[Asterisk-Users] Dial "T" option not obeyed with Grandstream BT101
andrew at envisionstudio.net
Mon Dec 1 14:33:43 MST 2003
It is appreciated if you submit your own code; otherwise I doubt
anything will be done. On the Grandstream phones I think the call would
be dropped if the transfer fails by disabling it in asterisk.
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Barton Hodges
> Sent: Monday, December 01, 2003 10:41 AM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Dial "T" option not obeyed with
> >> -----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
> >> 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
> >> 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
> >> 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
> > 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.
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