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[asterisk-users] Calls Being Randomly Bridged

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mike240se at straighta...
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PostPosted: Mon Jan 21, 2008 8:28 pm    Post subject: [asterisk-users] Calls Being Randomly Bridged Reply with quote

Yes these 2 options have been set to NO all along. I double checked
too.



________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stephen
Davies
Sent: Monday, January 21, 2008 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Being Randomly Bridged


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

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.



In the SNOM settings there are two options that you should set to "No".

That is "Call Join on Hangup" and "Xfer on Hangup". (Or names similar
to that).

Steve





This E-mail, including any attachments, may be intended solely for
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mike240se at straighta...
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PostPosted: Mon Jan 21, 2008 8:29 pm    Post subject: [asterisk-users] Calls Being Randomly Bridged Reply with quote

I do have queues set up but I would have to setup queues for all calls
then, even from other inside the office calls. Cause if I disable call
waiting, wouldn't that be the same as saying maximum sip connections to
the phone = 1?

Or is call waiting different on the snom phones?

-----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 9:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Being Randomly Bridged

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

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and not a public document. Pursuant to 42 CFR, any information in this
e-mail identifying a former, present, or potential client of Straight & Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail.
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mike240se at straighta...
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PostPosted: Mon Jan 21, 2008 8:31 pm    Post subject: [asterisk-users] Calls Being Randomly Bridged Reply with quote

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 Sad

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
Quote:
Quote:

-----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

Quote:
Quote:
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.
Quote:
Quote:

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

Quote:
Quote:
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

Quote:
Quote:
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. This message may include advisory, consultative and/or
deliberative material and, as such, would be privileged and confidential
and not a public document. Pursuant to 42 CFR, any information in this
e-mail identifying a former, present, or potential client of Straight & Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail.
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benny+usenet at amorse...
Guest





PostPosted: Tue Jan 22, 2008 6:22 am    Post subject: [asterisk-users] Calls Being Randomly Bridged Reply with quote

"Michael J. Liberatore" <mike240se at straightandnarrowinc.org> writes:

Quote:
I do have queues set up but I would have to setup queues for all calls
then, even from other inside the office calls. Cause if I disable call
waiting, wouldn't that be the same as saying maximum sip connections to
the phone = 1?

Call waiting off means that someone who calls the phone while at least
one call is ongoing will get the busy tone (or e.g. voice mail, if
that's what your dial plan says should happen).

You can just send the office-calls into the same queue as the calls
from outside. You can also give them their own queue, or give them
priority.
/Benny
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davies147 at gmail.com
Guest





PostPosted: Tue Jan 22, 2008 7:03 am    Post subject: [asterisk-users] Calls Being Randomly Bridged Reply with quote

On 1/22/08, Michael J. Liberatore <mike240se at straightandnarrowinc.org> wrote:
Quote:
I do have queues set up but I would have to setup queues for all calls
then, even from other inside the office calls. Cause if I disable call
waiting, wouldn't that be the same as saying maximum sip connections to
the phone = 1?

Or is call waiting different on the snom phones?


Call-Waiting can be disabled on the handset. With snom phones, this
can be set to 3 settings via handset, web interface or provisioning:

1) Enabled with beep (Never use this - it is horrible)
2) Enabled, visual only (Use this if you want CWI)
3) Disabled.

The phone will allow >1 outbound call, but will send a "busy" response
if a call is already in progress.

Cheers,
Steve
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davies147 at gmail.com
Guest





PostPosted: Tue Jan 22, 2008 7:22 am    Post subject: [asterisk-users] Calls Being Randomly Bridged Reply with quote

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:
Quote:
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 Sad

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


Quote:
Quote:

-----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

Quote:
Quote:
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.
Quote:
Quote:

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

Quote:
Quote:
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

Quote:
Quote:
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. This message may include advisory, consultative and/or
deliberative material and, as such, would be privileged and confidential
and not a public document. Pursuant to 42 CFR, any information in this
e-mail identifying a former, present, or potential client of Straight & Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail.


_______________________________________________
-- 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
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