search for: atxfer

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 see information in the CLI console, this command...

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 ha...

...; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8 ; Configure the pickup extension. Default is *8 [featuremap] blindxfer => # atxfer => * I have tried atxfer=># and atxfer=>*2 but nothing works, also, when I disable both blindxfer and atxfer, blind transfer still works. I have turned sip debug on, and I can see that asterisk does receieve the * keys parking calls does work btw. Kind regards, Joop

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 __ast_read: DTMF end '1' received on SIP/bmwgsjrponciuj-0000009f, duration 0 ms [Mar 22 15:46:27] DTMF[5910]: channel.c:3926 __ast_read: DTMF begi...

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 goe...

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/1...

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...

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+conf...

...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 to personalize the way of making transfers using the feature of atxfer. How can I do that? Thanks in advance.

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 successf...

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

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 hea...

...sk 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 => *0 automon => *1 and when I press *2 console says something like this: Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42 (*), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL...

Hello, Is it possible to set different features.conf dialing sequences (atxfer, pickup, ...) for different users ? For instance, what if I want Alice to dial *8 to pickup a call and Bob to dial ** to pickup calls ? I can see that features.conf includes application maps but can these be used for the above goal ? Cheers -------------- next part -------------- An HTML attachme...

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: [d...

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 the frames...

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 possible??? -------------- next part -------------- An HTML attachment was scru...

Hi, I cannot get attended working on my Asterisk 1.2.9.1 during an inbound call via an ISDN card to a Snom SIP phone. The called party is not able to transfer even if : 1 - atxfer is enabled (set to *7) in in features.conf 2 - the dial option is set to value 't' 3 - I see * and then 7 on Asterisk CLI when debug is set to DTMF Asterisk gets the right sequence from Snom phone (CLI does not lie) but for some reason Asterisk is not transferring the call while the calle...

...have to hangup but habe to press forward button on the phone. Can u check it ? This is my features.conf (only featuremaps section): [featuremap] ;blindxfer => ## ; Blind transfer disconnect => *0 ; Disconnect automon => *1 ; One Touch Record atxfer => ## ; Attended transfer 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: marted? 31 maggio 2005 16.21 A: asterisk-users@lists.di...

Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit "transfer" then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang