[asterisk-users] Asterisk Transfer/call patching support

[Dan Cropp]

I'm coming to Asterisk from a traditional PSTN environment, so forgive
me if I use the wrong Asterisk/SIP terminology.

 

I need to make a product where calls come in go through various menus
and based on various configurations perform attended transfers, blind
transfers, and patch callers together.

 

For patching two calls together, my thought is that this would be a
conference in Asterisk.  Is this correct?

 

For attended transfers, is there a way to perform this from a dial plan?
Or would I need to use AMI?

 

Also, with Asterisk transfers (SIP and PRI calls), will the transferred
call disappear from Asterisk?  For example, with PRI QSIG transfers, if
the external switch allows it, both parties of a PSTN call are removed
from a switch and instead the parent switch becomes responsible for the
bridged calls.

 

I'm using the current Asterisk trunk with plans to use Asterisk 1.8 once
it's released.

 

Have a great day!

Dan

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