similar to: 1.4.36 - Warning Dropping incompatible voice frame on Local/ on multiple atxfer a->b->c...->d...

Guys, I'm trying to compile the Xen 4.0.0 via debian/rules makefile but I got this: administrativo at sid01:~/xen/xen-4/xen-4.0.0$ make -f debian/rules build-tools ..... make -C lib all make[7]: Entering directory `/home/administrativo/xen/xen-4/xen-4.0.0/debian/build/build-tools/tools/blktap2/vhd/lib' make[7]: Nothing to be done for `all'. make[7]: Leaving directory

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s,

WorkShop Virtualiza??o com Proxmox VE ------------------------------ <https://www.facebook.com/sharer/sharer.php?s=100&p[url]=http://blog.konnectati.com.br/arquivos/1276> Dia 12 de Julho, S?bado, a partir das 08hs00 acontecer? o WorkShop Virtualiza??o com Proxmox VE. Venha conhecer uma das melhores ferramentas de Virtualiza??o do Mercado. Com este treinamento, sua equipe de TI ter?

Voc? tem uma nova mensagem no Badoo! Gilberto Nunes deixou uma mensagem pra voc?. Clique no link para abrir: http://us1.badoo.com/01097897898/in/BDGmMoCC9H8/?lang_id=61 E, outras pessoas estiveram procurando por voc?: Manoel Felipe (Joinville, Brasil)Mara Leal mendes (Joinville, Brasil)Tiago Gabriel (Joinville, Brasil) Se os links desta mensagem n?o funcionarem, copie e cole os links na barra

Hi! I compiled the xen-4.0.1 in Linux (Ubuntu 10.04), but as usually, the xen-4.1.0 is not behaving in a manner similar to the previous version. Am I required from version xen-4.1.0, transfer the kernel via the command "git clone git://git.kernel.org/pub/scm/linux/kernel/git/jeremy/xen.gitlinux-2.6-xen" from now? I mean, the "make prep-kernels" command of the xen-4.1.0 no

Humm, thanks for your reply But whats is the code in parkedcalls context. Please, can you give an example? Thanks very much. 2016-02-03 17:15 GMT-02:00, Richard Mudgett <rmudgett at digium.com>: > On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco <vitor.mazuco at gmail.com> > wrote: > >> Hi! >> >> I tried to use Parking Calls >> >> I use Asterisk

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 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 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")

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, 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

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'

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

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