[Asterisk-Users] attended call transfer
Alex Barnes
abarnes at ubiquitysoftware.com
Mon Feb 13 12:02:24 MST 2006
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling
> Sent: 13 February 2006 18:18
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] attended call transfer
>
> As I understand it, these issues only apply to DTMF based transfers.
> Why not use the transfer feature of your IP Phone or the Zap FXS port?
> _______________________________________________
Few reasons really:
1) Only one training method required for all users (we have multiple
types of phone within any single deployment what if a Snom user picks up
a call at a grandstream user's desk).
2) If we decide to start using a different phone for new deployments we
wouldn't need to reprint training materials.
3) Some phones don't support attended transfer at all.
4) Causes problems when you want to limit the number of calls any device
can make or receive. For example I have had to alter a script to allow
attended transfers to work but this means that internal -> internal
calls don't count towards the maximum limit.
5) Better CDR maybe, rather than having two calls and then joining them
together Asterisk itself knows it's a transfer so can log the action
much better.
6) The Snom 360 phones we use as reception phones do not behave well
either going from attended to blind (I think) although they don't hang
up the caller but rather than bring this up with Snom I believe this is
a PBX issue.
7) Asterisk shouldn't claim to have attended transfer capability if it
has fundamental flaws.
I cant think of anything else right now but am sure there's more :-)
But you are entirely correct, there is no option at the moment but to
use the IP phones attended transfer.
HTH
Alex
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