Summary:	ASTERISK-14862: Issue in Blind Transfer
Reporter:	Adan Shoeb (adnanshoeb)	Labels:
Date Opened:	2009-09-22 10:28:37	Date Closed:	2011-06-07 14:07:21
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Addons/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Here dialplan:- [transfer-context] exten => _X.,1, Dial(SIP/${EXTEN}|60000|60000) exten => _X.,2,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() Issue:- Can able to transfer call. Originator can not hear any voice. If hangup mediator then the originator also hangup. If hangup originator then mediator and the receiver still connected with proper voice. If hangup receiver then again voice ok between mediator and originator.
Comments:	By: David Woolley (davidw) 2009-09-22 11:15:01 If you have a connection between the mediator and the receiver, you have some form of attended transfer, not a blind transfer. Can you clarify "originator cannot hear any voice"; this is normal for the enquiry phase of an attended transfer, as long as the mediator and receiver can hear each other.  Do you have one way speech? You definitely need to provide the following information: 1) whether this is a SIP transfer or an Asterisk features transfer (if you don't understand, then describe exactly what you did); 2) all the information for a SIP problem as listed in the bug reporting guidelines, assuming that the caller and the mediator were connected using SIP. By: Adan Shoeb (adnanshoeb) 2009-09-22 11:46:39 i have installed asterisk 1.4.19 on my box, I have a setup 9 for blind transfer to call any inbound numbers in feature.conf. But when an incoming call to my sip user and they are connected then I press "9" to transfer call on local SIP extension , transfer prompt played but the issue same as I above described. But i am not getting why the incoming call is not transfer to any other local extension? By: Isabel Arias (icristy) 2009-09-22 16:42:39 I have the same problem but with the 1.4.23 version of asterisk, is there any solution? By: David Woolley (davidw) 2009-09-23 05:52:36 icristy:  Please provide the standard SIP debugging information (debug and history output). By: Leif Madsen (lmadsen) 2009-09-23 09:18:08 I'm assigning this to myself to reproduce, however, I *need* the information that was requested by davidw. I need to know HOW you're doing this transfer so I can reproduce easily. I also need to know which devices/softphones you're using while performing this transfer. Be very specific, otherwise I'll just have to close this issue and request that you use the asterisk-users mailing list to gather the required information. Thanks! By: Leif Madsen (lmadsen) 2009-10-08 11:25:17 Closed due to lack of feedback.