search for: hearding

Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing "#", and she heard a buzz noise in the phone and the

Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a "live human" on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of "report a spill" or crucial system of any sort.

I heard talk of a centos-supported xen dom0 for CentOS 6.4, but I haven't heard talk of such a thing lately, and I haven't seen where to download it, which could just be me being stupid.

Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey

I'm wondering if anyone has seen (heard!) this before. I have a site which has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice and I was quite angry when I heard they'd been installed )-: They have an asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to the telco (BT, in the UK) It basically works and does what it says it's supposed

Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks.

I built asterisk-trunk at 92526 and asterisk-addons-trunk at 496. I have my Bluetooth cell phone connected. It almost works. In mobile.conf, I have "context=incoming-mobile" for the phone, and that looks like: context incoming-mobile { _. => { VoiceMail(9999,b); Hangup(); }; } Calls to the cell phone get directed answered by Asterisk and directed to

I'm am currently using autuvb4 at q-2 mono 44100. This produces roughly 28 to 34kb/s. It's ok but I've heard AACPlus at 32kb/s stereo and it's definately better, unfortunately. Regards, Ross. >> I'm hoping that Monty and others can improve Vorbis to start >> competing >> with AAC+ at low bitrates. Monty said he had some ideas but I wonder >> if

Hi, I am having a little problem and I hoped maybe I could get some help here. I deployed a Asterisk 1.8 server of my own to make SIP calls just between my friends. The server is configured with a public IP address. My friends and I are using "Acrobits Softphone for iPhone" as a client. I am using its push service which is hooked up to my Asterisk server. Now, the current situation is

I have heard rumblings about the new kernel for 4.4 A change was made to the kernel for disk I/O to make it better/faster. I havnt heard/seen anything about what that really translates to. Any thoughts/opinions on this. jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

we are experiencing 30 second delay before voice is heard after answer when we ran wireshark it showed the problem between frames 634 (where the softphone answers) and 1366. Between those frames, asterisk receives RTP packets from both the softphone and the sip carrier, but doesn't forward them to each other.... Then we see a bunch of packets sent at once from the asterisk server, until

I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are receiving random DTMF tones on their side, but that are not heard by the outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have always had this issue. I am only using SIP on the Asterisk server and all extensions and trunks are set to rfc2833; outside of this issue DTMF operation works fine.

When experimenting with Opus encoder, I took a random FLAC file, encoded it, heard it, and heard the original file. When hearing the opus encoded file, I thought it was a flute solo, but when I heard the original, I nearly fell off my chain when I noticed the piano I couldn't hear from the Opus in the first time. Now I know there is a piano, I can hear it in the Opus file, but it is much

On Tue, Oct 13, 2015 at 4:23 PM Hal Finkel <hfinkel at anl.gov> wrote: > > From: "Tanya Lattner via llvm-dev" <llvm-dev at lists.llvm.org> > > > Some back story here. I have gotten many requests through email and > > at the developer meetings about having a Code of Conduct and > > specifically having one for LLVM Developer Meetings. It has been

Hi all: when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. _________________________________________________________________ Windows Live?: E-mail. Chat. Share. Get more ways to connect.

Can anyone suggest any apps/scripts for bulk generating passwords from real names (e.g. jsmith from John Smith) that would check for duplicated in an existing smbpasswd or passwd file and append a number to the username (e.g. jsmith1, jsmith2). Thanks Lee Baker MEng MIEE Music Technology Coordinator Email: <mailto:lbaker@mcauley.org.uk> lbaker@mcauley.org.uk

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Hi, Slightly off-topic (as it's not directly about using or developing LLVM): http://code.google.com/p/unladen-swallow/wiki/ProjectPlan "Our long-term proposal is to replace CPython's custom virtual machine with a JIT built on top of LLVM, while leaving the rest of the Python runtime relatively intact." Just curious, has anyone here heard more about this project? Regards,

Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We are of course having echo problems are on the far end due to digital/analog conversion on the local end using a channel bank. We have purchased an ADIT 600 and yes the complaints are "far less" however we're still getting them. While I have

I run CentOS at home, not just at work... Anyway, I've got a friend in Chicago who recently mentioned that they'd like to do videocalling. Now, I've heard of skype, but a quick google says there's some problems on Linux. I also see ekiga, and aMSN. Anyone here run such a beast, and have any recommendations or comments? Obviously, must work on CentOS, not Ubuntu, or Fedora, or