Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way.
Cheers, Steve On 1/22/08, Michael J. Liberatore <[EMAIL PROTECTED]> wrote: > Wow thanks so much for this, this is a lot of great info. Hopefully > enough to catch snom's attention to. Is it possible for you to try 7.x > on one of the phones and see if it corrects the problem? > > What it comes down to, is that the phone is too complicated to handle > multiple calls for non technical users. They have to keep track of way > too much, even a techie like us could get mixed up sometimes, especially > in a high stress doctors office where there are half of the number of > receptionists that are reeally needed. > > Mike > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steve > Davies > Sent: Monday, January 21, 2008 9:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Calls Being Randomly Bridged > > 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 > > > <asterisk-users@lists.digium.com> > > > > > > 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 > > > This E-mail, including any attachments, may be intended solely for > the personal and confidential use of the sender and recipient(s) named > above. 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