similar to: Can anyone help? Why does One-Touch record mute/disconnect callif not dialed quick enough?

Hi List, Have an odd problem with the one-touch record on asterisk 1.2.11. All works ok, however one of our users today discovered if he is a bit slow hitting the 1 key after he presses *, the call seems to stay connected but its almost like it is muted. Haven't figured out the delay yet but it seems to be if the 1 is not pressed within 1-2 secs this occurs. Any suggestions? I tried

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left

Hi, We have quite happily been running dovecot with no issues at all for approx. two years now and have had no problem whatsoever. Great work :) BUT (there is always a but!).... yesterday one of our users started complaining that they were having issues with their mail. I checked and although they can still view their IMAP mailbox ok it constantly throws up error messages such as: "The

Hi, Damn I hate upgrades. After updating at the weekend to FreeBSD 5.4-R in a fresh install our samba server just doesn't want to play ball. I have managed to get Samba3 working ok apart from when it comes to using Excel documents. If a user has an excel document open and another user tries to open it, Excel 2003 will fail with 'Unable to open the file, it may be read-only or

Hi, Is it possible to add users/change passwords for users in samba without it being dependant on anything in the UNIX password file? For example I want to add a user to my samba password file without them existing in the UNIX password file. If anyone has any RTFM info they can kick me in it would be most appreciated :) Thanks -- Jamie Heckford Network Manager Trident Microsystems Ltd Tel:

Hi, I'm having problems with an occasional disconnect from phone calls while my phone is on mute. This is a problem with long conference calls, for example. I've a GrandStream GXP-2000 and Asterisk 1.2.1. Anyone have experience with similar issues? Best, WILL

Hi All, We are currently evaluating the confbridge application while we prepare to upgrade our environment to asterisk v1.6.2.x. We have run in to two issues using it to kick/mute participants in a bridge and would like to ask for the experience of others running the application for any work-arounds. Firstly for kicking participants, would it be possible to use the softhangup application

Hi. I am using latest asterisk 1.2 and it would be nice in a meetme conference to be able to mute all but a particular user and then unmute all those users again with one command. Am I missing something or is this not available? Maybe I could write something, but I wanted to check first. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

Hello ppl, Is there anyway to control a call mid-way in terms of sending a re-INVITE with say sendonly, etc. to mute one call leg of a bridged call ?? Looked around, so far, doesnt seem to be possible. If it's not, I think it's quite an important feature (re-INVITES mid-session) for a B2BUA. cheers - Ben.

We are transmitting CELT encoded packages over a lossy channel, where from time to time complete packages do not arrive at the decoder. My questions: 1. What will be the best pattern to be fed into the decoder in order to minimize the audible impact and make best use of internal states from last block (like overlapp-add history etc.) and best blending with next block ? 2. Does this pattern

Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible).

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. I'm using the FreePBX web interface, and I can't

Hi, Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non admins from using toggle_mute to unmute themselves? If there isn't such an option on there any devs here that can ping me off line what it would

On 4/26/20 10:48 AM, Dovid Bender wrote: > Hi, > > Looking at > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there > is an option for admin_toggle_mute_participants however the non admin > users can still toggle toggle_mute. Is there any option for the admin > to disallow non admins from using toggle_mute to unmute themselves? If > there

Is there a setting in gconftool-2 to that sets up all the sound and volume settings? If so I havenot found it. Can someone point me in the right direction. Or is this just done directly with alsacntl? THanks, Jerry

Hi all! Using C6 on an Asus M5A99X EVO R2.0 motherboard with AMD six core FX6300. That board uses Realtek ALC892 audio chipset. For the first time since I got that motherboard I tried plugging in headphones to the front panel speaker output. I was somewhat displeased to note that the speakers don't mute when that happens. I recall seeing info on that in Fedora, but not Centos, and

CentOS-6.6 Does anyone know of an application that would provide a 'mute' buttons for video and audio that could be used from a desktop panel? Ideally this would show the status (muted/cloaked or open mic/recording) of each. Presently, turning audio and video pick-ups off and on is rather cumbersome. And I often forget what state I have left them in. Which occasionally makes for some

Dear R-helpers, I am using confint() within a function, and I want to turn off the message it prints: x <- rnorm(100) y <- x^1.1+rnorm(100) nlsfit <- nls(y ~ g0*x^g1, start=list(g0=1,g1=1)) > confint(nlsfit) Waiting for profiling to be done... 2.5% 97.5% g0 0.4484198 1.143761 g1 1.0380479 2.370057 I cannot find any way to turn off 'Waiting for. .." I tried