Avaya-style DISA

Asterisk Asterisk Support
1

Hi all !,
just joined the forums even if I’ve read them from some time now.

I’d like to do an avaya-like transfer. It’s an attended transfer done this way:

extension A calls (on line 1) extension B through asterisk
extension A calls (on line 2) a DISA number on the asterisk box. The DISA is filtered on A caller id. A inputs the DTMF digits amd is connected to C
When A hangs up B and C are connected.

I can do this by AGI but only for blind transfers. I check the active channels for callerid of A and bridge the other two channels at the end of the call.

Is there anything I can do to manage the attended call ?

A phone can’t send DTMF tones on existing call, only on new calls

TIA

giuliano

2

Does your endpoint not support attended transfer? Why do you have to involve DISA?

As a separate note, this capability works with analog phones attached to FXS ports on Digium telephony cards, provided that you’ve enabled threewaycalling in chan_dahdi.conf for the channel that’s doing the attended transfer.

3

Hi,
thanks for the info. unfortunately I have a complex setup and I must do something channel and phone independant. The only thing I know is that the phone will dial out using the DISA (for the first call) and use something else for the second call+transfer :frowning:

thanks

giuliano