similar to: Call recording - methodology

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it

Hi, could anybody suggest a usable doorphone and magnetic door opener "hardphone" system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. thank you, Csaba

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

List, In order to get rid of some old, unreadable S3 code in package sp, I'm trying to rewrite things using S4 methods. Somewhere I fail, and I cannot sort out why. In order to isolate the problem, I created two functions, doNothing<- and dosth, and both should do nothing. The issue is that in most cases they do nothing, but in some cases dosth(obj) changes the class of obj and breaks with

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

Hi everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150

I am unable to source a ".R" file using RJava. I tried a couple of different tests: 1) using java and the evaluation method core dumps 2) using ./RJava --example --gui-none to invoke source core dumps. 3) The line of R works if I go directly thru R and not RJava. Version info and code are below. Any help would be appreciated. --Laura O'Brien Applications

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi all, >From the manpage of rsync, I can see the following descriptions: --inplace The option implies --partial (since an interrupted transfer does not delete the file) So I do the following testings on the `--inplace' and `--partial' for resuming a file with the following steps: 1- rsync ftp.cn.debian.org::debian/dists/wheezy/main/binary-amd64/

I''m following the depot application in the rails bible Agile Web Development with Rails. In interation E1 NoMethodError in Admin#checkout Showing app/views/admin/checkout.rhtml where line #12 raised: You have a nil object when you didn''t expect it! You might have expected an instance of Array. The error occured while evaluating nil.include? Extracted source (around line #12):

Hello, My host just had a faulty power supply and therefore, my Asterisk server was down for 7 hours. It was a Sunday so no one was making calls, however if it happened during the week, I'd have problems. I was trying to find a whitepaper or advice on how to set up two Asterisk servers to provide some redundancy. I've been googling "asterisk redundancy" but all I've found

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

+the people I hashed this out with so many months ago I think it's a reasonable proposal, but obviously I floated it. :) Let's try to get a second opinion. Again, it's a syntax something like: define void @foo() prefix [i8* x 2] { i8* @a, i8* @b } prologue [i8 x 4] c"\xde\xad\xbe\xef" { ret void } On Thu, Aug 21, 2014 at 1:58 PM, Ben Gamari <bgamari.foss at

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher

Following patch for pfile.rb and pfile/source.rb allows the user to set a replace => false parameter on a file sourced by puppet, but not replaced if checksums do not match. This is for cases in which it is desired to distribute initial "bootstrap" files and ensure future existence, yet allow them to be modified on the node. RTS --- pfile.rb~ Wed Aug 2 04:47:05 2006 +++ pfile.rb

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.