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

[Alex]

2010/3/1 Leif Madsen <leif.madsen at asteriskdocs.org>:
> 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 am using 1.6 from trunk. I've checked out a couple of days ago.

> 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.

I'll give it a try, thanks for this information. However, IMHO the
issue you probably refers to (bug #16816 -
https://issues.asterisk.org/view.php?id=16816 ) does not affect my
scenario: in the case I reported, the call is transferred while the
3rd extension is still ringing, not after it has been already
answered.

Alex