> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Steve Davies > Sent: Monday, June 13, 2005 6:17 AM > To: [email protected] > Subject: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?) > > Hi, > > I am using a number of snom190 phones, and an asterisk "gateway" > server, and recently started experimenting with call transfers. The > snom phones provide support for attended and un-attended call > transfer, so I would rather use that than call-parking. > > I have found that un-attended transfer works fine, and that attended > transfer works fine if the originating phone call is NON-SIP (ie. > ISDN) > > I hope that some of this makes sense... > > When I look at the SIP trace for the sequence of A calls B and is > transferred to C, I see: > A makes call to B: > A calls B > B picks up > A and B are bridged (re-INVITEd) and talk directly. > B then puts A on hold: > (A and B are both INVITE to talk via Asterisk) > B makes a call to C, I see: > B calls C > C picks up > B and C are bridged (re-INVITEd) and talk directly. > B presses transfer: > (Same as putting B and C on hold, B and C are re-INVITEd to talk via > Asterisk) > B selects which line to transfer to C > B REFERs A to C by asking Asterisk. Asterisk accepts this. > B is notified that A is disconnected > B gets "BYE" for call to A > B gets "BYE" for call to C > C gets INVITE to talk to B via Asterisk <<<<<<<< Why????? Why not to 'A' > B requests that call to A is closed down. > > The upshot of all this is that B is correctly out of the loop, and > that Both A and C think they have opened communications with a new > phone, both via Asterisk. Unfortunately there is no Audio. If one of > the parties hangs up, the connection is correctly closed. > > I am curious why Asterisk would put a "From:" of "B" in the final > INVITE to bridge the calls. Perhaps this is just how SIP spoofs the > communication so that C does not need to know about the 2 callers? > > Is there some way I can track down where my audio is going? As > mentioned, this problem only seems to occur if A,B,C are all SIP > phones, but not if A is an ISDN call. > > Thanks, > Steve > _______________________________________________
Upgrade your snom firmware to the latest and make sure break key = off in the setup. Use the transfer feature in asterisk for attended transfers. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
