search for: sirens

I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see silk listed in menuselect as a codec. But I also don't see an asterisk 11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk. Do we use the asterisk 10 codec_silk.so ? sean

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

Sorry these haven't been regularly scheduled :) --------- <p><p>Ogg Traffic for Wednesday, December 12, 2001 Jack Moffitt <[1]jack@xiph.org> _________________________________________________________________ Table of Contents 1. [2]Support The Ogg Project 2. [3]Status Updates 2.1. [4]Monty 2.2. [5]Stan Seibert 2.3. [6]Jack

What is the proper version of the Siren7 codec to use for Asterisk 13.5.0? Since there's nothing later, does the version for 12.0 work?

> A Siren codec is not currently available and the one for 12 will not > work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why).

When I activate the equalizer in WinAmp, QCD, and Sonic Foundry SIREN Jukebox, I get a funny "washed out" sound to it. Vorbis sounds fine without it, but I miss the extra bass and what-not. I was wondering why this happens? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

Anyone have a status update on when Sonic Foundry plans on supporting RC2 ogg files? A while back on this list someone said they had a good contact at SF and was supposed to chekc up on it. I have contacted Ahead Software about including Vorbis support with Nero and they asked that a developer contact them about including it. Any chance a developer can contact them about support for Vorbis

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >> codec_silk for Asterisk 12 will most

Another crash with a packet: $10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640, mallocd = 1, mallocd_hdr_len = 324, offset = 64, src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318, uint32 = 2156475160, pad = "\030\063\211\200\000\000\000"}, delivery = { tv_sec =

> All patches need to go into JIRA with a license agreement to be > accepted. Understood, but I was using it as an illustration. Note, however, that, from a legal perspective, a patch such as this has no protectable IP (you can't copyright the only way of doing something) and the GNU projects have a formal rule that sufficiently-small patches need no assignments for that reason, which

Hi all, I've included copies of the relavent files below. I'm having a bit of a problem with Samba in that when I try to access files on its shared resources, I can browse to the shares fine, but the moment I try to open a file, the client's file browser hangs. I've verified that this is a problem on both my Mac (OS X 10.3.4), my Windows XP SP2 and WIndows XP SP1 machines. I'm

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

On Sun, 30 Jan 2005 14:39:18 -0800 (PST) Josh Coalson <xflac@yahoo.com> wrote: > --- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > > On Sun, 30 Jan 2005 10:45:05 -0800 (PST) > > Josh Coalson <xflac@yahoo.com> wrote: > > > > > OK, thanks, checked in. once linux-ppc is ironed out the > > > release should be ready to go. > >

I'm trying to enable offline auth in a Ubuntu 22.04 box, following: https://wiki.samba.org/index.php/PAM_Offline_Authentication using standard ubuntu samba package (4.15.13+dfsg-0ubuntu1.1). I've enabled workaround 'lock directory = /var/cache/samba'. Still does not work, and behave very badly with no conectivity or bad connectivity (wireless): sometime the

Hello folks, for a customer of us we are trying to make asterisk and windows RTC library work together, but without success. We *need* to use gsm codec, so in the "peer" section we have disallow=all allow=gsm the sip signaling is ok, and when sniffing we got this session description: INVITE from windows RTC: v=0. o=- 0 0 IN IP4 172.31.9.130. s=session. c=IN IP4 172.31.9.130.

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Hi there, I want to write helper functions for a base class, which will be used by its subclasses in the S4 world. This function ___will___ update certain slots of its argument object. Please help me decide which one of the following is a better approach with respect to coding style, memory usage and speed: o Write a regular function. o Declare a generic and implement it just for the base

Gnome Toaster works a treat. ----- Original Message ----- From: Patrick Masters <pmasters@fuse.net> Date: Tuesday, April 16, 2002 10:45 pm Subject: [vorbis] OGG CD Audio Burning > In talking with a few friends the other day, I showed them Vorbis. > Theyloved the sound quality and said it sounded than MP3. However, > theymentioned the one thing that I feel is stopping the massive

There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back