similar to: mute a call/ re-invite mid-session?

Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold,

Hello everybody, I'm having issues with calls being dropped on Aastra phones, when the call is on hold. Tested with models 6863i and 6867i. I've figured that the call is dropped by Asterisk when it reaches the rtpholdtimeout limit. I've reported the issue to Aastra, asking them to implement some kind of "RTP keep-alive" feature on their phones. Maybe the phone could send

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

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

I have a CyberPower PR3000LCDRTXL2U with a BP48V75ART2U expansion chassis, which I am monitoring using NUT 2.8.0 (on Gentoo Linux). TThe UPS appears to be telling me that the batteries need replacement due to age. CyberPower support has confirmed that for me and told me how i should be able to mute the alarm from the front panel until I can replace the batteries, but it doesn't appear

On Thu, Mar 21, 2024 at 12:23?PM Phil Stracchino via Nut-upsuser < nut-upsuser at alioth-lists.debian.net> wrote: > I have a CyberPower PR3000LCDRTXL2U with a BP48V75ART2U expansion > chassis, which I am monitoring using NUT 2.8.0 (on Gentoo Linux). TThe > UPS appears to be telling me that the batteries need replacement due to > age. CyberPower support has confirmed that for

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

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).

A final note: http://lists.alioth.debian.org/pipermail/nut-upsuser/2014-April/008977.html That message talks about the Eaton Nova AVR which on the Eaton 5E product page it says the 5E replaces the Nova. Do they perhaps share hardware/firmware? Thanks, Leith Bade leith at leithalweapon.geek.nz On 10 August 2014 10:16, Charles Lepple <clepple at gmail.com> wrote: > On Jul 26, 2014, at

Hello, all. I see there is an "o" option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

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

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

Hi, Any suggestions to below problem? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem

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

Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

On 3/21/24 14:00, Greg Oliver via Nut-upsuser wrote: > All I can say is definitely replace the batteries if you are going to > keep using it.? The CyberPower(s) I have had in the past all have the > battery check that cannot be turned off and when the batteries are > actually "dead" the unit will kill the load to everything every time it > runs the check.? Just FYI.

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?

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