https://lms.onnocenter.or.id/wiki/index.php?action=history&feed=atom&title=VoIP_Cookbook%3A_Trunk_Peering_in_Asterisk 2026-04-20T14:31:36Z Revision history for this page on the wiki MediaWiki 1.45.1 https://lms.onnocenter.or.id/wiki/index.php?title=VoIP_Cookbook:_Trunk_Peering_in_Asterisk&diff=17334&oldid=prev 2010-03-09T07:12:27Z

<p>New page: One of the main reasons why we use VoIP is to have free long distance or international calls. Imagine that if you&#039;re having a branch office or working partners who often communicate with y...</p> <p><b>New page</b></p><div>One of the main reasons why we use VoIP is to have free long distance or international calls. Imagine that if you&#039;re having a branch office or working partners who often communicate with you. You have to establish a private network between office branches or those working partners so you can bypass the PSTN. There are a number ways you can do this through Asterisk. <br /> <br /> * DUNDi, Distributed Universal Number Discovery protocol.<br /> * Centralized directory, such as VoIP Rakyat<br /> <br /> On this occasion, you will be shown a trunk peering process using VoIP Rakyat. The same mechanism can be applied to other SIP proxy across the world. <br /> <br /> In addition, we will also discuss the real problems we face in configuring network involving NAT/Proxy Server, as most networks are protected by firewall that blocks VoIP signal. <br /> <br /> We presume that we already have an account in VoIP Rakyat. In this sense, the given number and password are:<br /> <br /> number 2012345 password abcdef<br /> number 2055555 password 123456<br /> <br /> Next we will do a comprehensive configuration of file sip.conf and extensions.conf, including providing the facilities required for testing. <br /> <br /> In general, there are several important things in configuring trunk in Asterisk <br /> <br /> * Registration to SIP account in voiprakyat (sip.conf)<br /> * Creating username &amp; password for various extensions (sip.conf)<br /> * Configuring Dialout for a variety of configurations (extensions.conf)<br /> * Configuration for inbound call (extensions.conf [inbound-sip])<br /> <br /> With this configuration, we can now place outgoing calls using various available lines. In addition, we can also receive calls dialed from voiprakyat and the internet through inbound-sip module. The detail of each of a variety of configurations is available in the enclosed configuration.<br /> <br /> <br /> ==See Also==<br /> <br /> * [[VoIP Cookbook: Building your own Telecommunication Infrastructure]]</div>

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