similar to: Attended call transfer

Hi all, I have had quite a few customer complaints about attended transfer cutting off callers. The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. I have checked the scripts I don't *think* this is a dial plan error but if anyone has this working correctly on Asterisk

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

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

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

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

I'm using asterisk blind/attended transfer feature on a queue (also tried with sip phones feature), and both type of transfers work fine. The problem is that attended trasfers doesn't get logged to queue_log, but blind transfers are logged just fine. Anyone knows if this is the correct behavior? -- Regards, Miguel Paolino -------------- next part -------------- An HTML attachment was

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. I know most people expect a good SIP/IAX phone to do the job but I think it's nice to be able to do attended trasnfers with a simple ATA-connected analog phone. I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works

Hi, I'm still hassling with the consultative/attended transfer stuff. Someone please help me identify this A lot has already been said about the ATA186. Some report it works fine, others say it doesn't. Lets get clarity on this. My scenario is reasonably simple (I think) Phone A: SIP/video1 Phone B: SIP/werkkamer Phone C: IAX2/provider Phone A calls phone B, they chat: *CLI> show

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:

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup()

Hello, Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has completed. Is this a known issue? Any new flags that need setting, etc? Thanks Jason -------------- next part -------------- An HTML attachment

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]

Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus

Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit "transfer" then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang

Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when

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