[Asterisk-Users] Dial "T" option not obeyed with Grandstream BT101

[Barton Hodges]

In the following scenario, the user calling from a SIPphone registered
phone is able to transfer the called user to another extension.

sip.conf:
[general]
port = 5060
context = from-sip
register => number:password at proxy01.sipphone.com

extensions.conf:
[from-sip]
exten => s,1,Dial(SIP/111&SIP/117)
exten => 111,1,Dial(SIP/111,20)
exten => 117,1,Dial(SIP/117,20)

1. The calling user dials "number", which drops them into [from-sip]
2. Extensions 111 and 117 are Dialed.
3. The called user picks up extension 111.
4. The calling user presses "Transfer" on the Grandstream phone, then
dials 117 and presses "Send".
5. The called user on extension 111 is then transferred to extension
117.

I don't believe this is supposed to happen because I have not
specified the "T" option to the Dial command.  Even without any
options specified at all, both the calling and called users are able
to transfer the call.

I'm using a CVS snapshot from Sun, Nov 30th 04:04:45, 2003.

What am I missing here?

Barton