The thing is, when attended transfer happens, Asterisk connects
previously unrelated channels without going to the dialplan. I keep a
number of variables which help me track call state on both channels, and
their values become obsolete on attended transfer, as channels are now
in a completely new call. My idea was to gosub both sides to a dialplan
context so I can refresh call state variables. But that has proven
difficult if one of the calls is not answered. I am certain this can't
be solved by straight dialplan, not without modifying Asterisk code. I
haven't delved into ARI yet. I'd rather avoid it, unless it is the only
option.
On 24. 08. 2021. 20:51, Kevin Harwell wrote:
What's the overall scenario you are trying to solve? Perhaps there is
another way to do what you want to do without even modifying Asterisk
code? For example, maybe this is something an ARI application could
handle, or even straight dialplan using a combination of app_dial,
pre-dial handlers, and such.
On Mon, Aug 23, 2021 at 5:29 AM Nikša Baldun <i...@voxdiversa.hr
<mailto:i...@voxdiversa.hr>> wrote:
Hello,
I am trying to modify bridge.c (function
ast_bridge_transfer_attended)
in order to send channels involved in SIP attended transfer to the
dialplan. If both transferee and transfer target are bridged, that is
relatively easy. However, if transfer target is ringing, I don't know
how to find B-leg channels (there could be multiple, I suppose).
So the
question is, having a reference to A-leg channel, how to obtain a
list
of B-leg channels?
Best regards,
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