similar to: Polycom gain settings

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

John, By chance do you know how to set a default ringer? What I have done is the following: <DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="7" se.rt.1.callWait="6" se.rt.1.mod="1"/> As you can see, I want 7 to be the default ringer for line 1... For some reason, it doesn't take these changes.

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Thanks, I was afraid to try and change gains that I didn't know what they did simply because I don't want to blow a speaker or something.. :) I'll try it today. The only thing I haven't figured out is how to set a default ringer in the configuration file, set my time to EST w/ Daylight Savings and when receiving incoming calls if it's possible to see NAME & NUMBER instead

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

On 06/05/2015 03:43 PM, John R Pierce wrote: > On 6/5/2015 1:33 PM, Michael Hennebry wrote: >> It's a desktop in an old house. >> The outlets have ground-fault protection, >> but the third prong is ungrounded. > > not sure how GFI would function at all without a valid ground, unless > the GFI is wired to neutral, which is dangerous on its own. very. it defeats

On 6/5/2015 1:33 PM, Michael Hennebry wrote: > It's a desktop in an old house. > The outlets have ground-fault protection, > but the third prong is ungrounded. > not sure how GFI would function at all without a valid ground, unless the GFI is wired to neutral, which is dangerous on its own. you might get a 3-prong-to-2-prong adapter and plug the PC into that, leaving the ground

now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the "hands free" mode? Thanks

--- CE <ce7@sbcglobal.net> wrote: > I'm hoping to set up a headless system to playback my audio once I > get back home after an extended leave. Ideally I'd just ssh in and > use a curses based player to play back FLACs and mp3s. This leads me > to a few quesitons: > 1. What is the minimun processor speed needed to decode and play > flac files? I have an old

1. How do you set the music on hold to work with asterisk. Right now when I place a call on hold the caller hears nothing. MOH works with all my other IP phones. 2. Ringer Volume. How do you set the ringer volume? So that it's set on reboot. Thanks

Hi All, I'm using Asterisk for my PBX, I have a remote office that is connected by a VPN link. I am using Openswan on my side and a Linksys box on the remote side. I have a Polycom IP300 on the remote side configured with a static IP address. When I call the phone on the remote side, it rings and establishes the call fine. The problem I am having is that the remote side can hear the call

> > I've never seen stats, but it's probably a safe assumption that the > > majority of IP phones are sitting next to a PC and the additional > > expense has been incurred because "people want a phone that looks and > > works like a phone". That's certainly been my experience far > > outweighing any technical issues with quality or reliability

I'm hoping to set up a headless system to playback my audio once I get back home after an extended leave. Ideally I'd just ssh in and use a curses based player to play back FLACs and mp3s. This leads me to a few quesitons: 1. What is the minimun processor speed needed to decode and play flac files? I have an old P100 I would like to use for the task if it is sufficient. 2. When

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

I'm using Centos-7 on my PC, with ASUS motherboard. internal audio has always worked fine EXCEPT that I can never get audio input (for recording) to work. I want to feed audio from a phono turntable and from a cassette deck. I use a phono preamp for the turntable, or directly connect the tape deck. but either nothing, or low level with horrible distortion. so, I gave up (after years of

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

ACES - Asterisk Communications Endpoint System {the following could be used by any IP-PBX but the name pays homage to Mark Spencer and friends who cannot be lauded enough for their fine work} As you read this it will be obvious I am not a professional engineer but I do have enough knowledge to be fairly certain what I'm proposing is feasible from not only an engineering, but production

Paul, Ah, I see. Our echo is largly under control now. It took me a while to figure out the gains and get them tuned, and now the echo only leaves very small artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, and we find zero echo during those calls. Therefore, I assume that our handsets (Cisco

How do I make my headset work. Before I try it with anything else, I'd like to make it work with arecord and aplay. VLC and movie player can make sounds with my external speakers. though not in my headset yet. I've got an insignia NS-PAH5101 headset. As directed, the red connector is in the pink spot and the green connector in green spot in the back of my computer. arecord folloed by aplay