Asterisk
libresbc
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| Asterisk | libresbc | |
|---|---|---|
| 11 | 2 | |
| 1,944 | 325 | |
| 6.1% | - | |
| 9.3 | 9.0 | |
| 5 days ago | 2 months ago | |
| C | Python | |
| GNU General Public License v3.0 or later | MIT License |
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
- FreePBX – Open-Source PBX (PHP GUI for Asterisk)
- Incoming call not matching peer
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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.
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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 Ă empĂŞcher - Le Monde
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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
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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:
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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.
libresbc
- LibreSBC: An open-source Session Border Controller
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SBC
Here's a preconfigured kamailio+freeswitch distro that I found that looks pretty point-and-click. Maybe a better fit for your needs: https://github.com/hnimminh/libresbc
What are some alternatives?
Kamailio - Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
docker-freepbx - Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
freeswitch - FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.
Routr - ⚡ The future of programmable SIP servers.
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
chan-sccp - Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like.
SipXcom - Unified Communications System
sslsplit - Transparent SSL/TLS interception
Ostel