[asterisk-dev] Attended transfer: transferring a call as soon as the destination starts ringing

[Leif Madsen]

Alex wrote:
> Hi all!
> 
> Ext A, B and C are SIP phones.
> 
> Ext A receives a call from Ext B. Ext A wants to transfer the call to
> Ext C.  Ext A puts the first call on hold, dials Ext C, then simply
> hangs up as soon as the call to Ext C starts *ringing*. In other
> words, Ext A wants to be sure Ext C is ringing (i.e. it is not busy or
> unavailable) but doesn't want to talk to him.
> 
> Unfortunately, as soon as Ext A hears Ext C is ringing and hangs up or
> hits "Transfer", the call is closed and a *new* call from Ext B to Ext
> C starts. This way, Ext C sees an unanswered call from Ext A, which is
> an unexpected behaviour.
> 
> I played with directmedia and directrtpsetup, but no success so far.
> Any ideas, please?

Which version of Asterisk are you using? This sounds like a recent issue that 
may have already been resolved.

I'd try the latest checkout from the branch you're currently using to determine 
if the issue has already been resolved. There was a commit for transfers just 
today from what I've heard as well.

For example, if you're using 1.6.2.2, then you're using the 1.6.2 branch, so 
checkout the latest version onto a development system:

svn co http://svn.asterisk.org/svn/asterisk/branches/1.6.2

Then compile, install, and test as normal.

If you're still have issues, please check the issue tracker at 
https://issues.asterisk.org and check to see if an existing issue is already 
reported describing your problem. If not, then you may file a new issue.

Please describe how to reproduce the issue, along with console output with debug 
level logging enabled, and if SIP channels are involved, please add a SIP trace 
from the Asterisk console. Attach the files as text files to the issue and don't 
paste large blocks of text directly into the issue. (Files can be attached after 
you submit the issue -- you won't see an Attach button before filing it.)

Thanks!
Leif.