baresip
Baresip is a modular SIP User-Agent with audio and video support (by baresip)
Asterisk
The official Asterisk Project repository. (by asterisk)
| baresip | Asterisk | |
|---|---|---|
| 7 | 11 | |
| 1,623 | 1,948 | |
| 1.0% | 2.6% | |
| 9.4 | 9.3 | |
| 1 day ago | 3 days ago | |
| C | C | |
| BSD 3-clause "New" or "Revised" License | GNU General Public License v3.0 or later |
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.
baresip
Posts with mentions or reviews of baresip.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2023-09-02.
- Baresip – An Open Source modular SIP User-Agent with audio and video support
-
Connecting linphone softphone
I found this info on Reals on Linphone but could not get it to work with the strings there either: https://github.com/baresip/baresip/issues/383
- Decent linux softphone w/command-line or dbus?
- High bit rate recording sometimes causes this (I haven't noticed this with AVC encoding, only HEVC), is this normal? GPU: RX 590 / 22.3.1 drivers
- Not able to play audio from a voip daemon program that runs at boot.
-
20 Years of Sip – A Retrospective
Happy SIP user for nearly twenty years, which allows me to bridge three countries. Currently using baresip [1] and finding it to be remarkably reliable, but is there any hardware phone out there that I can put on my desk? Or is the sane thing to do to get a handset and hook it up to a computer via say USB? I have tried at least twice over the years to gain some clarity on these questions, but maybe I am using the wrong search terms?
[1]: https://github.com/baresip/baresip
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.
What are some alternatives?
When comparing baresip and Asterisk you can also consider the following projects:
tSIP - SIP softphone
libresbc - An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.
pjproject - PJSIP project
Kamailio - Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
waterfall - A cascading stream forwarding unit for scalable, distributed voice and video conferencing over Matrix
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Routr - ⚡ The future of programmable SIP servers.
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
SipXcom - Unified Communications System
Ostel
Tapir
EVE-IPH - Code for the EVE Isk per Hour program