similar to: Polycon IP 300 SIP vs Grandstream BT-101Deployment

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at

I am not sure if this is a bug or a feature... The 2.0.7 version no longer wants to connect to a DOS 6.22/DOS LANMAN2.1 (wg1049.exe) server. Version 1.9.18 does work. Both smbclient and (more importantly for our needs) smbmount fail. However, both work fine with win95 boxes. The problem seems specific to the DOS server. I could not find any reference to this issue. Am I perhaps overlooking

I had this happen to me on FC4 using 3.0.2-2 Xen. I updated the various external dependencies and rebuilt xen and tools and was able to sucessfully install FC4 in a hvm. I will give winxp a go later in the afternoon. C ________________________________ From: xen-devel-bounces@lists.xensource.com [mailto:xen-devel-bounces@lists.xensource.com] On Behalf Of Rene Purcell Sent: Friday, April

Just a couple of small bug-lets/stuffs in Sweave: - It stripes off the two lines starting with @ (this is a verbatim section showing plink's start-up message): ======================================= @----------------------------------------------------------@ | PLINK! | v0.99p | 17/Dec/2006 | |----------------------------------------------------------| | (C)

I just upgraded from Ubuntu 11.04 to 12.04 and am working to restore access to the windows shares I use at work. smbclient connects immediately: sudo smbclient //server/share -A /etc/.smb_creds.txt mount.cifs fails (with "mount error(13): Permission denied"): sudo mount.cifs //server/share /mnt/ -o credentials=/etc/.smb_creds.txt my credentials file is as

Hi, i am new to ajax and new to script.aculo.us. I created a blog, http://www.zorpal.com/blog/, using some of the script.aculo.us features and commands. My sites works great in firefox, which everyone should be using, but if you open it in IE it kills it. It only kills it after a script.aculo.us motion has been called and finishs it''s execution. i really like the layout and the visual

Hi All, does anyone know how to import binary .bed files generated by Plink (http://pngu.mgh.harvard.edu/~purcell/plink/ ) into R? the Plink FAQ explains how to conver other types of files, not the .bed. Cheers, Federico -- Federico C. F. Calboli Department of Epidemiology and Public Health Imperial College, St. Mary's Campus Norfolk Place, London W2 1PG Tel +44 (0)20 75941602 Fax

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 =-

Hi, ? I am very new to R. So, pardon my dumb question. I was trying to write my own function to run a different model (perform an ordered logistic regression) using the example in website http://pngu.mgh.harvard.edu/~purcell/plink/rfunc.shtml But R returns a error `R Error in eval(expr, envir, enclos) : object 's' not found' when I run it. What am I doing wrong here? Here's

I note that "current implementations of R use 32-bit integers for integer vectors," but I am working with large arrays that contain integers from 0 to 3, so they could be stored as unsigned 8-bit integers. Can R do this? (FYI -- This is for storing minor-allele counts for genetic studies. There are 0, 1 or 2 minor alleles and 3 would represent missing.) It is theoretically possible

Hi Everyone, I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the

If you get a lot of those messages and you don''t want them: peth1: received packet with own address as source address or peth0: received packet with own address as source address This is the way I found to solve this problem on Debian maybe someone can post more information about other distribution like suse or redhat... This problem will occur if you have more than one dom0 on your

All, I am setting up my first * box and am trying to configure SIP Softphone to SIP softphone dialing. When I dial ext. 2001 it rings once (Very short) then immediately goes to voice mail for ext. 2001. I keep seeing the "Got SIP response 482 "Loop Detected" back from 192.168.1.252", immediately followed by "No one is available to answer at this time". I've

Rui, >Yes, it's just as you said, when I create a new profile, it opens a >window with the fields(Proxy Domain,Account,Password,CallerID). but it >doesn't let me to input value to these fields. I don't what's the matter. Edit the profile, and on the "Initialization" tab and make sure the "Inquired" box is checked by the fields you listed above.

Rodolfo, I haven't looked up how to do this with sip phones, but the zap channels can be configured in groups that will hunt through the group until a non-busy line is found. http://www.voip-info.org/tiki-index.php?page=Asterisk%20ZAP%20channels#comments Here is a link to PBX hunting with the dial plan. http://www.voip-info.org/tiki-index.php?page=PBX+Hunt+Groups I haven't tried

Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of

Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/21cd5c04/attachment.htm

Hi. We just opened a gaming center and have chosen to run a FreeBsd box for our firewall. IPFW is configured at it's very basic running natd through rl0 and allowing any to any connections from the lan to the outer world. Natd controls access to the lan. We have a 6.0 mb/s ADSL net connection for all the gaming clients to use, however if a gamer starts downloading a file, that file

Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g.

Recently our Unix Administrator installed Samba v.2.07. It was configured to attach to our NT domain. When this occurred we started to experience problems with user authentication's to the domain and it appeared that the "Samba" server had become the master browser. Have you seen this problem occur at other sites? If so, what is the work around? We have been using Samba v2.06