similar to: G.729 and Voicemail

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

The G.729.1 "wideband" codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it

Opus or AAC will give you comparable results at reasonable bitrates (~128k). Though, I would suggest finding a way to get more storage. You could upload to Backblaze B2 or AWS S3 for pennies, if your current host won't let you upgrade. On Sat, Apr 15, 2023 at 3:36?PM D.T. <ohnonot-github at posteo.de> wrote: > Situation: > > - remote virtual server with very little

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

Hi, We've been running g.722 in asterisk 1.6.09 for awhile now,

[ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source

List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all ; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816

Dear ALl Can we use Asterisk for only for transcoding?. if yes how many concurent call we can transcode with help of Astetrisk? For this we only need to config SIP.conf or any other file too. Thanks Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100704/f6159f70/attachment.htm

how to make g.729 preferred, but failover to gsm I've purchased a few g.729 licences, and would like to set up iax.conf such that g.729 is used if they are available, but then it fails over to gsm. I'm not sure how to specify such a preference. I'll let the server transcode from ulaw (from the sip phones) to g.729. Got plenty of CPU for the number of phones we run off that

Situation:? * remote virtual server with very little storage (estimate: I can spare about 40G for music) * local music collection of ~80G in all sorts of formats - lossy in varying quality, some lossless too Vision: * stream my whole music collection randomized so I can listen to it anywhere Plan/Idea: * Locally transcode everything to one format that results in files that are?

Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com

Can Meetme be made to work with G.729? (I gather not) If a call comes in (internally or externally), the call comes in as a G.729 call, which then re-negotiates to a G.711u call when if gets transferred to a MeetMe room. Is there a way to set up asterisk that will allow me to have internal phones renegotiate to G.711, with the external lines instead transcoding within asterisk. (runtime is more

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

On Mon, 2007-02-05 at 12:00 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Mon, 5 Feb 2007 11:36:28 -0500 > From: Andy Davidson <andy@nosignal.org> > Subject: Re: [asterisk-users] Question on G.729 (was: H.264 *Not > Patented*) > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >

Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I

I'm trying to use the Digium suplied G.729 Codec, I have ran the register utility, and got my licenses written to /var/lib/asterisk/licenses, but when a start Asterisk I got the following errors: [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: This module is supplied under a

We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10, acting as gateways from SIP to ISDN PRI interfaces. Each has one Digium TE420 (with hardware echo cancellation) and one TC400B for transcoding, in order to handle 60/90 contemporary calls in peak hours. In my logs there are hundreds of thousand warnigs per day like these: transcode.c: no samples for lintoulaw transcode.c:

Hi to all, we are searching for a hardware based DSP solution for use with Asterisk based on PCI or MiniPCI to reduce main processor load and to use embedded boards with Digium E1/T1 cards like TE410P. does anyone know about any manufactorer of those cards or someone who is able to develop/build such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,