similar to: Cancel attended transfer

Displaying 20 results from an estimated 2000 matches similar to: "Cancel attended transfer"

2010 Jan 26
2
Attended Transfer with REFER
Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a
2015 Jul 15
2
how to return a transfered call to the transferrer?
Hi all Any of you guys could point me in the right direction? I need to make that a blind transfer to return to the transferrer when the transferee does not answer. Scenario: . Miss Jane Doe, our front desk attendant, picks up an external call to Mr. Smith; . Miss Doe flashes, dial Mr. Smith's extension and then hangup; . Mr Smith's phone rings until timeout; . At this point, how
2009 Sep 16
3
[asterisk-dev] MeetMe in Macro
Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub
2010 Sep 07
1
Solving the CDR mess of attended transfers
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body bgcolor="#ffffff" text="#000000"> <font face="Arial">Is there a way to solve the mess on CDR caused by CDR
2017 May 29
2
Best way to know a call is being transfered
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:
2010 Sep 03
3
How to tell if there is a transfer from CDR?
Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:
2009 May 18
2
From 1.4 to 1.6.0
Hi everyone, I was just wondering, does anyone managing production asterisk servers migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if there are some successful cases. Is your 1.6.0.X behaving well, with acceptable stability? Please share your experiences. Thanks, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57
2010 Sep 03
4
openvz
Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi
2003 Sep 18
1
CDR of calls transferred via IAX[2]
Let's say i have a network of * boxes connected via IAX, one of them is a "switch", one or more are the "gateways". - An IAX[2] "customer" register himself on the "switch" (and gets an accountcode for te purpose of cdr) - The customer places a call to the "switch", the switch does some magic and decides which "gateway" the call
2010 Sep 09
2
How to avoid interruptions with DIGIUM
Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the
2009 Mar 26
3
Know who's logged in
Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #
2008 Jul 09
2
transfers only work when voicemail enabled
Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart
2009 Mar 19
2
Script to softly restart Asterisk each midnight to clean locked channels
As Asterisk has inner problems and channels very often locks we have such script to restart Asterisk each midnight. We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions because there are too much changes which would brake our system (realtime/sip/iax2/cdr/etc/etc). Script soft hangups all alive channels in dirty way then kills Asterisk and starts it up. Hope
2018 Aug 08
2
Queue breaks Dynamic_Features on Attended Transfer
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
2009 Jun 15
1
Opinion on Attended transfer in features.conf
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
2006 Nov 01
3
Remote-Party-Id and Attended Transfers
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.
2004 Aug 19
3
GrandStream BT101 Attended Transfers
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
2005 Jul 21
1
attended transfert
hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks
2008 Feb 27
3
Attended transfers through a GUI
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
2006 Mar 14
3
Attended Transfer - transfer timeout, how to change?
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. --