Asterisk
The official Asterisk Project repository. (by asterisk)
Homer
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring (by sipcapture)
Our great sponsors
| Asterisk | Homer | |
|---|---|---|
| 11 | 4 | |
| 1,944 | 1,511 | |
| 6.1% | 2.5% | |
| 9.3 | 7.5 | |
| 5 days ago | 14 days ago | |
| C | Shell | |
| GNU General Public License v3.0 or later | GNU Affero General Public License v3.0 |
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.
Homer
Posts with mentions or reviews of Homer.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2021-04-08.
-
[Issabel] Recommendation for a Good monitoring system for Issabel
You can use something like Homer SIP Capture to gather all sip information and metrics, including jitter, it is a little overwhelming in the beginning but it is really easy to set up and use. You set up the containers and send traffic from the asterisk using heplify and you are good to go. GitHub repo: https://github.com/sipcapture/homer
-
Solarwinds Voip Monitor and Nextiva
Also https://sipcapture.org/. Although VoIP monitor is a more proven enterprise solution
-
Homer - sipcapture.org
Does anyone know of a subreddit that deals with Homer specifically? or a mailing list?
-
VoIP Monitor
homer
What are some alternatives?
When comparing Asterisk and Homer 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.
sngrep - Ncurses SIP Messages flow viewer
Kamailio - Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Routr - ⚡ The future of programmable SIP servers.
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
webrtc - A pure Rust implementation of WebRTC
SipXcom - Unified Communications System
OMD - OMD - Open Monitoring Distribution Labs Edition. Der deutschsprachige Open-Source-Monitoring-Workshop findet am 16./17.5.2024 in Neckarsulm statt. https://discord.gg/jDfPZ63FcJ
Ostel
Tapir