I found this problem sufficiently interesting that I went and had a play with our snom phones in the test lab to try and determine what the behavious is. This is with 6.5.13 phones, and I think the results are somewhat inconsistent, particularly if snom are reporting this behaviour as "intended" as was suggested elsewhere in this thread...
We already disable the "Call join on Xfer (2 calls):" setting, so that can be taken into account in the descriptions below. 1) Simple unattended transfer. This does what is says on the tin regardless of how many other calls are ringing one the handset. It will transfer the call that is "in-hand" to the number dialled. Achieved with: Transfer, dial number, Tick 2) Simple attended transfer - One caller on the line. Again, this works fine Achieved with: Hold, dial number, tick, wait for answer, transfer, tick Or: Hold, dial number, tick, wait for answer, Hangup Or: Hold, dial number, tick, wait for answer, Transfer, Tick 3) With multiple inbound calls, the behaviour is less well defined. Here is what I found: Call 1 arrives, answer call. Call 2 arrives Call 3 arrives Press hold, dial destination for transfer of call 1, press Tick. Now there are 2 alternatives. a) Unattended. While the call is still ringing, press transfer, you will be offered a list of calls in the order 1, 3, 2 - This is 100% fine. The default destination is call 1 - The last call we dealt with. b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? 4) How to connect two external callers (as per original email). This is a stretch, but I can see it happening... Answer a call, put it on hold, wait for an answer. Re-select the original caller's line to let them know you are about to transfer their call. Press transfer (another call has come in in the meantime) the list you are offered defaults to the new (unanswered) call, and not the recently dialled and answered transferee. Not good really :( Basically, whatever calls the operator has had DIRECT involvement with should be kept at the top of the "stack" of calls, so that any default operations relate to those topmost calls. New calls go at the bottom of the stack, and stay there until there is some direct interraction with them. How hard is that? Just my 2p. Steve > > > > -----Original Message----- > > Date: Sat, 19 Jan 2008 21:32:42 -0500 > > From: "Michael J. Liberatore" <[EMAIL PROTECTED]> > > Subject: [asterisk-users] Calls Being Randomly Bridged > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > <[email protected]> > > > > Hi i have a friend who i setup an asterisk system for at his doctors > > office. it has 3 snom 360 phones with 6.2.x stable firmware and latest > > asterisk 1.4 and zaptel. They have the digium 4 port fxo card. > > > > They are extremely upset because calls are being randomly bridged for no > > rhyme or reason. They say that callers will call in and sometimes get > > connected with other callers, or they will be in the queue and then be > > talking to another caller waiting in the queue or on hold. Or they will > > be talking to a patient and then have another patient end up on the > > conversation. > > > > They are freaking out because of hippa and laws that govern privacy but > > i have no clue why. I assume most cases are conference calls being > > initiated by accident. > > > > So any help would be greaat. maybe just disabling conference calls > > would be a good start but i dont know how with sip phones. or maybe > > this is a bug? unfortuinately they dont give me much info and i dont > > use the phones so i dont have any specific logs to show, they just call > > me freaking out saying this stuff but they rarely can give me a specific > > call cause they get so many. > > > > thanks > > > > mike _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
