similar to: Jittery meetme conference using Linksys 942 phones

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

http://bugs.freedesktop.org/show_bug.cgi?id=24596 Summary: Video adapter #0 is jittery, while adapter #2 isn't. Product: xorg Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at

On Sat, Jul 27, 2019 at 7:37 AM Ralph Corderoy <ralph at inputplus.co.uk> wrote: > The video plays, CPU load is less (my aim), but there's ‘tearing’ of the > picture as if small rectangles that are updates are appearing in the > wrong location, off by a little. If I step through the frames with > mpv's ‘.’ and ‘,’ then I've found a pattern: one frame's picture is

Às 14:46 de 27/07/2019, Ilia Mirkin escreveu: > On Sat, Jul 27, 2019 at 7:37 AM Ralph Corderoy <ralph at inputplus.co.uk> wrote: >> The video plays, CPU load is less (my aim), but there's ‘tearing’ of the >> picture as if small rectangles that are updates are appearing in the >> wrong location, off by a little. If I step through the frames with >> mpv's

Hi people, I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use the WAN port, most of the time when I try to connect to the network or restart the IP Phone I can't get internet connectivity . I tried using both static IP and DHCP, but the problem is the same. Some have had similar problem with this brand of IP Phone? Thanks for the help and attention. Hugs

Hello List, After seeing a few positive responses for the Linksys SPA-942 phones I was hoping to get some answers on the following questions: * How do the phones handling system wide paging? Is it similar to the Polycom phones? * Can a corporate directory be configured with the phones using Asterisk? * How is the speaker phone quality? Thanks Roy Anciso Director of Technology Manistee

Hi Ilia, A fortnight ago, you wrote: > > The video plays, CPU load is less (my aim), but there's ‘tearing’ of > > the picture as if small rectangles that are updates are appearing in > > the wrong location, off by a little. If I step through the frames > > with mpv's ‘.’ and ‘,’ then I've found a pattern: one frame's > > picture is good, followed by

Hi, I'm having trouble with hardware decoding of video with mpv(1) having switched from Nvidia's proprietary driver to nouveau due to Arch Linux removing its nvidia-340xx package. The machine is an Acer Revo. $ lspci -nn | grep VGA 01:00.0 VGA compatible controller [0300]: NVIDIA Corporation GT218 [ION] [10de:0a64] (rev a2) I think this is codename NVA8 with the

Hi I have a customer who is using Linksys 942 phones. When they try to transfer a call the Asterisk CLI reports that both legs of the call must exist on the server. The call they are trying to transfer then drops. Does anyone know why this is and how to fix it? TIA Regards Jon ___________________________________________________________ Want ideas for reducing your carbon

Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

Immediately after running winetricks a month ago, several very important directories disappeared from my hard drive, and despite paying a lot of money to good data recovery company, I have not been able to recover them. I have searched this forum and elsewhere for evidence of this happening to anyone else with no result. Is it at all possible that the script deleted my directories by accident

Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _____________________________________________________________________________ D?couvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et

It seems that VCPU system time isn''t monotonic (using 3.0.4). It seems it might be correlated to when a VCPU is switched across real CPUs but I haven''t conclusively proved that. But e.g.: { old = { time = { version = 0x4ec pad0 = 0xe8e0 tsc_timestamp = 0x22cc8398b7194 system_time =

Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use "canreinvite=yes". The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and

Hello all, We've been using our Asterisk system live for about a month now and I'm looking to tuning a few things. First, is echo, I receive a fair amount of echo during the first 10-15 seconds of incoming calls. Next is a very weird problem. We have serveral Polycom IP300's and one Budgetone phone. It seems that if we unplug & move the Budgetone (which happens a fair amount

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Has anyone tried using a debit/credit card terminal as such: Terminal <-> SPA-2000 <-> Public Internet <-> * <-> PRI I'm hoping someone will tell me they have done this successfully and rarely experience dropped calls. Though I'd like to hear from anyone who has tried and failed as well. Thanks, Trevor Peirce

Asterisk/Skype update available here - http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ .... It's definitely an update that updates absolutely nothing :-), more news at 11 :P Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: