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

[Andrew Joakimsen]

The T option is for the # transfer which is handled by Asterisk, in your
case the phone has a transfer button and is able to send SIP messages
telling Asterisk that the call should be transferred.

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Barton Hodges
> Sent: Sunday, November 30, 2003 10:18 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Dial "T" option not obeyed with Grandstream
> BT101
> 
> 
> 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
> 
> 
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