search for: jaymoor

...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 Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 >>> jaymoore@accu-com.com 4/3/2007 2:29:11 PM >>> Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the iss...

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

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

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 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

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

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

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

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

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?

Greetings, I am using MixMonitor to record my outgoing calls. It seems that MixMonitor will not write to a directory if it doesn't exist (ie - it doesn't create a new directory if needed). I have checked to ensure permissions are properly set, and if I manually create the directory, MixMonitor behaves normally. Rather than send several 'mkdir' commands each time I want to

Having a problem here that I can't seem to find a fix for. PSTN call comes in, operator answers, transfers call to a phone behind an IAXy. Caller hears no sound after being transferred. IAXy can hear caller, but not vice versa. Client reads: NOTICE[11342]: channel.c:1950 ast_read: Dropping incompatible voice frame on IAX2/jim-7 of format gsm since our native format has changed to ulaw

Hi folks. I'm having a problem with a SIP-enabled device that doesn't seem to want to register after it reboots. If I program the device manually via its interface, it registers just fine. However, once I reboot it, it fails to register with Asterisk, despite all the proper information being stored in its memory. I contacted the manufacturer of the device and was told to try this:

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

Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine and then a second or so of silence followed by the person talking again. I'm saving my calls as .gsm files and it's worked fine for the past 11 months. I make sure I remove the recorded files from my Asterisk box and put them onto our

Hi list. I'm new to IVRs and trying to set up one that toggles an auto-forward flag on or off for specific accounts. I'd like to have my users dial an extension and then be prompted to enter the account number. (done) Next I'd like it to jump to the appropriate line in the dial plan that corresponds to the entered account number (if it is valid) and have it play back the

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

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 access issues for local apartment complexes and businesses. Most of the time, we get