similar to: agi RECORD FILE with offset

Thanks for your quick replies - I had no idea this list was so responsive! One more issue for you - we are trying to build a custom voicemail application and so far everything works great by using AGI to play prompts, get input and record files. However one of our customers has asked for some pretty advanced functionality, specifically the ability to: * stop in the middle of recording (so far so

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

Hi All, I was wondering if there is any way to encrypt IAX traffic? I am aware of the ability to use md5 or RSA for authentication, but I'm talking about the packets themselves, after authorization has already occured... Forgive me if this is documented somewhere, but I all I could find online was a presentation with the statement "there has been talk of adding encryption to the IAX

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

Hello rsync people Today I was recovering data from a beginning-to-fail external USB hard disk. I started with my usual 'rsync -av --ignore-errors <source> <dest>', and that was fine until it got to the first I/O errors. It paused but continued after the first couple of errors, but then the disk started buzzing and rsync gave error messages for every file (I'm afraid

On 06/22/2015 07:20 PM, Frank Cox wrote: > The last time I wanted to record a voice announcement on my computer was a few years back, and as I recall I hooked up my Microsoft Lifechat headset and used Audacity to do the recording. > > Now I want to record a "recitation", which consists of me playing my piano and talking over it. The USB MIDI port won't give you audio, just

Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

Hi, I'm trying to calculate p-value to findout definitely expressed genes compare A to B situation. I got this data(this is a part of data) from whole organism , and each number means each expression values (that means, we could think 'a' gene is 13 in A situation, and it turns 30 in B situation) To findout probability, I'm going to use Audic - Claverie Method. ( The significance

Thank you for your response. The point is that I have configured my system to shut down at the event "upsc Inform@localhost ups.status"!="OL", not at "LB". So there will be no /etc/killpower file created. At an ups.status != "OL" event, the following script named "notifyme" is run: echo `date` >> /root/upsLog echo " No power, shutting

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

I've spent the last week or so troubleshooting echo problems at my Wife's business, and I've been able to clear up about 99% of the echo, but there is still a little residual echo that I can't seem to "tweak out". The users describe it as "buzzing or crackling", but what it sounds like to me is a slight echo, but just of one syllable of the word.

I've got a Digium TE405P feeding three Carrier Access AB-II units located in three separate suites of the complex my company resides in. I've had one AB-II connected over about 15' of cable and running for several weeks now, and it seems to be perfectly happy. Over the weekend, I added two more AB-IIs, one in each of the other two suites. The second unit is about 150 cable feet away

The last time I wanted to record a voice announcement on my computer was a few years back, and as I recall I hooked up my Microsoft Lifechat headset and used Audacity to do the recording. Now I want to record a "recitation", which consists of me playing my piano and talking over it. I have a Casio CDP-230 digital piano but I've never tried hooking it up to my computer and making a

Hi, I am attempting to create a intercom buzzer system using asterisk as a back end. Most is figured out except the actual action of buzzing the door. I need to detect whether a DTMF key was pressed by the the called party (the resident). Is this possible to do using just a dialplan? I can't see any options on the Dial command that would lead to this, am I looking in the wrong place? I looked

We are working on a new implementation of asterisk. We are using a fiber-served WorldWide Packet switch at the home that incorporates a VOIP T2 switch that feeds 2 POTS connections. We are told that the T2 is programmed with code that follows the PacketCable spec. This version has a problem with the 'congestion' message which is based on the MGCP RFC (2705) spec. This causes the T2 to

Hi list, I have a client with a strange requirement: putting a noise gate on the Asterisk channel. For those who are not familiar with them, noise gates are used in musical instruments to avoid entering low-level noise into the amp system. What they basically do is, they measure the volume of the channel, and when it's too low they just let the channel close, i.e send perfect silence,

I am running Asterisk on our LAN with Cisco IP phones and softphone clients, SJPhone and Firefly. Everything on our LAN works fine and the quality is good. We have recently registered other callers not on our LAN who are using SJPhone and Firefly as well. The audio quality is fine, but I consistently hear a buzzing sound in the background. We have experimented with the line in and microphone

I'm running FreeBSD 4.2. I downloaded and installed Icecast 1.3.10, Liveice, mpg123, and Lame. I know the Icecast streamer is working because I can feed it from WinAMP on my PC and listen to the stream somewhere else and it sounds fine. I know liveice is working too because Icecast shows that it connects and is sending audio to it. However when I listen to the stream on my PC with WinAMP