Asterisk-examples
Asterisk
| Asterisk-examples | Asterisk | |
|---|---|---|
| 1 | 11 | |
| 2 | 1,963 | |
| - | 3.4% | |
| 6.4 | 9.3 | |
| 15 days ago | 5 days ago | |
| Shell | C | |
| - | GNU General Public License v3.0 or later |
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Asterisk-examples
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Softphone for SIP instant messaging
This is the code I've been using: https://github.com/almeyras/Asterisk-examples/tree/main/classroomPABXvideoandim
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 à empêcher - 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.
What are some alternatives?
docker-freepbx - Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
libresbc - An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.
Routr - ⚡ The future of programmable SIP servers.
Kamailio - Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
soup - ☎️ Original open source call flooder using Twilio's API.
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
ggsignif - Easily add significance brackets to your ggplots
Browser-Phone - A fully featured browser based WebRTC SIP phone for Asterisk
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
SipXcom - Unified Communications System
Ostel