similar to: Attended Transfer using AMI

Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: > Doe anybody know of a way to redirect both channels from a bridge to > different dial plan extensions from the using the AMI. > > Currently, as soon as I redirect one of the channels the other appears > to be dropped and gets reorder tone (congestion, fast busy). > > I guess what I really need is a

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hello List, Please help with the following problem, I have a situation, where I need to play an audio announcement to the caller SIP channel once an attended transfer is successful. The attended transfer is done from client. I can see a transfer event in AMI. I am not using 'T/t' option in dial() command. The transfer is completely on client side using SIP signaling. 1. A calls B 2. B

Hi, How can you add specific statements into Asterisk dialplan (extension.ael, ...) for attented transfers ? I can see Asterisk sending Transfer or Masquerade events through AMI (in 1.6.1) but I could use an external program to catch those events but I would prefer to use dialplan instead. Any idea ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 -

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

Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. Thanks, Neil Cherry

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel:

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, 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

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

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

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.

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]

Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 channels below are somehow associated, but how can I be 100% sure they are related in order to implement a redirect of the incoming call to another phone ("attended" call pickup respecting call/pickupgroups). Uniqueid seems to be a