A recurring theme I’m seeing lately is people deploying VoIP, running into issues, and not approaching their issues from the perspective of taking all components into account. What I mean by this is that if you are deploying Asterisk it is only a single component, there are others. There is the Linux distribution it is running on, the infrastructure it is running on, the network connection, the switches, the endpoints. VoIP has a lot of moving parts and it is important to remember them all whe...
What Is A VoIP Gateway? ; A VoIP gateway is used to build a bridge between the worlds of legacy telephony and the VoIP. Gateways are typically used to connect legacy phone systems (PBXs or ACDs) with VoIP resources, or to connect modern VoIP phone systems with legacy phone lines. Adding VoIP to a legacy PBX system is a great way to add features and reduce costs. The gateway connects to the legacy system through either analog or digital trunk ports. The PBX sees the gateway as either the phone co...
The Asterisk Community has become the top influencer in VoIP with ambassadors and contributors from every corner of the globe. Leading the effort are the
Ready to Get Started with Asterisk? Asterisk is a free and open source framework for building communications applications and is sponsored by Sangoma.
Sometimes my voip provider shuts down service without ending calls which are in progress... I can only reset their state by restarting asterisk service. How do I set up asterisk to end this...
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Siperb is a WebRTC to SIP Proxy between your traditional VoIP PBX (like Asterisk) and a powerful WebRTC Browser Phone client.
VarPhonex is a service provider that offers Asterisk termination and high quality VoIP Phone Service at a reasonable price. Our Asterisk and Trixbox termination allows you to easily connect and get...
Asterisk VoIP Adapter. Contribute to schmupu/ioBroker.asterisk development by creating an account on GitHub.
burgerjaques ; I’m not an asterisk expert, and I’m stuck at this moment. I’m trying to setup an asterisk box with realtime. Most work, and my endpoints are able to make calls between each other. But I need to setup a SIP trunk to a VOIP provider, and I’m not sure how to do it, because what I’ve done does not work. The first problem is that my registration does not load. From what I understood the sorcery.conf file needs to be setup for the different objects. I determined that the type basically maps to the type in pjsip.conf, So I’v ...