[Asterisk-bugs] [Asterisk 0009939]: Transfer implementation

noreply at bugs.digium.com noreply at bugs.digium.com
Sun Jul 8 05:09:13 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=9939 
====================================================================== 
Reported By:                wedhorn
Assigned To:                qwell
====================================================================== 
Project:                    Asterisk
Issue ID:                   9939
Category:                   Channels/chan_skinny
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 67843 
Disclaimer on File?:        Yes 
Request Review:              
====================================================================== 
Date Submitted:             06-11-2007 04:33 CDT
Last Modified:              07-08-2007 05:09 CDT
====================================================================== 
Summary:                    Transfer implementation
Description: 
Initial patch adding transfer to chan_skinny. Works, but lots of bugs and
more work to be done. Basic functionality is <XFER> to start transfer, puts
call on hold and gives you a dialtone. Dial number and after connected
press <XFER> again and call is transferred.

After first <XFER> press and before call ringing/connected, you can press
<XFER> again to use a blind transfer. In this case transfer occurs when
either channel being dialed indicates ringing or answering. When in blind
transfer you can press <XFER> again to go back to attended transfer.

<HOLD> toggles between the transferee and transferor.
====================================================================== 

---------------------------------------------------------------------- 
 mvanbaak - 07-08-07 05:09  
---------------------------------------------------------------------- 
With this patch applied I get some weird stuff:
When a call ends, it is not removed from the phone's display.
Example: I checked my voicemail and hit # at the end to make asterisk
hangup the call. in asterisk CLI the call is indeed terminated but my 7960
still has the call in the display, with the calltime counter running.
A minute later someone calls me so I see a second call on my 7960 but I
cant pickup that call. The remote party hangsup and my 7960 now still has
the 'active' call to the voicemail and the incoming call on the display.

Had to restart asterisk so the phones reboot and everything is working
again for the first call made/received. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-08-07 05:09  mvanbaak       Note Added: 0066744                          
======================================================================




More information about the Asterisk-bugs mailing list