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[mike240se at straighta...]

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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[tilghman at mail.jeffa...]

On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote:

[tzafrir.cohen at xorco...]

On Sat, Jan 19, 2008 at 09:32:42PM -0500, Michael J. Liberatore wrote:

[fons.vanderbeek at 84-...]

Tilghman Lesher schreef:

[mike240se at straighta...]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Fons van
der Beek
Sent: Sunday, January 20, 2008 3:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Being Randomly Bridged

Tilghman Lesher schreef:

[fons.vanderbeek at 84-...]

Michael J. Liberatore schreef:

[michiel at vanbaak.info]

On 13:06, Sun 20 Jan 08, Fons van der Beek wrote:

[mike240se at straighta...]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel
van Baak
Sent: Sunday, January 20, 2008 7:27 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Calls Being Randomly Bridged

On 13:06, Sun 20 Jan 08, Fons van der Beek wrote:

[oza-4h07 at myamail.com]

2008/1/21, Michael J. Liberatore <mike240se at straightandnarrowinc.org>:

[Usman.Tahir at snom.de]

Hi Mike,

For starters disable "Call join on Xfer (2 calls):" on the phones. Since the setup has 6.2.x, it most likely doesn't have the setting "Allow incoming calls redirection through programmable keys" available on 7.1.30 for snom360. You might wanna try this version on a test system and see if it helps in that environment.

The problem, as discussed, seems to be originating when calls are parked on orbits that are mixing the two calls together. As long as you are debugging the issue, you should probably ask your friend to disable this practice and have a look at the call parking mechanism.

Regards,
Usman.

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snom technology AG
www.snom.com

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-----Original Message-----
Message: 11
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>
Message-ID:
<C091D65E9441814881CBCE751847902C81673D at sn-exch01.sn.local>
Content-Type: text/plain; charset="us-ascii"

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

[davies147 at gmail.com]

Hi,

My personal experience of this is that the call transfer facility on older
vesions of snoms (6.2.x is rather old now) is quite hard to get to grips
with - Particularly when managing multiple calls. Newer versions seem to be
better, but generally you need to train people to look at the screen and use
the silver keypad to choose the call to transfer to.

The worst situation is where 2 calls come in with no caller-id, so you have
no clue which call to transfer, and the phone does not store sufficient
state to automatically transfer the "last call I was on" to the "current
call I am on", or even make this the default transfer target, which is going
to be the requirement 99% of the time...

We use 6.5.12 firmware and upwards to 6.5.15. We have an open support ticket
on 7.1.30 causing calls to hangup when put on hold, so are not brave enough
to go there yet.

Regards,
Steve
On 1/21/08, Usman Tahir <Usman.Tahir at snom.de> wrote:

[davies147 at gmail.com]

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

[davies147 at gmail.com]

Oh, and the workaround is to disable call-waiting on the snom phone,
and use a queue to hold callers if the line is busy.

Regards,
Steve

[benny+usenet at amorse...]

"Steve Davies" <davies147 at gmail.com> writes:

[stephen.l.davies at gm...]

On 20/01/2008, Michael J. Liberatore <mike240se at straightandnarrowinc.org>
wrote: