[Asterisk-Users] Bug in attended transfer or as expected?

[Moises Silva]

Its not so hard to look into the source code and make small changes.
Im not sure how hard it would be to implement what you want, but i
have tested it, and yes, you are right, the call get disconnected, and
i agree that souldnt be that way. You may want to open a feature
request in bugs.digium.com

Regards

On 1/25/06, Alex Barnes <abarnes at ubiquitysoftware.com> wrote:
> Sorry for bumping my own thread but just hoping that someone out there
> can help, I don't want to raise a bug on * if this is a strange issue
> with my dial plan
>
> Just to clarify this is attended transfer using asterisk and not a phone
> feature (not joining two held calls etc)
>
> Could someone with 1.2.x give the following call flow a try?
>
> - Connect a call between two phones on *
> - Called party initiates attended transfer, e.g *1 depending on how your
> features.conf is setup.
> - Dial internal extension
> - Ringing from transfer extension
> - Hangup original called party
>
> *I / most people would expect the caller to be connected to the ringing
> extension but instead on my * they get disconnected.
>
> In other words in this scenario users expect the attended transfer to
> switch to the same call flow of blind transfer.
>
> I would look into the code but am a Java / PHP dev :(
>
> Thank you for your help
>
> Alex
>
>
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