search for: earphone

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 implementations with Asterisk and Sipura SPA-841 on different clients and network topolo...

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 tho...

...on-zero on 'DAHDI/1-1'" have relation with the problem? I was testing calling from my cell phone to an analog telephone and if the other person hangs before I do it, I see that in the my cell phone the call even continues persisting so that if the person of the other endpoint take the earphone again after to hang, we can continue speaking :-D It will be some trick of the telephone companies to collect more with the unwary subscribers? :-D Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoh6HsACgkQZpa/GxTmHTca+wCfd7ogHaozBDc37DVnT0lrMmYU vY...

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

...reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@list...

...ting Call mute Call forward Call transfer 3-way conference Voice mail SMS over SIP Phone book - (1000 entries with photos) Incoming call prompt with picture View phonebook during call Enter sketch pad during call Adjust volume during call Auto-answer/flip answer Quick silence Turbo dial Manual/Auto/Earphone redial Call history (20 entries) Data Application Features POP3 E-mail client (SSL support) > 100 full E-mails with attachments up to 200KB > Document viewer for MS-Office and PDF files Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0 Instant messaging: QQ Multimedia Features...

...reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@list...

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...

...ied with wave file instead of realtime, but it wasn't reduced. Just when I'm using both aec and preprocessing on realtime, it was reduced. Do you have any idea about that? Some times It isn't reduced when it doesn't work well(In this time I can hear my voice a little bit from my earphone). I thought most reasoon is because of realtime situation. Please help me and give me some hint. Thanks a lot. Regards Seo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20140710/e76228ad...

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

...;t know what 'Hangup' from Outlook is. > > When I clik 'Hangup' in Outlook there is nothing in Asterisk > debug/cli window. > > Only problem is that Outlook still thinks that call is active > even if you hangup the phone manually.. I mean, when I put > the earphone back to base/station/phone.. whatever. Dialing > works just fine. > > Because of that you need to close that window 2 or 3 times if > you want to call same person/contact again. > > Bye, > > Tomislav > > > > > -----Original Message----- > From: ast...

...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