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Asterisk and VoIP GSM gateway

VoiceBlue with Asterisk IP PBX How to

The artical is publiced originally at www.2n.cz.

The present document is a step-by-step guide for configuring the Ateus VoiceBlue VoIP GSM gateway and Asterisk software IP PBX. It includes essential steps for interconnecting the two products rather than a detailed configuration analysis of the two units. For more information on the Ateus VoiceBlue gateway configuration refer to the manual placed at http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html. For more information on the Asterisk PBX see www.asterisk.org.

What is Ateus VoiceBlue?
The ATEUS – VoiceBlue GSM gateway is an ideal complementary product to any SIP-based IP PBX. It’s suitable for small and medium companies with IP infrastructure and for companies with international affiliates. With VoiceBlue you gain significant savings on outgoing and incoming calls from IP to GSM networks and backwards. Thanks to the efficient and powerful Least Cost Router (LCR), the VoiceBlue GSM gateway always chooses the cheapest possible way to route the call (according to GSM prefixes, free minutes on SIM cards etc.).Using the VoiceBlue gateway you achieve complete independence when connecting to GSM networks and maintain 100% control of your GSM call costs. With the help of voice prompts and efficient Dynamic Clip Routing, VoiceBlue routes incoming GSM calls to the right IP phones.

What is Asterisk?
Asterisk is a software IP PBX originally developed by Mark Spencer. This software IP PBX is designed as an open source, which means that it can be downloaded from the Internet, namely from: http://www.asterisk.org/index.php?menu=download, installed and used without any limitations. The fact that the Asterisk is a software IP PBX indicates that VoIP is the native environment for this PBX. Asterisk supports VoIP using three protocols, which means that its users can use a wide range of VoIP telephones, both software ones and hardphones.
Together with the above mentioned VoIP services, Asterisk supports connection of TDM equipment by means of PCI cards supporting both digital (ISDN PRI/BRI) and analog trunks. Since the PCI bus on the PC is not unlimited, Asterisk users employ as much VoIP equipment as possible in order to save the TDM interface for other services, such as PSTN connection, or use of analog telephones. This is why most Asterisk users use the Ateus VoiceBlue VoIP GSM gateway for their GSM calls. You find below how to interconnect these two types of equipment.
For more information on the Asterisk software solution see www.asterisk.org

Main scenario

Suppose we have an IP network to which an Asterisk IP PBX, several SIP telephones and an Ateus VoiceBlue VoIP-GSM gateway are connected. This typical configuration is 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 6,7,8 and containing 9 digits. For configuration simplicity, use SIM cards from one GSM provider.
Now say that all incoming calls are answered by the gateway, which replays the invitation message and waits for 10s for another DTMF dialling. After this timeout, the gateway dials extension 111, which is a dial-in to the operator.

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