search for: flashhook

...th my problem. I can't get the verify.agi script to work in Asterisk This script is available for download at http://www.sineapps.com/downloads.php The agi script works for recording and playback when accessing it directly at it's extension, but will not record anything when doing the flashhook procedure during a call. Recording is cut off after the flashhook Thanks for any pointers (or workarounds) and help , much appreciated rdgs Derek

I've seen a lot of traffic on the list recently about which phones can do supervised transfers and which cannot, and I have to admit that I'm a bit puzzled. Our existing PBX, which is software based, handles the transfer functions for our call center -- the agents never touch their phone, and instead use software. We can plug any old phone into it, and it'll work just the same. So

Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages

Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb

Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I

Anybody know anything about using "Asterisk vertical service activation codes" as described in the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+vertical+service+activation+codes Specifically I'm interested in *0 that (apparently) flashes an external trunk on bridged channel. Nothing seems to happen when I use it though. Do I need to do something special in

Hi everyone, I have encountered a hard problem that when i connect my anology phone to channelbank ,I found that i dial a number and create the call,then ,I hangup the call,but ,very quickly,I listen the ringing im my phone,I pick it up ,and found it noting, anybody can tell me this reasons,and how to solve it,Thanks! -- Best regards! jordan pan Location:Shenzhen China

...ces hanging off the adtran. The digits are announced to the fxs devices between ring 1 and ring 2 (If I pick up the anlog phone after ring 1, I can hear them so I feel good about this stage). Problem is that my analog device "once it is finished with the call" needs to be able to "FLASHHOOK"<exten> to transfer the call to a SIP Handset. Problem is that as soon as the FLASH (which is 700milli-seconds) is heard (and I can hear this by BARGING into the channel and listening) asterisk hangs up. I have tried all combinations of busydetect ( and and off) as well as every combin...

Hi All, Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Potentially using a mix of phones would create confusion in a user base, any ideas on attended transfer or how to achieve this / mods to dial plan etc would be greatly appreciated. I have been on an almost vertical learning curve with Asterisk and Linux for 6 months this is just