Accessing AllStarLink from a SIP telephone
I have been using a SHARI AllStarLink node for some time as a means to tinker and learn about AllStarLink (ASL) and to explore the wide web of nodes around the globe that are connected to this vast network. ASL (HamVoIP being the flavor of it on my node) is based on Asterisk, an open source PBX system that is capable of handling the most complex telephony tasks and could conceivably be used to replicate or even replace nearly all voice switching capabilities of the public switched telephone network. On the K6TZ repeater system, we have configured the repeater node to handle phone patch requests over VoIP, eliminating the need for a copper line to offer radio to telephone network connectivity. This means that members can key in a code followed by a phone number on the 446.400 repeater to initiate a phone call to any number in the USA. I had also heard that ASL could provide for direct access to radio linking through SIP connections, sort of a reverse autopatch, whereby one could dial into the node from a SIP telephone to access and link to nodes throughout the ASL network. In setting out to configure my own SHARI node for this, I ran across a few inconsistencies on how to accomplish this so I am writing up this article to document what worked for me. The first place I turned was the official ASL wiki documentation. It is helpful but somewhat incomplete in its explanation. Below I’ll outline the steps and try to explain them in a bit more detail, adding some additional configuration to support multiple SIP extensions. The first step is to locate and edit the /etc/asterisk/modules.conf file. I am including the full path to all files since it can be challenging for newbies and experienced users alike to find them, especially when tinkering in ASL is not a daily task. We must edit the modules.conf to be sure the SIP module is loaded by ensuring that the following line is present: load => chan_sip.so ; Session Initiation Protocol (SIP) You may see this line in your file beginning with noload. Simple change it to load and save the file. Next locate the /etc/asterisk/sip.conf file and open it in your favorite editor. In this file we will append the following code to add one new SIP extension to our node. Note that I am calling… Continue reading