similar to: Transfering not working - how to debug?

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

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

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, 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:

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

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

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.

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 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

Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer.... also can

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to doing a transfer from an analog extension to a SIP extension but until the moment I was not successful. I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer => #1 atxfer => *2 verifying previously that the extension uses the arguments "tT" with the Dial

I have a problem with my Asterisk system. When I use my phone to call my office mailbox I have to end my password with #. (The office do not use Asterisk) " # " is also used as a transfer button on my asterisk, so when I press it I hear my Asterisk trying to transfer the call. Is there any way to change the transfer button or remove it ? Fredrik

Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently

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

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

Hi, I have been trying to enable attended transfer for callee. When the callee pressed *2, DTMF tone was heard by the caller. But when the caller pressed *2, attended transfer started. It's strange. I used two SIP phones. My Asterisk version is "Asterisk CVS-HEAD built by root@router on a i686 running Linux on 2005-06-27 06:07:18". In features.conf, I have: [featuremap]

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

Hello, while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers a "featuremap" to assign a key to blindxfer and atxfer and they work fine if the call is still in the same starting context, but if the call has moved in another context, then the new call will be started from such context with unpredictable