similar to: Re: Firefly 1.9.5 released (gARetH baBB)

On Mar 21, 2024, at 9:02?AM, gARetH baBB <hick.icecast at gink.org> wrote: > Or you could just use ffmpeg: Since we?re discussing HLS encoder options, there is also: https://github.com/ElvishArtisan/GlassCoder Which does not require LiquidSoap, ffmpeg or even Icecast. Just aim it at a web server or CDN publishing point. Cheers!

On Thu, 17 Feb 2005 11:22:49 +0000 (UTC), gARetH baBB <hick.icecast@gink.org> wrote: > On Thu, 17 Feb 2005, Michael Smith wrote: > > > There's one thing that icecast doesn't do (except for ogg streams) > > that other servers might (but from what I'm told, shoutcast doesn't): > > start sending from some valid synchronisation point (e.g. the mpeg >

Just a quick announcement for Firefly users that Firefly 1.9.5 is out. Mainly just a bug fix release as we get ready for Firefly 2.0. One notable feature added is DTMF via SIP INFO. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe is the URL As always, send me any bugs, features or suggestions. -Adam

Using Firefly 1.9.5 (thirdparty) on Win2k Using Asterisk CVS HEAD 20041117 (also tried with 20040806 and 200410-something) IAX2, no NAT. Firefly->Asterisk audio works, but I can't hear anything from the other side. Using GSM codec, also tried ulaw. Any ideas? -A. relevant bits of iax.conf: [andrew-bt] type=peer host=dynamic trunk=no [andrew-bt] type=user context=fxs secret=12345

On Thu, 21 Mar 2024, Wayne Barron wrote: > Looking through HLS, I found this. > https://github.com/mbugeia/srt2hls Or you could just use ffmpeg: ffmpeg -hide_banner -i "${icecaststream}" -c:a copy -vn -strftime 1 -f hls -hls_time 6 -hls_list_size 10 -hls_segment_filename "${hlspath}radio-%Y%m%d-%s.ts" -hls_flags delete_segments -segment_format mpegts

Tried with http://zhongshuobeijing.dyndns.org:7000/streamname.m3u but was blocked. And at last resolve: http://139.168.32.224:7000/streamname.m3u is that what you mean? no, perhaps you mean the localhost? "bicho" is its name. Pardon me if i'm confused on this. Will also, tomorrow, try your suggestion of port 443 and Ross' suggestion of the web port. cheers, iain On Wed,

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

On Wed, 2005-05-11 at 10:37 +0100, gARetH baBB wrote: > On Wed, 11 May 2005, Iain Mott wrote: > > > Tried with http://zhongshuobeijing.dyndns.org:7000/streamname.m3u > > but was blocked. And at last resolve: > > http://139.168.32.224:7000/streamname.m3u > > Well, 40 minutes later that host is certainly not responding. > good morning No, I shut it down at the

This is release 1.0-rc4 of Wine, a free implementation of Windows on Unix. What's new in this release (see below for details): - Bug fixes only, we are in code freeze. Because of lags created by using mirrors, this message may reach you before the release is available at the public sites. The sources will be available from the following locations:

Hello all: I have an annoying problem with smbmount; My Win2k server has many folders and files named with accented characters (cp850 and/or cp860 or European w/0 euro or Portugese). My mount command is: mount -t smbfs -o username=leo,codepage=cp850 //<servername>/d$ /mnt/pt to mount my d: drive at /mnt/pt I am prompted for my password and the shared folder is mounted. I can see all

Karl Heyes wrote: > On Tue, 2005-04-12 at 14:44, gARetH baBB wrote: > >>How is fallback to file actually meant to work ? > > > just state a filename within webroot. The server creates a source thread > so that it interacts with the existing fallback mechanism. No timing > goes on with the data stream, and usual rules for fallbacks apply, ie > same format and same

Hi, LLVMers, has someone implemented bitwidth analysis for LLVM? I was looking for something similar to the bitwise compiler described in "Bidwidth analysis with application to silicon compilation, by Mark Stephenson, Jonathan Babb and Saman Amarasinghe" e.g.: http://portal.acm.org/citation.cfm?id=349299.349317 all the best, Fernando

Hi All BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with

I did get it to work, and can place and receive calls through the Firefly network via *. Compared to iaxtel or FWD, there is a significantly higher amount of latency, but it is workable. For some reason, this needed to be the last entry in my iax.conf or it would try to authenticate with a different user ID when receiving calls (and obviously would fail. Relevant section from my iax.conf:

I found version 2.0.0 of Firefly on the Freshtel site, but it only has the network setup options for the Freshtel network, despite the final statement on the page http://www.freshtel.net/firefly/download/ that says: ----------------- Standalone SIP / IAX mode: If you want to use Firefly on our network (with your own voicemail etc.) you will need to register a Firefly number. However, you can

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hi wxRuby 1.9.5 is now out with a full set of binary gems. It can be downloaded from the usual place: http://rubyforge.org/frs/?group_id=35&release_id=19932 Or via rubygems: gem install wxruby == WHAT''S NEW == This doesn''t have any major new features, but has a lot of bugfixes, including some nasty crashers, and additional useful methods within existing classes. It

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

There's a new firefly release out for those who are using firefly with your lovely asterisk / SIP server. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe the main changes are improved GUI fixes (mouse wheel works now :) ), few url parsing fixes, mic volume control and improved compatibility with SIP servers (namely SER). Send me all bugs, problems and suggestions (even

hello all... newbie question: I have FireFly setup on my laptop and I would like to test this out using IAX in this scenario: FireFly Softphone > Asterisk > TDM Gateway i do not wish to use this on the firefly network, but simply within my own "3rd party" network as the website and setup of FireFly defines it... does anyone have a sample iax.conf and extensions.conf i