similar to: New native assisted transfer (atxfer) usage info required

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

Eric Wrote: ----------- The trick is not to use options you don't understand. "show application dial" will show you what the t and T options are for. Most people use the transfer feature of their phone, rather than using the T/t hack on the Dial line. Sounds like you are using CVS-HEAD and so will have to configure stuff in /etc/asterisk/features.conf. /Snip/ Eric, Thanks for

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

Because some users have requested transfer beep confirmations I've switched our phones over to using the asterisk transfer feature instead of the built in transfer functions of the phones. While testing it was working fine, but I changed something in features.conf and suddenly any time I hit transfer (*2), I can only enter one digit before asterisk immediately tries to dial that extension.

Hi, I recently moved all of my conf files over to a new Asterisk 1.2.4 server and every works except the features enabled in features.conf. Was there a syntax chnage in 1.2.4? Or is there something else... Here is my features.conf: ******************** [general] parkext => 880 ; What ext. to dial to park parkpos => 881-890 ; What extensions to park calls on context

I need attended transfers, but I do not have time to talk to another extension and see if they accept the transfer, my features.conf is: [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 220 ; Number of

Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set "atxfer = *" (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to

Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=>parkedcallsexten => 4000,1,Dial(SIP/4000,60,tT)exten => 4001,1,Dial(SIP/4001,60,tT)exten => 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext => 700 parkpos => 701-720 context => parkedcalls [featuremap]blindxfer => # disconnect => *0

Hi - I'm trying to implement the applicationmap stuff in features.conf, and I can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom IP500s and Snom190s. My features.conf looks like this: [general] parkext => 700 parkpos => 701-720 context => parkedcalls parkingtime => 240 transferdigittimeout => 2 ;courtesytone = beep

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here