similar to: Problem with attended transfers...

I'm trying to get ChansSpy to work. It works, in the pass/fail sense, but it is difficult to understand the various speakers. I can hear users on our end just fine, but the other end sounds like their going through a vocoder, if I can understand them at all. Otherwise it is just garbled. We are using the following setup: all of our phones are SIP phones; for our outgoing calls we make use of a

Greetings list, Have there been any further developments recently regarding presenting the original caller's caller ID to SIP devices after an attended transfer? I've googled around on the topic, but most of the threads I've found (some from this very list) are all dated back in mid-2006 and I wondered if there have been developments on the issue. To recap, the desired behaviour

I need attended transfers, but I do not have time to talk to another extension and see if they accept the transfer, my features.conf is: [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 220 ; Number of

I was given a custom CentOS 5.4 DVD, containing some Digium software for one of our customers. I need to turn this burned DVD into an image that can be written to a USB thumb drive. First, I ripped the DVD to an ISO image. That part works OK, my testing VM can load and run the custom kickstart script on the image. After totally frying my unetbootin install, I decided to try and use isohybrid

The Asterisk Development Team has announced the release of libpri 1.4.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Outgoing

Recently I asked a question about possibly unwanted calls due to extended transfer rights after attended transfers using DTMF sequences (http://lists.digium.com/pipermail/asterisk-users/2013-September/280536.html). Obviously, transferring with SIP INVITEs (hold + transfer keys) is not immediately affected by the this, but it is not always possible to enforce this. Meanwhile I have changed the

I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the addons. However, now it's up and running, only blind transfers work with "#", and I

Hi, we use Polycom SP IP 501 phones. We use the standard key/softkey configuration to do attended transfers. The only thing we miss is the CID info of the original caller after the call is transfered. This behaviour is different from the blind/direct transfer. With blind transfer method the original CID info is displayed. We already opened a call (in 2006) with Polycom JIRA. This is what they

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug.

Greetings list, I've been playing around this afternoon with Flash Operator Panel, trying to get it to do attended transfers. I am running the latest version. Has anyone managed to get this working reliably, and if so, would you mind sharing how you did it please? Alternatively, are there any other GUIs (free or commercial) that reliably support attended transfers? I'm trying to

Hi, I've got attended (superivised) transfer working with a handful of SIP phones, connected via different ATA's to an Asterisk CVS-D2005.05.28.22.00.00-07/12/05-20:47:08. pingu*CLI> show features Feature Default Current ------- ------- ------- Pickup *8 *8 Blind Transfer # ** Attended Transfer

Hi folks, I have a simple question regarding attended transfers. I have some queues where agents take calls and I have configured attended transfers between queues. That is, the agent dials the attended transfer extension that routes it to the aproppiate transfer queue where the second agent answers and they both talk for a while. Finally the transferrer leaves the call with *, connecting

Hi all, I'm new posting here, though not to perusing. I'm having an issue with attended transfer and was wondering if anyone had heard of the problem/had any suggestions... Apologies in advance if this post is excessively newb-oid. - An incoming call C is passed to A, a POTS telephone connected via a Handytone 286 ATA. - A presses atxfer key, then dials B, a Win XP laptop running

Hi, Asterisk: 1.8.4.2 I've just managed to configure attended transfers using Asterisk and Grandstream GXP-2000 phones. The only way I've got it to work is by using one of the out-of-band DTMF modes on the phone (either RFC or SIP-info). I think I can understand why - as Asterisk wouldn't be "seeing" the DTMF tones during the call if they are inband (or am I wrong)? I

We have contracted with an outside call center to provide sales for a certain product. We want to be able to transfer people over to those dedicated sales agents using an attended transfer (so we can prepare them with as much information as we have), to a regular extension. So far, so good. All of this is working just great. We want the caller's information presented as the CallerID so

The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes "hold", "transfer", dials the extension and announces the call. When the attendant pushes "transfer" the second time, the original call is lost. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that

Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real "simple" answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken

The Asterisk Development Team has announced the release of libpri 1.4.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.15 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix

The Asterisk Development Team has announced the release of libpri 1.4.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.15 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix