similar to: ATXFER discussion, what's your opinion ?

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'

Hi All, I installed qcc package and the dependency packages. For the first time I can use the function : process.capability.sixpack(). But later when ran the code again I always got the following error: Error: could not find function "process.capability.sixpack". I tried reinstalling the qcc package but didn't help. Does anyone have this kind of experience? Thank you! Regards, Yan

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'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 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

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

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

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 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 Folks, A couple of questions about making SMB (3 or 4) authenticate to an external (anonymous) LDAP server: 1) A typical LDAP user record is below. Is there anything lacking in this record that would prevent Samba from authenticating against our LDAP server? Note the sambaSID is as is, gobblygook info: dsAttrTypeNative:eduPersonAffiliation: Employee Member dsAttrTypeNative:givenName: David

Hy everybody, I'm using rsync to backup/synchronize folders to/from USB connected external hard drives. But I can't find an answer to a doubt. Does rsync use encription also for local tranfers? For "local transfer" I mean a transfer that doesn't go through a network like folders synchronization with external hard drives. I'm asking because the speed of local transfers

On Mon, Aug 07, 2023 at 04:00:49PM -0700, Chris Maness wrote: > When are these likely to hit the mainline release code? linux-next tomorrow. The first 7 patches are scheduled for mainline as they were merged + tested without any hiccups. These last few patches I'll wait and see. If nothing blows up on linux-next perhaps I'll include them to Linux for mainline during the next merge

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

Hi, I'm trying to make a attended transfer through AMI. I though i could use Atxfer, and it seems ok, but nothing happens. And I can't find any how-to or description on how to do this. What more do I have to do to make this work? In Asterisk Call Manager: Action: Atxfer Channel: SIP/36-xxxxxx Exten: 33 Priority: 1 Context: Phone Response: Success Message: Atxfer successfully queued

Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye

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 there i just can't find an answer on the featuremap config i want all phones to use the same method for transfering a call on all phones but i just can't get the atxfer or other functions to work on my grandsteam and sipura spa 2000 it's confusing for users with different phones to transfer a call i know you can use the transfer button but i wan't to use a code *1 not

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 =>

Hi. I've been trying to develop this module since some time now. CVS already has a dial version with atxfer. When trying this, using the modifiers tT and having configures features.conf accordingly, i havent been able to use such a feature in the called party. I also tried using t and T separately. I've tried to understand why this happens, and started to watch the "copy" of