similar to: "Non-blocking" Dial (and other commands): is there a way?

(Cross posted to speex-dev from iglance) Enzo -- I haven't tried the fixed point engine, though I've considered it for the WinCE port. For the desktop/laptop edition I'm assuming the slight short->float conversion cost will be made up by the improved performance of the floating point implementation. But I could be wrong: 1) Can anyone recommend whether Speex performs better

When Asterisk uses an ISDN interface, it periodically sends to CLI messages such as: == Primary D-Channel on span 1 down [...] == Primary D-Channel on span 1 up Is there a simple programmatic way of capturing them for monitoring purposes? Enzo

Hello, I am having trouble getting the output from the tapply function formatted so that it can be made into a nice table. Below is my question written in R code. Does anyone have any suggestions? Thank you. Geoff #Input the data; name <- c('Tom', 'Tom', 'Jane', 'Jane', 'Enzo', 'Enzo', 'Mary', 'Mary'); year <- c(2008, 2009,

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

On Monday 20 Dec 2010, Olivier wrote: > 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk> > > > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application > > (written by someone else before me) which sets up calls by creating > > files of > > the general form > > > > Channel: SIP/$INSIDE_NUMBER > > Context: $CONTEXT >

I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I have just about everything working except for the message waiting indicator. I have the following setup in context [ccm] in my extensions.conf file: ;MWI exten => _2807XXX,1,SetCallerID(${EXTEN:3}) exten => _2807XXX,2,Dial(SIP/28888@65.202.115.240) exten => _2807XXX,3,Answer exten => _2807XXX,4,Wait,1

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

I am sure this is simple, but have been struggling. I want to create a call file that dials out a particular Dahdi channel to enable call forwarding on a POTS line. I have this in extensions.conf: [custom-callfwd] exten => s,1,Answer exten => s,n,Dial(DAHDI/4-1/*717157750) exten => s,n,Verbose(${DIALSTATUS}) exten => s,n,Hangup [custom-callfwdcanc] exten => s,1,Answer exten

Hello, I am at my first attempts with scripting, cron, and rsync, so please bear with me... The scope is to backup two servers from a dedicated backup pc, with the following script: mv *.log archive rsync --log-file=oracle.log -av -e "ssh -i .ssh/rsync-key" gian@oracle:/home/gian/exp* oracle/ rsync --log-file=vib_home.log -av --delete --exclude=".*/" -e "ssh -i

Hi! I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt. I put here my callfile and that I get when asterisk begins to do the call If anybody has idea, pls. Tell me TIA ;;----CallFile----- Channel: Zap/g1/8093908270 Callerid: 8093908270 MaxRetries: 2 RetryTime: 300 WaitTime: 45

Like some other people on here, I am trying to integrate Asterisk for VM with CCM version 3.x. I've got gnugk and Asterisk running, I've got CCM registering with the GK, I've got the voicemail pilot and profiles setup. A call comes into a CCM phone, it rings, rolls to the correct VM on ASterisk and asterisk emails the voicemail and I can check the voicemail, but I cannot get MWI

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

Hi I'd like to you know if this is possible to do this with swfdec, or even to know if one day, it will be possible I have a swf player that grabs some piece of sprites, music and move all this in some kind of a cartoon Is it possible with swfdec to record this cartoon into an avi file ? Thanks Enzo ___________________________________________________________________________

I’ve installed R 3.1 with the latest Fedora rpm (R-3.1.0-5.el6.x86_64.rpm) and I run into some issues with openblas. I’ve documented this with an open question on stack overflow here: http://stackoverflow.com/questions/24158372/openblas-r-3-1-and-fedora-centos-dist Basically before 3.1 I had 3.02 and I was able to install and use openblas following the instruction from official CRAN

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

Hello, I'm using Asterisk 1.6.2.0 and I like to call extension #8 from callfile. Unfortunately the #-char ist interpreted just as comment. I got a "Invalid file contents in /var/spool/asterisk/outgoing/callfile, deleting" from asterisk. When I try to escape with \ oder use quotes, I got: \#8,1 failed so falling back to exten 's' or "#8",1 failed so falling back

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider