>> -----Original Message----- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Behalf Of Barton Hodges >> Sent: Sunday, November 30, 2003 10:18 PM >> To: [EMAIL PROTECTED] >> 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:[EMAIL PROTECTED] >> >> 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 >> [EMAIL PROTECTED] 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
