Category: Archive
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Installing OpenSIPS v1.6.1 on CentOS 5 64bit
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Smartvox Limited offered consultancy for installation and configuration of OpenSIPS, Asterisk and PSTN gateways. If you want to do it all yourself, here is a set of instructions that should help. Installing OpenSIPS v1.6.1 on CentOS 5 64 bit is very straightforward. All the dependencies can be installed using YUM. The packages required will depend […]
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Asterisk behind NAT
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Scenarios in which NAT may adversely affect Asterisk SIP connections The Asterisk Server is behind NAT The Asterisk server could be on the LAN (or in a DMZ) with a NAT firewall between it and the Internet. When it communicates with external peers or devices, the network connections have to pass through the local NAT […]
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Taking the plunge with SIP Trunks – Part 3
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Outbound calls; Matching a DDI/DID; Diagnosing problems; Internet bandwidth. Configuration for Outbound calling Part 2 looked only at the configuration for receiving inbound calls, but the SIP Trunk configuration form in Trixbox/FreePBX has to also include the settings for making outbound calls. It may not even work at all if you only use the inbound […]
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Taking the plunge with SIP Trunks – Part 2
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Recap In part 1, I explained how a SIP Trunk is really just a virtual connection between your IP-PBX and the VoIP service provider. Standard SIP signalling is used on the trunk, but more than one simultaneous call is allowed and you may have more than one DID number. Basic requirements to enable your SIP […]
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Taking the plunge with SIP Trunks – Part 1
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Many VoIP service providers offer SIP Trunks as a standard product, but just what is a SIP trunk and what can you use it for and how do you make it work with Asterisk or FreePBX? I will attempt to answer these questions in this 3 part blog, starting with the basics. What is a […]
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RTP pass-through mode in Asterisk
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What is RTP pass-through mode in Asterisk? The speech for VoIP calls uses RTP (Real Time Protocol) to get from one end to the other and it is compressed using one of the many speech compression codecs available. Commonly used codecs include G.711 and G.729. When you call someone through an Asterisk PBX, there will be […]
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How can an Asterisk IP-PBX benefit my business?
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In a Nutshell: The benefits of upgrading to an Asterisk based IP-PBX
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SIP Subscribe/Notify and Asterisk Hints Explained
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The SIP SUBSCRIBE/NOTIFY mechanism – what it is and how it works The SIP protocol includes a standardised mechanism to allow any SIP client (an IP phone being an example of a SIP client) to monitor the state of another device. Details are provided in the SIP protocol document RFC 3265. Basically, it works like […]
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Configuring and Using SIP Domains in Asterisk
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What are SIP Domains? SIP requests delivered to a Proxy server or other SIP device must contain details of valid destinations and end-points that are to be reached or which are to receive responses. The address of a SIP device is generally referred to as its URI (Uniform Resource Identifier) – it has to uniquely […]
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Asterisk SLA (Shared Line Appearances) – Part 4
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Link to part 3 What is good and what is bad about the Asterisk implementation of SLA? Benefits of Asterisk Shared Line Appearances Replacement of legacy systems: The main reason for wanting to configure Shared Line Appearances in Asterisk is because the Asterisk system is replacing a traditional key and lamp telephone system. However, this […]
