docker-asterisk
Browser-Phone
| docker-asterisk | Browser-Phone | |
|---|---|---|
| 4 | 4 | |
| 143 | 433 | |
| - | - | |
| 4.7 | 4.8 | |
| 9 days ago | 16 days ago | |
| PHP | JavaScript | |
| MIT License | GNU Affero General Public License v3.0 |
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docker-asterisk
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Asterisk server for home intercom - voip/SIP phones
I'm also not sure how feature rich the Asterisk server needs to be. There are small lightweight docker images, as well as more feature rich images. I'm not sure how my usecase translate to necessary features.
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Audio does not pass (From Trunked Calls) until DTMF tone is sent.
Latest 18, through this docker container: https://hub.docker.com/r/mlan/asterisk
- PBX on Synology
- Are there any open source options for an analog version of "asterix"
Browser-Phone
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Building a Personal VoIP System
Which clients do you use? And how do you connect to a SIP trunk / DID without Asterisk?
I use WebRTC with Asterisk, and Browser Phone for the client (https://github.com/InnovateAsterisk/Browser-Phone). I don't use it much, but good enough for the rare times I have to use the phone.
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Create a javascript client over Astersik's webrtc feature
Also have a look on this browser phone written using sipjs library for any help - https://github.com/InnovateAsterisk/Browser-Phone
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Chat solution with SIP integration
If you want to go webrtc route you can check https://github.com/InnovateAsterisk/Browser-Phone which works with FS too.
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Cant Connect WebRTC from Issabel to External SIPPhone.
with this SIPPhone in WebRTC( https://github.com/InnovateAsterisk/Browser-Phone.git)
What are some alternatives?
docker-freepbx - Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
SIPSorcery - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
SIP.js - A simple, intuitive, and powerful JavaScript signaling library
PHP SIP Parsing/Rendering Library - :phone: SIP Parsing/Rendering Library for PHP
mirotalksfu - 🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
MadelineProto - Async PHP client API for the telegram MTProto protocol
Asterisk-examples - PABX examples used in "Fixed and Mobile Telephone Systems" VET classes.
sccp_manager - SCCP Manager
slipstream - NAT Slipstreaming allows an attacker to remotely access any TCP/UDP services bound to a victim machine, bypassing the victim’s NAT/firewall, just by anyone on the victim's network visiting a website
sccp_manager - SCCP Manager
OpenQueueMetricsAddOns - A set of community-contributed scripts for QueueMetrics