
Since the 2N VoiceBlue Next system works with the SIP, modify the /etc/asterisk/sip.conf
file where the 2N VoiceBlue Next section could looks as follows:
[general]
port = 5060
bindaddr = 0.0.0.0
allowgues=no
context = sip
disallow=all
allow=ulaw
[VoiceBlueNext]
type=peer
host=10.0.0.20
username=voiceblue
secret=password
fromdomain=10.0.0.20

Again, restart the Asterisk after saving the file. Then the Asterisk will be ready to receive calls
coming from the 2N VoiceBlue Next gateway.

What to do in case of trouble:


First of all, check our webpage faq.2n.cz and try to see if there is a solution to your problem.
In case, you cannot find the proper answer, use the link: How to report an issue on the 2N
VoiceBlue Next.
Here is the direct link:
https://jira.2n.cz/confluence/pages/viewpage.action?pageId=22513331

2N TELEKOMUNIKACE a.s.
Modansk 621, 143 01 Praha 4
tel.: 261 301 111, fax: 261 301 999,
e-mail: [email protected]
www.2n.cz

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