Tag: asterisk
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SIP and NAT: Why is it a problem?
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Why is it a problem using SIP Clients behind NAT? What is NAT? To understand why SIP Clients behind NAT are a problem, you need to first have some understanding of what NAT is and what it does. NAT stands for Network Address Translation. Unless you are using One-to-one NAT, then a NAT device may also perform Port Address Translation (PAT). For a detailed explanation of NAT click here. If you want, you can also read…
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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 call settings shown in part 2. Caveat – this only refers to chan_sip and not pjsip. You will need a peer…
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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 trunk A SIP Trunk may be used for both inbound and outbound call traffic. To make outbound calls, your Asterisk, FreePBX…
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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 SIP Trunk? A SIP trunk uses standard SIP signalling, but the endpoints of the trunk are fixed – typically one end…
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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 two “legs” to that call. The first leg is from your phone to Asterisk and the second is from Asterisk to…
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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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High Availability and Failover options for SIP and Asterisk
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Overview What’s the disaster we are trying to avoid? The assumed scenario is this: Some kind of centralised VoIP service is being offered to a number of users; the service operates on servers located at a data centre or office and the users each have a SIP client device, such as an IP phone, that connects to the centralised service over the Internet or over the company network. That is the typical setup for an Internet…
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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 this: If client device A wants to be informed of changes to the status of device B, it sends a SUBSCRIBE…
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What is NAT?
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What is NAT? NAT stands for Network Address Translation. Many devices sold as Firewalls or Routers are actually combined Firewall, Router and NAT device in one box. NAT is the mechanism that allows you to have many PC’s on your LAN all connected to the Internet through a single external IP address. When one of the PC’s on the private side of the NAT device initiates a connection with a server on the Internet, then it…
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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 define the location of the device on the Internet or within the scope of the network infrastructure that is being used.…
