similar to: Call transfer questions

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

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

Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when

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

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

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? I Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So, I've configured some keys to transfer the calls like this: [featuremap] blindxfer => #2 ; Blind transfer (default is #) disconnect => *0 ; Disconnect (default is *) ;automon => *1

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

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

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 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]

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 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

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:

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

Hi, Is there any way to run a command (AGI script, whatever else) at that moment when the call that was in the queue is being bridged to a specific agent? An examples of what I would want to do with this is, for example, have Asterisk ask the caller for his 4 digit customer ID before being put in the Queue. Once I know who the caller is being connected to (which agent) I'd run a

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. 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 =>

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

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website