chan-sccp
Asterisk
chan-sccp | Asterisk | |
---|---|---|
3 | 11 | |
172 | 1,963 | |
1.2% | 3.4% | |
5.8 | 9.3 | |
about 1 month ago | 1 day ago | |
C | C | |
GNU General Public License v3.0 or later | GNU General Public License v3.0 or later |
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.
chan-sccp
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new to voip, need help
https://github.com/chan-sccp/chan-sccp should get you started. Follow the docs and their wiki also some good info. You will get base config files and what not to prep your tftp server. Just be forewarned its not a trivial task to get everything setup, I have a dedictaed voice vlan with special entries for DHCP options for tftp and ntp. Also be warned that the sccp phones are basically not secure, if all that doesn't scare you cool, so long as they are just on your local LAN and asterisk is secured properly it really doesn't matter. Also check out this here if your interested in running newer SIP cisco phones as well and want to keep all the UCM features.
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Anyone getting any use out of old Cisco IP phones (79xx) in 2022?
I've got them running SCCP firmware, connected to FreePBX with chan_sccp and managed with SCCP Manager.
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Cisco 7962G phones with FPBX and chan_sccp
I will soon, but if you want something right now their guide is pretty good: https://github.com/chan-sccp/chan-sccp/wiki/FreePBX_Installation
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?
sccp_manager - SCCP Manager
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 -
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.
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Routr - ⚡ The future of programmable SIP servers.
MissedIT - Fully Featured hack Always Free As Feedom
ASTPP - Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org
harvey - A distributed operating system
SipXcom - Unified Communications System