Inside the Asterisk

Keyword: Configuration

STIR/SHAKEN in Asterisk

What is STIR/SHAKEN? STIR/SHAKEN (Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs) is a new technology that the telecommunications industry is using to help combat telephony fraud. We’ve all received spam calls, and some of us may have even received a call from a caller ID we recognized, but ended up

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Common SIP URI Issues

When transitioning from the chan_sip channel driver to chan_pjsip one of the items that can catch people off guard is the use of SIP URIs within PJSIP. This is because in chan_sip these are generated on your behalf based on different configuration options while in chan_pjsip we leave this up to the user. Let’s take

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SIP and RTP Routing

One of the most common issues I see when people deploy SIP is calls hanging up after approximately 30 seconds or traffic not going to where it should. This can be hard for users to grasp and is primarily due to the fact that SIP embeds routing information (IP addresses and ports) within the signaling

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Announcing a New Compile Option for app_voicemail Storage.

Asterisk offers its users great flexibility in most of its features. One of them is the choice between three different modules for different ways to store voicemail. These include: To a file  (app_voicemail – default) To an ODBC database (app_voicemail_odbc) To IMAP (app_voicemail_imap) The good news for Asterisk administrators is starting with version 17, we

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PJSIP Configuration Design

A comment that I see frequently when helping people with PJSIP is the lack of a general section (with global options) and how this causes their configuration to be larger than it needs to be. I thought I would take this blog post to explain some of the design choices that went into PJSIP configuration

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Improved PJSIP Qualify Support Performance

One of the most difficult things in PJSIP is ensuring that the experience is the best it can be for not just people who configure their Asterisk from normal configuration files but also from a database. This presents quite a challenge and one of the areas that has been problematic has been qualify support. Qualify

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Receiver Estimated Maximum Bitrate Support

For the last few months I, along with Ben Ford, have been working on improving the user experience side of the WebRTC support in Asterisk. When one thinks of user experience the first thing that comes to mind is usually a user interface but in this context I’m referring to underlying technology. Ben has been

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Asterisk Call Party, Privacy, and Header Presentation

Asterisk allows users to manipulate call party identification information through mechanisms like configuration options and dialplan functions (for instance CALLERID and CONNECTEDLINE to name a couple). This grants the user freedom to adjust values with regards to what call/caller information to expose and/or override. With this freedom, though, comes some complexity, and confusion. Especially when

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WANTED: Dead or Alive!

The Story of Asterisk and Keep-Alives The vast majority of VoIP communications is done via UDP datagrams.  It’s a no-overhead protocol which makes it fast and although it also makes it unreliable, the SIP and RTP protocols and our own ears and eyes can tolerate a certain amount of packet loss quite easily.   From a

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Upcoming WebRTC Improvements in Asterisk 15

In my previous post I talked about what WebRTC support is like in Asterisk 14. Since Asterisk 15 is going to be released soon let’s take a look at how WebRTC support differs in it from Asterisk 14. The “webrtc” PJSIP Configuration Option As the WebRTC specification has evolved and changed the functionality in Asterisk

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