similar to: FW: Polycom 501 issue with latest firmware : sluggish keys

Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the "latest publicly available" from Polycom's web site). I've noticed that some keys get "sticky" though. Soft buttons for example (i.e. "end call") need to be pressed 2-3 times for them to react. I've downgraded to

Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying

Here is what I had to change on the phone1.cfg file: I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : reg.1.server.1.expires="30" Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

Hi All, I'm not sure if this is related directly to asterisk or not but on my Polycom 320 when I try to dial a number smaller than 4 digits I get an error on the phone saying "Enter more digits". The dial plan section is listed below. <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap

> I ran into this problem with an early batch of IP650s. Polycom's firmware > version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I've noticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number

Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size

Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten => 110,hint,SIP/110 exten => 111,hint,SIP/111 exten => 112,hint,SIP/112 exten => 113,hint,SIP/113 exten => 114,hint,SIP/114 The hints seem to be

I recently had a bear of a time getting a Polycom Soundpoint 500IP up and registered.. Now that its registered I ran into a problem w/ the dialplan. Needing to dial x101 I'd dial 10 - then get a fast buzy.. Also making a local call - dialing 95551212- would give me a fast busy after the 7th digit - so 9555121.. Same w/ LD calls... This dialplan really got me down as I didn't find

Hey all,  I've been running a Win10 Pro (64bit) guest for some time now.... and originally it was pretty snappy and I had no complaints. Somewhere in about the last 3-6 months I've been noticing sluggish performance -- but not overall. The system is mostly remote so I do a lot of my admin via RDP or the local console with the KVM Manager. I recently did a complete fresh install of

Hi Reuben, I think I remember having the same issue when I first used icecast. I think from the best of my memory I swapped from using shout to IceS and this solved the issue. I could be wrong though my memory isn't too good and you may actually be experiencing something totally different than me, however I would say with your bandwidth a 128 stream shouldn't be a problem. Hope this is

Greetings, This is my first stab at this mailing list. Forgive me if I am over/under verbose. We have been running KVM for a while now and love it. Recently I have noticed a weird anomaly that is getting on my nerves. When I clone (GUI or virt-clone from CLI) both* virt-manager* and *virsh* enquiries get extremely sluggish. Using top I cannot see any resource issues. No I/O wait, the load

Hi, I am a newbie here. I just tried to set up mp3 streamer server on Redhat Linux 7.2, using icecast and shout. I then tried to connect to the server, from another machine and different LAN, using xmms. And the stream is so sluggish, it gets cuts off for every 2 seconds or so. Both LANs (the server and where I am connected from) are T1. My mp3 was all encoded in 128 KBps. I then asked someone

Hi, I am a newbie here. I just tried to set up mp3 streamer server on Redhat Linux 7.2, using icecast and shout. I then tried to connect to the server, from another machine and different LAN, using xmms. And the stream is so sluggish, it gets cuts off for every 2 seconds or so. Both LANs (the server and where I am connected from) are T1. My mp3 was all encoded in 128 KBps. I then asked someone

Slackware 10.0. Wine version whatever was current a month ago, how do I tell what wine version I have? Anyhow, I'm experience two issues: I have KDE setup with 4 virutual desktops. If I'm running WinMX via wine, and if I try to switch to a different desktop, it immediately switches back. I basically can't use the other desktops because I can't switch to them. Also, I can't

Thomas suggested posting these here so they don''t get lost. Making Scriptaculous Snappy: *) Drag/Drop - make release cost proportional to remaining distance For drag/drop - change the ''release'' animation time to be proportional to the remaining distance the dropped item has to travel. If a user drags the item almost all the way to the final location, the release time

On Thu, Feb 18, 2016 at 10:41:42AM +0100, John Obaterspok wrote: >2016-02-18 10:13 GMT+01:00 Martin Kletzander <mkletzan@redhat.com>: > >> On Thu, Feb 18, 2016 at 08:49:38AM +0100, John Obaterspok wrote: >> >>> Hello, >>> >>> I'm using virt-manager on my F23 box to run a Windows 10 image but the >>> performance is so bad it's