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

[Alex Barnes]

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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