Asterisk
The official Asterisk Project repository. (by asterisk)
Routr
⚡ The future of programmable SIP servers. (by fonoster)
Our great sponsors
| Asterisk | Routr | |
|---|---|---|
| 11 | 12 | |
| 1,944 | 1,328 | |
| 6.1% | 2.6% | |
| 9.3 | 9.6 | |
| 5 days ago | 5 days ago | |
| C | TypeScript | |
| GNU General Public License v3.0 or later | MIT License |
The number of mentions indicates the total number of mentions that we've tracked plus the number of user suggested alternatives.
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
Asterisk
Posts with mentions or reviews of Asterisk.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2024-03-11.
- FreePBX – Open-Source PBX (PHP GUI for Asterisk)
- Incoming call not matching peer
-
Serial connection to TI Silent 707
More useful is that it confirms that the 707 complies with the Bell v.21 standard. This is only 300 baud. However that makes a phone line very easy to emulate within the popular, open source PBX software Asterisk.
-
My collection of Ansible roles for self-hosting everything with Rocky Linux and FreeIPA
Asterisk VOIP PBX
- La fraude au faux conseiller, nouvelle arnaque bancaire difficile à empêcher - Le Monde
-
Hi! I’m working on a PBX in Asterisk and I need to get the CDR with an specific group of columns, like call ID, source, destination, date of the call and time of the call. The problem is that I can’t separate the variable “start”, that contains date and time, in two variables. Can someone help me?
[a] https://github.com/asterisk/asterisk/blob/master/configs/samples/cdr_custom.conf.sample
- The project with a single 11,000-line code file
-
Help deciphering systemd service status
EDIT: Given that the example asterisk unit file uses a "notify" type service, which is just an exec with bells and whistles, I'm leaning towards:
-
ARI: Trying to play a sound file from URI
In this case, that registered provider is res_http_media_cache which handles http and https. Since there's no provider today that handles file, those error out.
Routr
Posts with mentions or reviews of Routr.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2024-03-04.
-
Building a VoIP Network with Routr on DigitalOcean Kubernetes: Part I
Please see the Official Chart for many more options for your deployment.
-
Browser-to-Browser calling with SIP.js and Routr
Routr SIP Server
- Routr: The future of programmable SIP servers
- The future of programmable SIP servers (v2.1.2 release)
- Latest version of Routr comes with support for ephemeral agents
- Routr (v2): The future of programmable SIP Servers
-
Routr (v2): The Future of Programmable SIP Servers
And that's just the start - there are even more features and improvements in this release! You can find the full details on the GitHub repo.
-
Routr (v2) Beta is Here - Bringing Exciting New Features and Improvements!
👋 Hi, everyone, maintainer here. I'm thrilled to announce that Routr(v2) beta is now available, and it comes with some amazing features that bring us closer to matching Twilio's Elastic SIP Trunking. Routr (v2) is built from scratch, incorporating all the lessons we've learned from deploying v1 in cloud environments over the past few years. TLDR; → Routr(v2) at GitHub Here are some of the key features of Routr (v2) that we're excited about: ☑️ Common SIP Server functions; Proxy, Registrar, Location Service ☑️ Multi-Domain with Domain level Access Control List ☑️ Load balancing strategies against Media Servers like Asterisk and FreeSWITCH ☑️ Programmable Routing ☑️ In-memory and Redis Location Service ☑️ Server management with a gRPC API ☑️ Helm Chart for Kubernetes Deployments And that's just the start - there are even more features and improvements in this release! You can find the full details on the GitHub repo. If you work with SIP, we would love for you to try out Routr and share your valuable feedback. Together, we can make Routr even better 🙏
- Routr 1.0.0-RC6 is out for testing and will probably be the last release candidate
What are some alternatives?
When comparing Asterisk and Routr you can also consider the following projects:
libresbc - An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.
Kamailio - Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
docker-freepbx - Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
SipXcom - Unified Communications System
Ostel