similar to: Recorded calls skipping

I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay

Howdy, folks. I'm having a problem finding a way to auto-record calls (both incoming and outgoing). I know how to make it so either party can initiate recording, but I want it done as soon as both ends are connected (or prior to that if that's what it takes). It's probably right in front of me and I'm just missing it. Any help would be much appreciated. Thanks, Jay

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

How do I include a file (not a context) in AEL? #include "filename" returns an error. Thanks, Jay

Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my

Greetings, List. I have my Asterisk box setup with 8 Centrex lines that were "left over" from our old PBX system. My boss is asking me to set up Asterisk so that he can flash hook and make an outgoing call on the same line to have a 3 way call. This is what he wants to do: 1) Incoming call on his Centrex line 2) Flash hook and dial a new number (goes out the same line) 3) Flash

Greetings, folks. I'm having a problem with blind transfers. It seems that, despite not having the T flag set, callers are able to use the blind transfer option. Scenario is this: - Asterisk 1.2.14 - Caller calls into our call center on one of our many phone numbers. - Call gets placed into queue. - Operator answers call. - Caller is able to hit our blind xfer key sequence (#0) and dial

Here's how it's currently working: 1) Call comes in 2) Operator parks call (700) 3) Operator picks up call on another phone (701) 4) Operator tries to transfer to a different phone (we use #0) but the transfer doesn't work. We can transfer initial callers all we want and it works fine. Once a call is parked, however, we can no longer transfer the caller. Any ideas? Thanks, Jay

Before I make any serious gaffes, is this an acceptable place to post PHPAGI questions as well? I can't seem to find a dedicated mailing list for it. If not, any suggestions? Thanks, Jay

I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or C?dric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

What is the best alternative for getting the IVR and other prompts recorded for Asterisk. Regards, Sanjay.

Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what

Not sure if my original message made it through. Going to try this again. :) ------- Greetings, List. I would like to implement a procedure in my call center but am not sure the best way to implement it. I'm hoping I can describe it here and that I'll receive some feedback and/or suggestions on how to proceed. Here's my situation: My call center fields calls regarding internet

Greetings folks. I seem to be having a problem where calls made from an IAX device (three single-line phones attached to IAXys) do not play the ring tone when calling out. There's a dial tone when I pick up the phone, and the call goes through just fine, it just doesn't ring. All my SIP phones ring normally, however. Is there an option I need to enable that I'm missing?

This may be a dumb question, but I'll ask because... :) I've got dovecot installed and running successfully on OpenBSD 3.9 (this *really* was an easy setup, my hat's off to the developer(s)). And the other employees can now use MS Access to read their mail (they didn't like mutt and pine via SSH - go figger'). So my final challenge is to get things configured so that

Hi list I''ve got a bit of a philosophical question. After just having discovered Selenium /Selenium IDE and the rails selenium plugin I''ve realized that it''s a whole lot easier to do functinoal testing than writing the traditional rails functional tests. However it implies that the development database consists of standardized test data (e.g the test & dev

Hello,For the time being I am using the following line to play the original saved message by Asteriskexten => 7001,n,Playback(vm-nobodyavail)Now I am trying to use the other features for Asterisk's voicemail. I have recorded a message, and I can see it saved on the system, but still Asterisk keeps playing the original message... Is there something I can add to let the subscriber plays his

Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI> moh reload voip*CLI> 1