search for: transferring

...within the oVirt/RHV environment, eg. on an oVirt node Change the wait loop to consider all cases: - Transfer failed and was removed - Transfer failed and will be removed soon - Transfer paused by the system (cancel required) - Unexpected transfer phase (cancel required) - Timeout waiting for TRANSFERRING state (cancel required) Reported-by: Xiaodai Wang --- I could easy simulate the case when the system paused the transfer by injecting an error in vdsm, failing transfer initialization. The import fail with: nbdkit: python[1]: error: /home/nsoffer/src/virt-v2v/tmp/rhvupload.1DgXyh/rhv-upload-plu...

...nt, eg. on an oVirt node > > Change the wait loop to consider all cases: > - Transfer failed and was removed > - Transfer failed and will be removed soon > - Transfer paused by the system (cancel required) > - Unexpected transfer phase (cancel required) > - Timeout waiting for TRANSFERRING state (cancel required) > > Reported-by: Xiaodai Wang > --- > > I could easy simulate the case when the system paused the transfer by > injecting an error in vdsm, failing transfer initialization. > > The import fail with: > > nbdkit: python[1]: error: /home/nsoffe...

Hi, Need to get an answer to the following. I find the naming to be little redundant regarding the 'input.transfer.boost.low - hige'. The same goes for input.transfer.trim.low - high. As I find it, there is only one point where the boost or buck (trim) kicks in. Let's say that the nominal voltage is 230 volt, then we have a input.transfer.boost point on say 220 volt. If the

...> > Change the wait loop to consider all cases: > > - Transfer failed and was removed > > - Transfer failed and will be removed soon > > - Transfer paused by the system (cancel required) > > - Unexpected transfer phase (cancel required) > > - Timeout waiting for TRANSFERRING state (cancel required) > > > > Reported-by: Xiaodai Wang > > --- > > > > I could easy simulate the case when the system paused the transfer by > > injecting an error in vdsm, failing transfer initialization. > > > > The import fail with: > > &g...

Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer

I have a query that is intended to find all "transfers" based on a condition the uses a second table. In the controller, it looks like this: def find_protocols @transfers = Transfer.find(:all, :from => "transfers, protocols", :conditions => "transfers.protocol_id = protocols.id AND protocols.name = \"#{params[:protocol]}\"") respond_to do

Hello, I tested to transfer all 7 FSMOs from Windows Server 2003 Enterprise to Samba 4.3.0 DC. (I think some users still use Win2K3 and want to migrate.) But unfortunately is failed. To reproduce, (1) Promoto Win2K3R2 Enterprise host to first DC. (2) Join self-built Samba 4.3.0 to the domain. (3) Run 'samba-tool fsmo transfer --role=all' root at jessie64-1:~# samba-tool fsmo

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. -- -Barry Flanagan

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111116/c...

Fix NameError introduced by inocomplete change to extract create_transfer() by extracting cancel_transfer() function. Finally extract finallize_transfer() function, dealing with the poorly documented and over complicated oVirt SDK. Tested with: libguestfs-1.40.2-4.fc30.x86_64 nbdkit-1.12.8-1.fc30.x86_64 Tested flows: - Successful import - Error in close() - by injecting long sleep in vdsm

We can, and do, suspend a TFTP transfer simply by delaying the ACK packet as necessary. On September 29, 2015 7:45:27 PM PDT, Patrick Masotta <masottaus at yahoo.com> wrote: >>>> >On 09/26/2015 10:36 AM, Patrick Masotta via Syslinux wrote: >> >> About PXEbc Peter has recently said: >>> The problem with the pxebc is that it only supports one concurrent

Hello, > On 07/10/15 00:33, TAKAHASHI Motonobu/高橋 基信 wrote: > > Hello, > > > > I tested to transfer all 7 FSMOs from Windows Server 2003 Enterprise > > to Samba 4.3.0 DC. (I think some users still use Win2K3 and want to > > migrate.) But unfortunately is failed. > > > > To reproduce, > > > > (1) Promoto Win2K3R2 Enterprise host to first DC.

Hi All, Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Potentially using a mix of phones would create confusion in a user base, any ideas on attended transfer or how to achieve this / mods to dial plan etc would be greatly appreciated. I have been on an almost vertical learning curve with Asterisk and Linux for 6 months this is just

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

After invoking transfer_service.finalize, check operation status by examining ImageTransferPhase and DiskStatus. This is done instead of failing after a predefined timeout regardless the status. * not verified * Bug-Url: https://bugzilla.redhat.com/show_bug.cgi?id=1680361 --- v2v/rhv-upload-plugin.py | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git

Hi Kevin, I haven't specified --whole-file. After entering an rsync command the terminal always reads "delta-transmission disabled for local transfer or --whole-file" but I assume that is just a standard phrase that always appears. So, if I am running partial (-P) and not using --whole-file or disabling the delta-transmission, why would an incomplete file be deleted and the

Hello, I’m running rsync -avPhi to move large video files to a remote server. Often we have to stop a transfer midway through to push something else to the server. My hope was that the -P flag would invoke --partial and the transfer would pick-up where it left off. This does not seem to be happening. Rather, the transfer starts over from the beginning, erasing the partially transferred file.