similar to: VoIP experiences with Cable and DSL

Hi Stephen, Thanks very much for getting back to me. My problem is described below. Any help would be greatly appreciated. Thanks, James Hi, Sorry for bothering you because I know that your time is voluntary, but I would really appreciate some help. I work in a hospital in part of Ireland?s national health service, a service which was struck by a massive cyber attack a couple of years ago. Since

The article on BMA (Bayesian model averaging) presents most valuable tools for model selection, but I find one detail confusing in Example 1. In page 4 of RNews 5/2, second paragraph says that the probability of Time variable not being in the model is 0.445. It seems to me that the figure should be 1 - 0.445 = 0.555, because p!=0.445 is the prob. of Time variable being in the model. The plot in

Hi, We?re looking at options for logging agents into the system programmatically via Perl/PHP and I was wondering if anyone else is doing this and if so, how. We're using AgentCallbackLogin now but would like to set up a web interface instead. I've been looking at Asterisk::Manager and didn't see anything relevant and wanted to ask the group before we dove into the Asterisk source.

Hi, I have a valid SPSS .sav file (which I can open happily in SPSS v11 on Windows XP). Opening it in R2.41 on Linux we get this message :- > HSE3023 <- read.spss("HSE.sav") Error in read.spss("HSE.sav") : error reading system-file header In addition: Warning message: HSE.sav: Variable X234 indicates variable label of invalid length 256 Now variable X234 has indeed a

Hi, I am just wondering if there is a test available for testing if a linear fit of an independent variable in a Cox regression is enough? Thanks for any suggestions. John Zhang ____________________________________________________________________________________ [[elided Yahoo spam]]

Pardon the newbie post. I installed Asterisk on a "test" system using the Asterisk@home cd image. When you boot from Asterisk@home is installs an O/S and Asterisk on your PC. How cool is that. But I was wondering if someone could point me in the right direction for updating the version that I have. I'm new to CVS, how do I determine what version to build? Is there a primer on how to

For anyone interested, our company released a PHP/MySQL based content manager for the Cisco 79XX series IP Phones compatible with the SIP load yesterday. It's available via: http://www.sourceforge.net/projects/open79xxdir Best wishes, -Corey ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

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I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve szmidt Sent: Monday, June 27, 2005

Hi, I am want to setup linux box as router with fallback routing. ISP 1 (Cable Modem, Static public IP) ISP 2 (ADLS Router Dynamic IP SUA NAT on router) ISP 1 ISP2 | | | Dyn IP (SUA NAT) | DLS router Cable Modem | (Static IP) | (192.168.1.1 | | | | eth1

Hello! My name is Tubby Frederickson, I'm a computer guru, and an independent associate with a NASDAQ corporation that offers very innovative products to broadband and non broadband telephone users. I saw your email address on a forum that was discussing broadband and telephone issues. Right now we are offering unlimited local and long distance calling throughout North America for only

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other

I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt "They

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

Can anyone that can & is willing to do some IAX usability testing from Costa Rica, San Jose to my Asterisk PBX please pmail your contact info thx tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080210/b0b0d6e3/attachment.htm

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date:

I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? I ask, because I spent yesterday giving it a whirl, and everything seems to go just fine till the very last minute. Zaptel, libpri and asterisk compile just