search for: remington

...en => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I will check out the queue system you referenced. Thanks! Wiley -----Original Message----- From: Seth Remington [mailto:sremington@saberlogic.com] Sent: Saturday, July 17, 2004 7:45 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Using a group variable for a groupofextension to dial It would ring all three at the same time. You are probably looking for a roundrobin call queue -> htt...

...ample right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause it to ring through each of these extensions or would that result in these three extensions all ringing togeher? Thanks! Wiley -----Original Message----- From: Seth Remington [mailto:sremington@saberlogic.com] Sent: Saturday, July 17, 2004 6:35 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Using a group variable for a group ofextension to dial Maybe I am misunderstanding your question but are you looking for the '&' operator? Dial(t...

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej -------------- next part -------------- An HTML attachment was scrubbed...

...lt, ". This should not happen")) RVAL <- list(statistic = c(W = sw$w), p.value = sw$pw, method = "Shapiro-Wilk normality test", data.name = DNAME) class(RVAL) <- "htest" return(RVAL) } <environment: namespace:ctest> -- Richard E. Remington III Statistician KERN Statistical Services, Inc. PO Box 1046 Boise, ID 83701 Tel: 208.426.0113 KernStat.com

I'm using rsync to backup files from host A to host B. host A is running Solaris 2.6 with rsync 2.3.1 host B is running Solaris 7 with rsync 2.5.5 host A runs as root and syncs to host B through OpenSSH 3.4p1 to a non-root user on host B. If I bring a file back from host B to host A, I find that the original owner has been changed from whatever it was on host A to the non-root user on host

Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? Appreciate pointers.. Thx a lot, J --------------------------------- Do you

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

Hi All, I am using the allShortestPath function based on Floyd's algorithm in e1071 package. It runs great when I have less than 5000 nodes. But when I tried to work on more than 5000 nodes, I ran into memory problem. The problem I really want to solve has 10000-15000 nodes. Does anybody know how to deal with this problem? Are there any other packages in R that can handle this problem?

...sage----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Foster Sent: Thursday, July 29, 2004 9:13 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] BugetTone Bug Showstopper, On Wed, 28 Jul 2004 23:31:06 -0400, Seth Remington <sremington@saberlogic.com> wrote: > On Wed, 2004-07-28 at 21:00, James Gardiner wrote: > > How do I get Asterisk to recognise the # key from the granstream phone for > > doing transfers? > Make sure the Grandstream is configured to send DTMF via SIP INFO > instead of in...

Hi folks I have more than 12 years experience with UNIX system administration, but I am too stupid for programming. My only programming experience is shell scripting. I tried to learn Java, but don't understand it because it is too complicated for my limited brainpower. What programming language should I learn? A friend said that C-Sharp (Mono) is very simple. Is this true? cheers Sven

Hello All, I saw an add in my latest Linux Journal advertising Sangoma's new "AA series" of FXO/FXS analog cards with on-board echo cancellation, but I can't find any information at all on them. Even the link given in the advertisement is a dead end as far as I can tell. Anybody else seen/heard anything about this? -Seth

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.

Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. Thanks in advance, Hamish ------------------------------------------------------------------- |

Well this is disappointing. Time to find somebody else... -- Wes -----Original Message----- From: NuFone Operations [mailto:support@nufone.net] Sent: Tuesday, April 18, 2006 3:44 PM To: wbaehr@totalmac.net Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate

...of conversions or weaves, but, to be able to handle a large number of conversions or weaves, I'm hoping that some faster code already exists (ideally calling C or FORTRAN). Any suggestions for going base10 to base2, base2 to base4, or digit interweaving would be appreciated. regards, Richard Remington -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe" (in the "body", not the subject !) To: r-help-request at stat...

Hi! I'm doing Monte Carlo analyses of the distribution of the t-statistics of the parameters of models evaluated with the lm( ) function. Is there an easy way to recover the t-statistics (similarly to using coef to recover the coefficients)? Thanks, joseph --------------------------------- [[alternative HTML version deleted]]

hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank

Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still

Hello, First, sorry for my english. I'm a french student. I have a problem with asterisk. I use Budgetone SIP phones. When I dial 555 (VoicemailMain), I hear "You have 5 new messages, 1- Read your messages, 2- , etc ... ) But when I dial 1 or 2 or everything else, nothing happen. Are they some lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas