similar to: Stereo Conferences?

How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(<terminal>) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan dial

I currently have an AGI that calls the Festival text2wave app to write a wav file that my dialplan plays into a call with the Background() command. But the voice sounds terrible: like SAM, the 1980s 6502 voice synthesizer. I tried to slow it down by calling (text2wav -eval "(Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 [...]), but it still sounds like it's talking while

Hi, Two related news here. First, I've merged in support for stereo (or more channels) acoustic echo cancellation in mainline. The standard AEC remains the same, but I added a speex_echo_state_init_mc() call that takes the number of speakers and microphones. Now, one of the consequence of this new work is that by merging this work into the main tree, I had to do things in git that svn just

Hi, As I´ve understood things, the Ogg Vorbis format supports more that two channels (stereo). Is there any tools to encode x sourcefiles into one .ogg file? I am a musican and am thrilled with the ide of makeing music in surround (or in more than stereo). This leads to my next question: is there (developing) any decoders for multichannel oggs to, let us say, 4.1 or 5.1 surround? Wouldn´t it

Is there any calendar client that can point at OX for calendar data, which client can display multiple calendars simultaneously as *overlapping layers* in the GUI? With UI to de/select calendars from view, one by one. That is, a single grid of days displayed, with the events in each day displayed in the same day's view list, as if the layers were all events in a single calendar. And is there

OpenWengo has just released WengoPhone v2.1.0: http://www.openwengo.org/index.php/openwengo/public/homePage/news?payload[newsId]=0 . Has anyone had success (or notable failures) using it as a client for Asterisk? Any advice on integrating it into dialplan, apps, config DBs, etc? -- (C) Matthew Rubenstein

I've got a Cisco 7970 that's not completing its network registration to Asterisk. The "Registering" message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes "Updating CTL" then "Loading...", then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed

If call recordings were stored in stereo and the callers were evenly distributed along the stereo spectrum. BAM. Just a cool idea I thought up, but probably completely impossible, and even if not, likely too much work for too little reward. Even less likely would be live stereo conference calling. But hey, RTP certainly supports stereo streams, right? I don't know if any of the used codecs

I have found a site that list the following (no date in the post, so it may be old): "since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down" Does that still applies in asterisk 1.2.14/1.4.x ? Or do we have to tweak source code to balance loads (transcoding,etc) between

Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date:

If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

On 27/01/17 12:29 PM, Jon Lederman wrote: > Thank you. Yes, we do need both channels independent. So, if we > encode each channel separately, we will be sacrificing the > compression ratio we would achieve with stereo encoding, correct? Not necessarily. Stereo makes two assumptions: 1) It assumes the two channels are somehow correlated 2) It assumes the two channels are meant to be

Hi all, I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant). I'm using it at home for a month. I wondered how much it could be loaded, so I tested it with pbx-test: I could place up to 15 simultaneous SIP calls before it got no more responsive. All in all a good, stable and cheap solution for home

I noticed that my previous message is not very complete so I send here an "enhanced version". Please disregard the old one an reply to this one only. ( you can delete the ATTENTION word from subject ) Wilson (defiler@null.net) wrote : > There are two ways to decode multi-channel audio. In hardware, or in > software. > Hardware: A receiver or processor takes a Dolby Digital

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

I've been doing a fair amount of work with Vorbis support for Ambisonics, which seems to be going along nicely. It seems that there is signifant interest in coding 5.1 material with Vorbis esp as tarkin becomes more complete, so I've decided to take a break from pure ambisonic work to look into this. I've decided that the best (from a pure elegance and patent avoidance) way to handle

I've been doing a fair amount of work with Vorbis support for Ambisonics, which seems to be going along nicely. It seems that there is signifant interest in coding 5.1 material with Vorbis esp as tarkin becomes more complete, so I've decided to take a break from pure ambisonic work to look into this. I've decided that the best (from a pure elegance and patent avoidance) way to handle

The US has a Do Not Call list to which people can subscribe to prevent being called by advertisers. Federal laws (strengthened by some state and more local laws) assign penalties for calling people/phones on the DNCL. Is there a query gateway that Asterisk (or an app using Asterisk) can filter through to ensure a number is OK to call (not on the list) before calling it? -- (C) Matthew Rubenstein

Wow! Ogg is sounding _really_ fine these days. Waaaay past my ability to tell from CD on my crappy sound card, even at pretty low bitrates. Which leads to my question: there's a huge difference between what my sound card puts out and what my CD player can do (Rotel RCD-950 to Classe' Audio Twenty preamp to Acurus A80 amp to Epos ES-12 speakers, in case anyone cares). If my computer is