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

[Barton Hodges]

>> -----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
>> 
asterisk-users-admin at lists.digium.com wrote:
> 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.

That confirms my suspicions.  What is the correct avenue for reporting
this, and a few other problems as bugs?  I am also interested in
submitting some patches.

Barton