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BitVoice v0.1.2
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BitVoice v0.1.2 is the office phone system release: the features people expect from FreePBX and Asterisk, all set up in the web interface.
New
- Conference rooms: PIN and leader PIN, wait for the leader, members joining muted (press 1 to ask to speak), lock, recording, and a live view to mute or remove people.
- Forwarding and do not disturb from the phone:
*72/*73always,*90/*91when busy,*52/*53when not answered,*78/*79/*76do not disturb. - Follow-me (
*21): the desk first, then other numbers one after another or together; press 1 to take the call, so a mobile's voicemail never answers it. - Call pickup (
*8for a group,**and an extension for one phone), intercom with auto-answer (*80), and company and personal speed dials (*0). Every code can be changed on the new Feature codes page. - Staff portal at
#/me: people sign in with their extension and voicemail PIN to set do not disturb, forwarding, follow-me and speed dials, hear their voicemail, record their greeting and see their own calls. - Hosted organisations: many phone systems on one server, each with its own data, web address or domain and administrators, with limits on extensions, calls at once, storage and AI use, and suspension. Emergency calls are never refused.
- IAX2 trunks (RFC 5456), registering either way, tested against Asterisk 23.
- Codecs: G.726 (32 and 16 kbit/s) and GSM, converted to and from all the others.
- A new manual: 63 chapters in six parts with 63 diagrams, about 90 pages in the PDF.
Changed
- The 911 answering-point profile, the radio dispatch console and NG911 Core Services are now a lab and design-partner track: off the menu until Public-safety profiles (lab) is switched on in Settings, with a notice on their screens. A server where one is already set up keeps them in the menu. They are not certified and not yet tried with a live ESInet, P25 system or CAD. Emergency calls from your own phones are unaffected and always on.
- A test now fails if any setting has no screen in the web interface.
Fixed
- Windows: when the SIP port is chosen automatically and the first pick is reserved by Windows, BitVoice tries another.
- A race between two parts of a call when a transaction ended, found by the race detector.
Not in this release: telephony cards (DAHDI): analogue lines and ISDN connect through SIP gateways. iLBC, Speex and AMR codecs.
Each archive holds the program, LICENSE, NOTICE and the manual as a PDF. Check downloads against SHA256SUMS. Builds are not signed for the in-app updater yet.