search for: earphones

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Warning: Sligthly off topic. http://shelleytherepublican.com/2006/04/linux-european-threat-to-our-computers.html Quotes: > And guess what software Osama Bin Laden uses on his laptop? > > If you guessed it was Linux you would be 100% right. > Next time somebody asks you how Al Queda agents pay for their > rifles and rocket launchers, you can tell them that foreign hackers >

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thou...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems,

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

I'm running C7, fully up to date. I'm also running the MATE desktop (from epel) though the original installation was Gnome. a couple weeks ago my USB headset worked as both earphones and mic. today when I plug it in I get nothing. right-clicking on the speaker icon in the upper panel I can choose "sound preferences" (as an aside, how can I figure out what program that actually is? it has no help or about option) and in the "Hardware" tab it shows several d...

Hello, I'm from Korea and I'm trying to implement realtime AEC with 2 computers and using speex. Actually , It works well with 2 computers and they are communicating.(It looks like double-talk but actually one aec and preprocessing algorithm is running in each computer) Anyway the problem is after processing(I'm using aec and preprocessing for noise suppression), the output

Hi I was using the Playback application to play an MP3 file after compiling and installing asterisk 1.8.7.0 with format_mp3 and it seems to me that asterisk is transcoding the file to an slin on the fly rather than playing the mp3 itself. Is this what it does? Also, does this mean I might as well change the format of MP3s to WAV seeing as I'm used to doing that anyway? Thanks Ish --

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using

Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent:

...raints fulfilled... therefore i admit I have not tried = it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the "mute" switch to disable the callers to talk to = the conference.. (so to say it mutes the callers microphone, not his earphones.... am I wrong?=20 nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all = feature you ask for in it On Jan 16, 2013 5:37 AM, "Yves A." &...

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Hi all! I've finally been reduced to having to install Skype on my Linux box. I resisted for years, but now ended up trying it. and while the latest RPM installs just fine, it refuses to acknowledge that I have a microphone! In fact I have two: 1 in the USB web cam (it finds the cam), the second in a Plantronics USB headset, which works fine but not with skype. it is as if it doesn't

I'm a mathematician and programmer working on experimental surround sound techniques. Some of the ideas I'm working on require dozens of channels. These channels are often highly correlated and are very well suited to compression. I'm new to Vorbis. Does it attempt to address such issues? Thanks, --Richard --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN