similar to: Opus in VOIP

Hello, I'd like to ask whether there is some documentation with recommended parameters for transcoding voice codecs such as G722, G711a/u <-> Opus with near-transparency. My Idea is to have something like: HW-Phone <-> Asterisk <---------> Asterisk <-> HW-Phone (G722) (Opus) (G722) in order to lower the bandwidth between the two

Hi! I'm using the opus-tools package from the current Debian (it says it is 0.1.2). The output of opusinfo is strange: $ opusinfo demo01.opus Processing file "demo01.opus"... New logical stream (#1, serial: 0000456a): type opus Encoded with libopus 1.0.1-rc3 User comments section follows... ENCODER=opusenc from opus-tools 0.1.5 Opus stream 1: Pre-skip: 356 Playback

My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph

We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed

[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) &

Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

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I have been using the ECDF function in the Hmisc package to produce cumulative distribution function plots. The problem is that for small datasets the steps "look bad" (not my characterization but from the client). Is there a way to get the same information but smoothed? I have tried the densityplot (lattice), which gives a smoothed line, but this does not give the cumulative density.

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

All, After months of intensive development, we're finally coming to a stopping point in our rewrite of the libtheora reference implementation. All major features for the 1.1 are in, so it's down to bug fixing now. Please try this first beta and give us feedback. Most notable is that the encoder's rate control is much more configurable, and more capable in each configuration than

All, After months of intensive development, we're finally coming to a stopping point in our rewrite of the libtheora reference implementation. All major features for the 1.1 are in, so it's down to bug fixing now. Please try this first beta and give us feedback. Most notable is that the encoder's rate control is much more configurable, and more capable in each configuration than

Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine

Hello Developers, I am considering using SPEEX on an embedded processor that does not have a floating point unit. Does anybody have a SPEEX performance characterization on a fixed point processor? More specifically, I am interested in knowing how the MFLOPS values from Table 9.2 in the manual translate to fixed-point instructions when SPEEX is compiled with enable-fixed-point option. Any help

I am running R version 2.9.2 on Windows XP OS with RODBC version Version: 1.3-0. Has anyone out there in the R user community successfully appended records to a DB2 table on a remote database using the sqlSave function in the RODBC package? (or by any other means from R?) I posed a similar question a few months ago and unfortunately, did not receive a response. I was hoping recent upgrades to

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hello everyone. What I'm doing: I've made a replacement for app_queue that uses MeetMe to connect the calling party with the agents. When the call comes in it gets put into a MeetMe room with a nice AGI_BACKGROUND so the calling party can listen to music and announcements until an agent becomes available. So far everything works fine. Now I want to give the calling party an

Hi all, Newbie to the group. Just started using Opus as part of a WebRTC project and amazed by the versatility of the codec. Great stuff!!!! I have been trying to understand the performance of Opus inband FEC in the presence of bursty loss. Although I do not have exact characterization of the loss profile, we are seeing issues over WiFi. RTCP reports about 33% loss, but I am guessing a lot of it

Hi. I'm working in a project about web-users behaviour analysis. In a few words, it consists on: - log-files recopilation and pre-analysis - basic stadistics extraction (pages most visited, session lengths, etc.) and on-line report generation in a web-viewable format - more advanced analysis for web-users characterization The purpose of my project is implementing my own data analysis, so