[asterisk-users] Calls Being Randomly Bridged
Michael J. Liberatore
mike240se at straightandnarrowinc.org
Sun Jan 20 20:22:57 CST 2008
-----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:
> Michael J. Liberatore schreef:
> On the snom 360
> If you pay close attention when you transfer the calls, you can see
> the names/numbers of the calling partners by using the "cursor" button
> (the round button with arrows) you can select to who you want to
> transfer to.
> It's an user issue, but you can't "blame" the user when there is a lot
> incoming traffic it takes too many button presses and careful
> attention to make a correct transfer.
>
> How to disable it?
> I don't know but i faced the problem that users occasionally want to
> bridge calls.
> e.g. someone calls for a person that only can be reached by Cellphone,
> this can be accomplished by asterisk and is often needed.
>
> Personally I'm still looking for a good solution for a central station
> that is easy to use and has a professional appeal, i thought the
> linksys
> 962+932 was it, but it has also some drawbacks.
> One(or two) button attended transfer is not reliable. certainly not
> when there are 2 or three simultaneously incoming calls. It gets
> confusing at that time.
>
> If anyone has any suggestions don't hesitate to make them!
>We noticed the same problem.
>.We tracked it down to this:
>snom gets a call and answers it.
>snom talks to the user. While talking to the user a second call comes
in (callwaiting is enabled) user wants to be >>>transferred so the snom
operator hits the transfer button.
>snom automagically selects the second incoming call as target and
bridges them.
>We called snom and they told us it's by design.
>We have not tested the new 7.1.30 firmware, but there have been a lot
of changes in the hold/transfer/fwd functions, so >>maybe they fixed it.
>We replaced the phones by aastra's on this particular location and
everything is fine now.
--
Michiel van Baak
michiel at vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
Thanks for the info, anyone else think this is CRAZY!!?? To assume that
you want to bridge the 2 calls when you press transfer is crazy. I am
on the phone with patient, another call comes in, I want to transfer
call to another receptionist so I can handle the new call, and when I
hit transfer it bridges the 2 incoming calls? Does anyone else see the
dumbness to this? 99% of the time you wouldn't want them bridged, so
having it as a default feature by design that cant be changedseems nuts.
Unless I am understanding what you are saying wrong.
I am def. gonna try the new 7.x firmware just released and hope it fixed
the problem.
It's a shame cause snom's could be great phones but the firmware has
always sucked.
The new polycoms look nice but they don't have the line buttons like
snom does, I need to have the blf buttons with lights for like 3 or 4
lines, and then the other extensions with blf enabled. The polycom's
don't have this, only on the screen which non tech users HATE.
Aastra I tried once and I think it had the blf buttons but not as many
as snom and I had trouble with the firmware, I don't remember which
model.
I have a couple linkssy sphones, they are nice but again missing the
blf/line buttons so do cisco's.
Does anyone like cisco with asterisk? I would assume if you get the sip
firmware that they are quite reliable, since lots of large corp's use
them. But they have similar issues with no blf/line buttons.
The granstream gxp-2000 has the blf/line buttons but they are terrible
phones.
Am I missing any phones? Any other suggestions?
How do you get around the no blf/line buttons on polycom and linksys?
No tech users hate it. Anyone use the new polycoms? They seem nice.
Now going back to the issue, I will never need to bridge 2 outside
calls, is there a way to disable it in asterisk some how? Never let 2
outside callers get bridged? Maybe in configs or code?
Thanks
Mike
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