search for: callie

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:

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

...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 /va...

Is it just me or do others have a problem with the ATA-186 de-registering? Every 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

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system 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

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"

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 comma delimited or MySQL).

Hi! Sure, just look for: Wonder Shaper. It's a HTB based shaper configuration wich have some very good features, I use a variation of that here at my College. http://lartc.org/wondershaper/ 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

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

...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 --

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

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:

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>

Hello, I have been wrestling with installing the CAPI drivers for AVM Fritz in order to use chan_capi with Asterisk. I use an SMP machine, RH 9. I have found rpm's for CAPI and 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