similar to: Passing arguments to AGI script

So, if I have a 50,000 flac files with perfect metadata encoded with <=1.1.2, I should... find . -type f -name "*.flac" -exec flac -8 {} \; -print And come back to my computer in about 24-48 hours? I've tested it on a small dir tree and it seemed to be fine. I just wanted to make sure though, before having to re-tag a couple terabytes of data. This is actually a serious

How do I make multiple console connections to one virtual machine? If I try to connect two different terminal windows to one virtual machine, they both connect to the same console. Is there a way to get more than one consoles to a single virtual machine? I cannot use a network interface to accomplish this either, so ''telnet in'' type solutions won''t do me any good.

hi i am installing scalix mailserver while installing RPM i am getting this error: Failed dependencies: compat-libstdc++-296 is needed by scalix-server-10.0.1.3-1.fc4.i386 libcrypto.so.5 is needed by scalix-server-10.0.1.3-1.fc4.i386 libgssapi_krb5.so.2(gssapi_krb5_2_MIT) is needed by scalix-server-10.0.1.3-1.fc4.i386 libical >= 0.24.RC4.20050413 is needed

Helo List, Sorry I missed the rest of my email in my previous post. Please see below. I'm having an issue using the AND (&) operator evaluation in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in two diferent contexts in the dial plan.

Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833.

Hi List, I'm having a problem with detecting incoming dtmf tones with chan_capi, using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting that the capi module will detect the tones, but it did not. I also set both to 1, expecting that the asterisk dsp functions will detect them but it did not either. Can anyone provide any ideas how to overcome this issue?

Hi Can someone please help with the following, We are using asterisk@home 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice

Sorry for asking this question again but before I blow 100 dollars on a X100P I need to know this info: So does "SET EXTENSION <new extension>" allow for you to set which extension the rest of the call will occur over? So if a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database

I am running AGI script written in C. My script gets triggered but there is no action. I have taken the samplec file from http://home.cogeco.ca/~camstuff/agi.html. it says launched AGI script /var/lib/asterisk/agi-bin/sample.c exec format error . Did I miss anything? I am running asterisk1.0.7 can anyone please help me on these. thanks, Somesh -------------- next part -------------- An HTML

I have a problem receving calls via the ISDN line, using the followin components Asterisk 1.0.9 with asterisk@home chan_capi-cm-0.6 AVM Fritz card datalink protocol = point to multimode I can make calls out with no problems so the issue is only incoming calls. When I make the call from an external line to the ISDN line connected to asterisk, I get a busy signal after about 5 seconds. I have

Hi Folks: I used that one example for AGI script on C web, only to fill the working with the Asterisk. I compiled and it worked great. I executed accidentally the ls -l command in directory where was the source and executable, I noted and was surprised that because the executable size was to further 20 times more than source. I executed the gcc -Os source.c -o executable.agi command several

Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier.

i've just spent the pass 2 days trying to get AGI to work with PHP; i made a lot of silly mistakes along the way which could have been avoided if only there were some kinda howto or samples. at the risk of looking stupid, i decided to shared my experience in hopes that it might help some newbie get going with PHP. 1. first order of business is to be aware of your php environment; i m NOT

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Helo List, I'm having an issue using the AND (&) operator in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in my dial plan. When a caller presses 1, yes is passed as a varialble and same when 2 is pressed a "no" is passed.

Hi List, I am experiencing an issue in a server that I have installed asterisk; configured an AVM FRITZ card to work with the capi module. Once istalled the card works perfect; however every time I reboot the machine I found that I have to re install the capi4k-utils before I can load asterisk otherwise the capi module will not loadup. Can anyone direct me in the right direction in order to