similar to: Graceful Asterisk Shutdown

Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Thanks, Jeng ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good

On Dec 10, 2007 10:36 AM, Jeng Yu <jengyu2007 at yahoo.co.uk> wrote: > Thanks, All! And thanks, Oquendo! I will experiment > with this suggestion. I was actually thinking in terms > of a situation where it would be done > non-interactively. > > Jeng > > > --- "J. Oquendo" <sil at infiltrated.net> wrote: > > > Jeng Yu wrote: > > >

Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: ---------------------------- /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field ?owner? specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: ***

Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___________________________________________________________ Yahoo!

Hi Asterisk Gurus! My lab asterisk box has 1 FXO and 1 FXS ports in it. I connect a GSM phone to the FXO port. I connect a regular corded phone to the FXS port. The Dial() application for both incoming and outgoing calls specifies the A(hello-world) flag. From another GSM phone, if I call the extension (corded) phone attached to the box, it plays the hello-world file when I pick it up. But

So I was bored yesterday and tried solving a few problems with one stone: 1) Notify me of potential brute forcers (multiple attempts to register multiple numbers from one address) 2) Notify me of (l)users who are having password issues So I whipped up a simple script to run in cron and notify me that UserX from X_IP_Space had X amout of password issues. I'm currently running this from cron

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My

Has anyone created like a broadcast dialplan, if so care to share it. What I'd like to do is create an extension so when someone calls that extension they can leave a voicemail. Right after it is recorded, I need that voicemail to played on all phones on that system... E.g.: 1) Administrator --> Dial special number 2) Record emergency message (e.g. Snow day don't come in) 3) Hang up 4)

described (stop accepting calls and shut down when all calls have completed). If you don't want to stop accepting calls, but still want to stop Asterisk when there are no active calls, you can use "stop when convenient". The same qualifiers ("gracefully" and "when convenient") can be applied to the "restart" command. Cheers, AR On Dec 10, 2007 7:29 AM,

Hi People! Is there an underground asterisk command reference manual that the Gurus here share amongst themselves only? :-) The reason I ask is that sometimes I see mention of an asterisk command and I scramble for my asterisk book (pdf) to look it up but can't find it in there. For example, I saw here last week people talking about the Set() application with the "If" conditional

Hi Gurus! Please excuse this pesky Asterisk rookie....:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. Thanks, Jeng ___________________________________________________________ Want ideas for reducing

Hi All, The Asterisk book (the pdf version) is excellent!! I want to thank all the guys that put it together. I am most grateful for it. There is something about writing a dialplan that I'm not clear about. What I'm trying to figure out how to do is this: when I transfer a call to the destination number, and the person called picks up the call, I want to play a greeting message to the

Hi All, Can I overlap Playback() with Dial() in a dialplan? For example, I have this scenario: A call comes in, Asterisk picks it up, does Background(enter_number), then does Playback(bulletin_message), and while the Playback() is still going, I want to execute Dial() to the target extension so it overlaps with the Playback() and the call will be bridged instantly upon completion of Playback().

My Esteemed Gurus! Still learning... I need to read about and learn how to configure Asterisk box here in the lab for digital cards. I'm about to get one of those E1/T1 interface cards from Digium. Please point me to docs that I can read so I don't have to bug you with too many questions :-). Sample config files would greatly help. In particular, suppose I have a T1 card in my box

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

Hi Friends! I need help! I'm still Asterisk rookie, so please forgive me. My Asterisk is no longer answering incoming call on the phone line. I call the phone and it rings but asterisk is not picking it up. The phone line is attached to port 4 (FXO) on my digium TDM411P card. I am running Asterisk 1.4.11 with zaptel-1.4.5.1 and libpri-1.4.1 on Fedora Core 5, Linux Kernel

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Quick and dirty too hungover to think/search... ;) Sorry list AsteriskBoxA -> SBC/PSTN voodoo -> world wide Interweb -> AsteriskBoxB Account on BoxA UsernameJohn<12125551212> Account on BoxB UsernameJohn<102> They're both the same users, had to do some funky trunking (managed firewall provider is playing not in my backyard games)... So anyway, I need to take specific

Curious to know if anyone's created something similar to the following, if so and you'd care to share an AGI or dialplan, much appreciated. I will be eventually write a script to answer the nTH call. (if I can't find it (why reinvent wheels). Looking to do some testing sending anywhere between 50-200 calls to a machine. I'd like a Snom/Polycom/whatever to pick up after the nTh

Saludos, estoy intentando usar la función reprojectHDF() ( http://r-gis.net/?q=ModisDownload). #### source('ModisDownload.R') library(raster) library(RCurl) path<-"~/R/MODIS/Data/Test" input<-list.files(path,"*.hdf",all.files=T,recursive=T,full.names=T) input outname<-paste(substr(input[1],40,45),".tif",sep='') outname