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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users