Attended transfers manager or phone

2 msg	bad sound quality after Redirect
4 msg	volume problem
2 msg	SVN Server Issue?
1 msg	help Unable to dial _99XXXXXXXX
2 msg	Channel fallback
2 msg	chan_mobile type=
6 msg	WARNING[31046]: chan_sip.c:4978 process_sdp:Una...

Attended transfers manager or phone
\ Christian Ejlertsen (15 Jan 2008)
. \ Mojo with Horan & Company, LLC (16 Jan 2008)
. . \ Christian Ejlertsen (16 Jan 2008)

2 msg	inbound Audio problems probably not NAT related?
2 msg	asterisk 1.4 context
2 msg	Record calls then send them to users voicemail
2 msg	Heartbeat
5 msg	Interrupt the swift text
15 msg	Discover Asterisk 1.4 :: SIP Subscriptions
1 msg	sip channel error - extension pattern matchingp...
1 msg	busy/congestion random
6 msg	cisco ip phne 7911G with asterisk
2 msg	Playing DTMF tones down a channel
7 msg	SIP Reason
2 msg	Fax machine detect

Subject:	Attended transfers manager or phone
Group:	Asterisk-users
From:	Christian Ejlertsen
Date:	15 Jan 2008

Well I'm sure this issue has been bean up a few time since it's one of the
only ones I can't find a real "simple" answer to.

I'm trying to find away to do attended transfers through the manager
interface, for a pc switchboard / Agent client solution, but so far coming
up short.
The action Originate is part of the solution, but what really I want is the
phone being taken off-hook and then being able to dial the number without
having to answer the dial-back first.

1. One solution, though an ugly one, would be using Originate, but use a
phone that has some sort tcp/ip interface that allows for taking the phone
off-hook.

2. A Better solution would be using a phone that allows dialling and taking
the phone off-hook on-hook etc. via some tcp/ip interface.

3. Yet another solution, though I do not favour this one since I really
don't want to maintain the sip phone code, would be programming a soft sip
phone with all the bells and whistles and adding the switchboard
functionality to that (name searching, status email so on and so forth.

In the end all I need is just a software or hardware phone, sip/iax, which
can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps
status requests. If such a phone exists that would do the trick, the rest is
manageable via the Asterisk Manager console.

I'm guessing some people have messed with this problem before so I hope that
someone has some information about this kind of thing :)

Thank you in advance
Christian

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