search for: calli

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 2002, Vikram Adve wrot...

...ewhere in this great Asterisk community there are some clever people who might have some good suggestions to help me improve the voice quality on this system. I have a phone switch configured with Asterisk Business Edition (B.2.3-2-1) and I have installed a single license for the Cepstral voice, Callie. I am using the Swift application call within the asterisk dialplan: Within /etc/asterisk/extensions.conf -- exten => s,n,Swift(Callie^Hello this is a cepstral voice named Callie.) And the php class phpAGI2, for interfacing with the Asterisk Gateway Interface: Within /v...

...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

...issing, with no default Enter a frame number, or 0 to exit 1: test(5) Selection: 1 Called from: eval(expr, envir, enclos) Browse[1]> Recall() --- And here the R GUI crashes, i.e a dialog pops up saying that R has encountered an error and must close. I'm not sure what I was thinking when calling Recall(), but I shouldn't be punished this harshly for being stupid, methinks. best, Gustaf -- email:gustaf.rydevik at gmail.com tel: +46(0)703051451 address: Kantorsgatan 50:190 75424 Uppsala Sweden

As I was working on my extensions.conf file, I started to segment calling privileges. For the everyday workers, I don't free reign to LD access unless it is business related. So I was wondering if there was a way to implement some type of accounting code to be entered before accessing LD, which of course would be noted in the CDR (however it is implemented, either...

...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 mu...

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 scrubbed... URL: http://lists.digium....

...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

...gh the if statement, it might be useful to convert this CFG into a diamond and hoist that portion of block C into both block B and the new block. Consider the following example: int test(int *Ptr, int a, int b, int c, int d) { int Ret = 0; if (__builtin_expect(!Ptr, 0)) { Ret = calli(a); // return Ret / (a|1) / (b|1) / (c|1) / (d|1); // Copy return to here } return Ret / (a|1) / (b|1) / (c|1) / (d|1); // This can be simplified to return 0 } In this case, replicating the return statement in the branch allows the optimizer to prove the value of Ret at the end of the f...

...apper(address space) plus the wrapped address node. What has me mystified is that in some of the patterns in MSP430InstrInfo.td take the form def : Pat<(i16 (MSP430Wrapper tglobaladdr:$dst)), (MOV16ri tglobaladdr:$dst)>; and others ... def : Pat<(MSP430call (i16 tglobaladdr:$dst)), (CALLi tglobaladdr:$dst)>; In other words, some patterns rely on MSP430Wrapper being part of the pattern then extract the wrapped info anyway, and some others just directly match tglobaladdr and friends. I have noticed that many other backends use the same idiom but so far I can't see what it...

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()); DirectF->setName(CI->getCalledFunction()->getNameStr() +...

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.