similar to: Attended transfer problem

Thanks a lot Dona and jg for your inputs. I'll try to find some way to do this from Dialplan or AMI and let you guys know soon. Please share if you have some more ideas. Regards, Rajib Date: Tue, 11 Jun 2013 18:34:46 +0200 From: jg <webaccounts at jgoettgens.de> Subject: Re: [asterisk-users] announcement to be played for attended transfer call To: Asterisk Users Mailing List -

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

Hi, During an attended transfer using the SIP phone feature buttons, I'm getting a few complaints from recipients that they can't tell when the call they are receiving has been transferred. Is there any way (even if it's complicated) to generate a beep tone to the recipient of the transferred call when the transfer is completed? I know you can do this with DTMF codes but they want to

I've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

Hi all! Ext A, B and C are SIP phones. Ext A receives a call from Ext B. Ext A wants to transfer the call to Ext C. Ext A puts the first call on hold, dials Ext C, then simply hangs up as soon as the call to Ext C starts *ringing*. In other words, Ext B wants to be sure Ext C is ringing (i.e. it is not busy or unavailable) but doesn't want to talk to him. Unfortunately, as soon as Ext A

Hello List, I want to play an announcement for attended transfer calls. For example, "A" calls "B", "B" answers the call and transfers (attended) to "C" - once transfer is complete "B" should hear an announcement saying "you call has been transferred". Is there any configuration in asterisk to implement this behavior? I have not

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 guys, I'm vainly trying to figure out how to setup quite a strange customers requirement: they require that when during attended transfer, (A->B->C) whenever B hangs up the call before it's connected to C, the call just returns to B, instead of changing to blind transfer. I tried using atxfer drop call option (enabling null channel in sources), but to no avail.. Is there

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

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com> wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no longer works. > > > > It only

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Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works. Cases that do work are as follows... Calls using both Queue() and

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

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 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

Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the

Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone

Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work

Hello, Sorry for a bit of a newbie post but we all had to start somewhere right .. I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.