similar to: LD accontability

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

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

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:

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

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.

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:

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

hello i am trying to make a callback solution. client will call callback number and call is terminated. now callback server will create a call for that client. actually i have a problem in this process. that server is creating call to client (UA) when previous call is not disconnected yet. UA---------->Asterisk(callbacknumber) callis answered UA<----------Asterisk(callbackserver) call is

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

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

My configuration for my CAC is 12FXO/12FXS. A few days ago, I was looking at my zapata.conf and started wondering if 1 maybe my configuration could be off. The following is my current zapata.conf: ; Configuration file ; Set initial values for all channels in the channel ; bank, and apply those settings to channels 13-24. ; ; This is "group 2", and is indicated by g2 in Dial ;

I have been using the ATA-186 with good success (with exception of the fact that you have to recycle it from time to time). The one thing I haven't been able to do is to figure out how to make use of parking, transfers, etc. My cordless doesn't seem to pass DTMF, so I haven't had any success with parking. As for transfers, I can do a flash and get dial tone, but then what are my

I was finally playing around with the various databases, blacklist and cidname. The format for these are family name, key and value. Would that mean that the key is the name to enter, and the value the number? For instance: database put blacklist "Kim Callis" 3235551212? And if that is in fact correct, what happens if I don't add a key? -------------- next part -------------- An

Several people have asked about integrating asterisk in a company that currently has a PBX installed. Considering the capital outlay for most PBXs, depending on the model and age of the system, most companies do not want to look at replacement. So, what should one look at in a PBX in order to integrate asterisk into a pre-existing PBX? I have read of some systems with T1 interfaces, but what

I've installed Warcraft3 Reign of Chaos with wine 0.9.46 nut when i start it , it crashes.The error is: This application has encountered a critical error: FATAL ERROR! Program: C:\Programmi\Warcraft III\War3.exe Exception: 0xC0000005 (ACCESS_VIOLATION) at 0073:7D477AAC The instruction at '0x7D477AAC' referenced memory at '0x00000034'. The memory could not be 'read'.

sorry, but im new to linux and wine, but i was reading the tutorial on how to install steam and the third step said: Install Gecko Engine by running -wine iexplore http://winehq.org- what dose it mean to "run" something? thanks for any replys -REIGN

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 --

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