search for: callis

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030710/b81442d3/attachment.htm

LLVM, What do I pass into the DSG in order to access the globals vector of functions that a function pointer may be calling. The code: CallInst *calli = dynamic_cast<CallInst*>(*i); std::vector<GlobalValue*> funcVect = theGraph.getNodeForValue(calli->getCalledFunction()).getNode()->getGlobals(); Doesn't appear to work... getCalledFunction() returns 0 Dave On Fri, 6 Dec

Hi all, I hope that I'm not breaking protocol too much by posting a message in this group about a problem that I'm having with an Asterisk Business Edition installation, but the reason that I'm posting here is because the problem that I'm having isn't really with the Business Edition, it is with the Cepstral text to speech product that I'm using with it, and also because

...couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030702/49736fd1/attachment.htm

...em from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or maybe have a call back with a DISA and then just dial my phone number I am trying to reach. Just a thought! Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030717/2d13207b/attachment.htm

Hi all, I know I shouldn't really expect the following to work, but it provokes a crash of the GUI on my computer (Win xp professional). --- >sessionInfo() Version 2.3.1 (2006-06-01) i386-pc-mingw32 attached base packages: [1] "methods" "stats" "graphics" "grDevices" "utils" "datasets" [7] "base"

...roceeds, if not, a message is played for the tech person to the effect of something like, "That account code is not valid" and notates that in the cdr as well. This is my view of how it would work, but if anyone has another way to provide accountability that would work as well. Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030704/40e64092/attachment.htm

...haper/ It is the page (a simple google search). Also make sure to uncomment the line tos=lowdelay in every config file of asterisk that have it. Hope it is usefull, sincerely, Ildefonso Camargo Administrador de la Red de Datos - UNET icamargo@unet.edu.ve >Message: 2 >From: "Kim C. Callis" <kcallis@c2associates.net> >To: <asterisk-users@lists.digium.com> >Date: Tue, 22 Jul 2003 20:09:49 -0700 >Organization: Callis, Chung & Associates, Inc. >Subject: [Asterisk-Users] QoS for Asterisk >Reply-To: asterisk-users@lists.digium.com > >This is a mul...

Hello,I was using the ResetCDR() right after my IVR answered the calls to set them unanswered until or if somebody pick up the callI just upgraded to version 1.4.19.1 and now ... when the ResetCDR() function is called it seems to just erase the recordIs there another to set the status of the call unanswered ? is it a bug ?Thanks, -------------- next part -------------- An HTML attachment was

Steve Prior's mention of using Allison's voice with Cepstral reminded me to ask: for a listing of the text for the built-in recordings. I found a web page but I'd prefer not having to scrape the info out of it. I didn't notice anything while wandering through the source code/files. I want to rebuild them for my system using Callie's voice. Thanks, Rod --

...Here" back from 66.12.5.14 -- Got SIP response 488 "Not Acceptable Here" back from 67.15.17.22 So is there something I need to set on the 7960s in order to make use of that particular codec? Am I missing something in the compiling of *? Any help would be very appreciated! Kim Callis

...meone attempted to start flaming Asterisk, and swearing by VOCAL, while another was babbling about the wonders of Bayonne. The only thing that was successful in that meeting about VOIP solutions was tabling that discussion until a future (as in way, way in the future) date. Just a thought! Kim C. Callis

Hello, I am having a problem with a new server running FreeBSD 4.9-PRERELEASE. Once every two days or so, it reboots itself.. there is no exact time of the day it reboots.. It rebooted itself at 4:25 PM EDT on 9/25, and 3:05 AM EDT on 9/29 (today). I thought it's the bad memory or some sort, so I've been running memtest today and memtest reports no sign of trouble with the memory...

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030718/42e33864/attachment.htm

Hi All, We have recently found some optimization opportunities created by replicating code into branches in order to enable optimization. In general, the optimization opportunity we are pursuing is like the following. Given pseudo-code: // block A if (some condition) // block B // block C If it can be efficiently proven that some portion of block C can be simplified had control flow not

Hello, I'm considering creating an LLVM backend for a 16 bit processor and modelling it around the (experimental) MSP430 back end. When reviewing MSP430InstrInfo.td I see def MSP430Wrapper : SDNode<"MSP430ISD::Wrapper", SDT_MSP430Wrapper>; and can see in MSP430ISelLowering.cpp that ISD::GlobalAddress: ISD::BlockAddress: ISD::ExternalSymbol all get lowered to

Hi, I want to clone a given function, and add an argument to it. I then want to add a call to that new function. I have a callInstruction CI, which I want to transform to call this new function, and to take a new argument. The code I added was as follows CI->getCalledFunction()->dump(); Function* DirectF = CloneFunction(CI->getCalledFunction());

Hi, all I want to create a function in LLVM IR, whose type is: void _to_prof( FILE* ptrF); I do it within LLVM, but I don't know how build the parameter type for FILE* ptrF. Best Regards. Eric Lu -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20140627/c49bdb2a/attachment.html>

...AVM drivers (namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8- avmfcpci-03.11.02-08.mungo.RH9.i686.rpm), but I believe that they support only single processor machines. I've already spent too much time with chan_modem which gives me problems (like no audio until the callis answered, or kernel crashes probably because of the isdn4l drivers). So, I can't afford to go back to isdn4linux drivers and the HiSax card that I used. Does anyone have RPMs or source code that I can use for Fritzcard (PCI) and SMP? Thanks in advance for any assistance, Vlasis.