similar to: Attended transfers and orginal caller ID

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

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

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, I have been trying to enable attended transfer for callee. When the callee pressed *2, DTMF tone was heard by the caller. But when the caller pressed *2, attended transfer started. It's strange. I used two SIP phones. My Asterisk version is "Asterisk CVS-HEAD built by root@router on a i686 running Linux on 2005-06-27 06:07:18". In features.conf, I have: [featuremap]

Hello list, I'm sending calls to a queue in the attended way, that is, *1.* the original call is put on hold, *2.* a second line is open to call the queue, *3.*after an agent is connected the original call is transfered to its final destination. 1. Zap/1-1 <--> SIP/agentA-tag1 2. SIP/agentA-tag2 <--> SIP/agentB-tag 3.

Hi, using asterisk@home, with quadBri from junghanns.net I am facing a strange problem: I have set incoming routes for some extension / DID: [ext-did] include => ext-did-custom exten => 23,1,SetVar(FROM_DID=23) exten => 23,2,Goto(ext-local,23,1) exten => 57,1,SetVar(FROM_DID=57) exten => 57,2,Goto(ext-local,57,1) exten => 66,1,SetVar(FROM_DID=66) exten =>

Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has

Hi I have an ISDN connection with 100 DIDs assigned to it... What I'm trying to achieve is set the proper outgoing callerID while showing the local caller's extension in the CDR. There is a behaviour that I just can't explain. the callerid field in sip.conf is set as : callerid="Jean-Yves/E" <300> the callerid in iax.conf is set a: callerid="Jean-Yves/E"

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

We have two Asterisk servers running CVS-HEAD (06/02/05 and 06/28/05). Most of our calls are either incoming or outgoing to external (PSTN or non-Asterisk) numbers, and only our internal users can initiate the transfer. Only half of the attended transfers work. It goes like this: 1)Extension 8123 calls number 19876543210 2)During the call, extension 8123 dials *2 to do an attended (non-blind)

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

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

--- Matt Zimmerman <mdz@debian.org> wrote: > retitle 119396 flac: Decoded raw data does not match original raw > data > thanks > > On Tue, Nov 13, 2001 at 02:38:32AM +0000, Adam M. Costello wrote: > > > Matt Zimmerman <mdz@debian.org> wrote: > > > > > Based on a few quick tests, this kind of problem seems to occur > with > > > any

I have an urgent need to install the original Icecast (1.3.12). I have a piece of broadcasting software that can only receive a stream from the "old" version of Icecast. I've tried building it under Debian 9, but failed. Can anyone provide any assistance? I'd prefer to stick with Debian, but I'm willing to use a different distro if necessary. I've worked in IT for

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

retitle 119396 flac: Decoded raw data does not match original raw data thanks On Tue, Nov 13, 2001 at 02:38:32AM +0000, Adam M. Costello wrote: > Matt Zimmerman <mdz@debian.org> wrote: > > > Based on a few quick tests, this kind of problem seems to occur with > > any file smaller than 4k (at least when using 44100Hz/16bps/2chan). > > I don't think the problem

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