[asterisk-users] Calls Being Randomly Bridged

Tue Jan 22 06:22:28 CST 2008

Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way.

Cheers,
Steve

On 1/22/08, Michael J. Liberatore <mike240se at straightandnarrowinc.org> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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" <mike240se at straightandnarrowinc.org>
> > > Subject: [asterisk-users] Calls Being Randomly Bridged
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >         <asterisk-users at 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
>
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