There are several calling scenarios - typical Class V - where multiple SIP dialogs may be involved. And to make it work, you need, from one dialog, to access the data that belongs to another dialog. By data we mean here dialog specific data, like dialog variables, profiles or flags, and, even more, accounting data … Continue reading Cross-dialog data accessing
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As we get closer to the much-anticipated release of OpenSIPS 3.1 LTS, a new major feature has been merged into the master branch: full support for standards-based SIP Push Notifications (short: "SIP PN"), a.k.a. RFC 8599! In this two-part series, I'm going to present two sides of the story: first, the IETF document itself: the … Continue reading SIP Push Notification with OpenSIPS 3.1 LTS [RFC 8599 support][Part I]
In advanced Class 5 SIP systems, the "intelligence" is not a characteristic reserved 100% only to the SIP Server (or Application Server component). In such system, the "intelligence" - meaning the ability to control and operate the Class 5 specific features - is actually shared between all the SIP phones and the SIP Server. This … Continue reading DFKS or the “Key to Synchronization”
The Call Center module in OpenSIPS proved to be a great asset for the Class 5 ecosystem. This module provides a powerful and flexible call queuing and distribution engine. Still we cannot say it is an 100% Call Center solution as the module does not provide any support for the agent side, but only for … Continue reading Call Center – an easier integration with OpenSIPS 3.1
While there are numerous external, open-source rating and billing engines available in the wild (e.g. CGRateS, ASTPP), having a quick and easy way of putting a price for a call, without relying on external applications, can be a valuable asset to have. With the addition of the rate_cacher module, OpenSIPS 3.1 gains the ability of caching … Continue reading Real-Time Rating and Cost Based Routing in OpenSIPS 3.1
The BLF support (or dialog-info) is a highly useful feature in Class 5 switches (or PBXes). BLF is more than a lit led on your phone, letting you know if a party is engaged in a call or not. The BLF, as a mechanism of sharing the status of the parties and their calls, is … Continue reading BLF reloaded, or a more accurate and detailed approach
SIP proxies, by definition, lack the ability to do any media handling, due to the fact that they only stand in the path of call signalling, not call media. Therefore if you want to enhance your SIP proxy with any extra media capabilities, you have to inject a new component in the media path, that … Continue reading Enhanced media capabilities in OpenSIPS 3.1
Overview This post is a sequel to the initial write-up on the recently introduced qrouting (Quality-based Routing) module. During the last month, the module has received several key additions, aimed at both improving the data format (gateway statistics, thresholds and scores) as well as the runtime behavior, with a new traffic balancing algorithm having been … Continue reading Quality-based PSTN Routing in OpenSIPS 3.1 LTS [part 2]
DTMF (acronym of dual tone multi frequency) is the signal that your phone sends to your carrier when you press your phone's touch keys during an ongoing call. The primary usage of DTMF tones is to control IVR (Interactive Voice Response) applications, by using the tones to choose certain options from a list. From a … Continue reading Call control using DTMF in OpenSIPS 3.1 LTS
Introduction Note: some information in this post may be outdated. Make sure to also read the follow-up post for the final view on qrouting's behavior. Up until today, the open-source SIP server garden seems to have yielded quite a handful of ways to perform PSTN routing. Some VoIP operators prefer to pull the latest rate … Continue reading Quality-based PSTN Routing in OpenSIPS 3.1 LTS [part 1]
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