similar to: Attended Transfer (ATXFER) with CVS asterisk r 1_

Hi, I'm trying to make work chan_capi with last asterisk CVS. I installed last zaptel,libpri,last cvs ana patched chan_capi 0.35 with the patch kindly suggested me by Jason Williams: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 First I received error 127 that I resolved commenting the line CC=gcc-2.95 but now I have this error: chan_capi.c: In function `load_module':

I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. I know most people expect a good SIP/IAX phone to do the job but I think it's nice to be able to do attended trasnfers with a simple ATA-connected analog phone. I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works

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'

Hi, I have a problem for install chan_capi My pc: Suse 9.1, with asterisk current stable en cvs I have download http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz And the path from http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 And patch the chan_capi # cd chan_capi-0.3.5 # patch -p1 < chan_capi-0.3.5-patch.diff # make /for install My error

Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to

Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate("unknown", "unknown") does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa'

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 everyone, I'm trying the new atxfer functionality. All seems to work fine at the beginning, but there is no audio between the party at the end of the transfer. Plus, after that, even normal calls won't work well (they can't hangup!). I'm using the Asterisk CVS from 2005-01-10 with Asterisk-OH323. Here is my dialplan: [default] exten => h,1,NoOp(bye) 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

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

Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get "transfer" I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial("Local/18005558355@jesnjer-f97a,2", "/18005558355")

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

Hello all I found a bug in the chan_capi driver (really a not implemented message handling and then a false error condition) and I guess I have wrote a patch to fix it (basically I searched the internet for other capi open source implementation an borrowed the code snippet) but I don't know where to send the report and bugfix. I also found some miss-behaviours that I would like to share

Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon:

Hi, I successfully installed 2 avm card in my asterisk box but I'm unable to make call. My capi.conf is: msn=0721111,07211115 incomingmsn=* controller=1,2 softdtmf=1 context=default echocancel=yes callgroup=1 devices=2,2 my capi info : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. my extensions.conf : exten =>

I am trying out the new atxfer feature from CVS-HEAD. I set atxfer equal to *7 and it seems to work OK. I am having a problem getting it to work the way a receptionist would want. If an extension calls me, I hit *7 and I hear the voice say "transfer". I dial another extension. If the newly dialed extension goes to voicemail, I can't figure out how to get the original call

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 to all, I'm trying to make outside call in this way : ignorepat => 0 exten => _0.,1,Dial(CAPI/xxxxxxx:b${exten}) But the first number 0 is not ignored. I'm doing something wrong ? Bye

Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a

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