On Mar 2, 2006, at 2:15 AM, Vahan Yerkanian wrote:
Reboot once again and it picks up the new config. Two-step
provisioning takes a couple of reboots to insure the device has
reconfigured itself. Applies to 2100, 3000, 841 and 941 models.
I've had good results on our 942 by setting the resyn
(now that I've remembered which address is subscribed to this list)
Does anyone with one of these phones have any sort of presence
working? I'm looking to monitor the DND state of the phones, if
nothing else. Setting the SIP-B bit enables SUBSCRIBE/NOTIFY, but
the "dialog" package is the
On Jun 6, 2005, at 4:39 PM, [EMAIL PROTECTED] wrote:
I want to manage this dialplan variable for each extension separately,
unfortunately this doesn't work:
**77,hint,DS/splat${CALLERIDNUM}
Do you have an idea for that?
Is there an easy place to patch it in asterisk 1.0.7 stable?
Will it be p
On Jun 4, 2005, at 4:52 AM, [EMAIL PROTECTED] wrote:
I would like the SNOM extension light to permanently
reflect the current toggle status of my application logic/asterisk DB
variable.
There's a phantom device in bristuff that can be used for this sort
of thing. When you toggle the dialpla
On Apr 17, 2005, at 12:23 AM, Lance Grover wrote:
I have rebooted the phone and restarted asterisk after each change.
Did you do it in that order? If so, that is probably a source of
trouble (you should restart or reload asterisk before the phone boots,
not after).
--
Joshua P. Dady
http://www.
(boy mail in this list piles up fast when I can't check it)
On Apr 8, 2005, at 10:03 AM, Michael George wrote:
- It appears that the extension used with the "hint" must be the same
as the
extension used to dial that channel. So if extension 22 will ring
Zap/2,
then "exten => 22,hint,Zap/2" w
On Apr 3, 2005, at 3:37 PM, Philipp von Klitzing wrote:
- quality of the handset and the speaker phone?
Our primary issue with the handset on the 220 was that the hook was
really easy to miss; the 360 is easy to hang up without making your
caller think you dropped the phone several times trying t
Okay, after talking with Sven today, it turns out my problem
description is wrong (I was combining to cases, one of which does work
in the current firmware):
- Multiple incoming calls (works already)
- Incoming call while dialing (or waiting for answer of) outgoing
call (doesn't)
--
Joshua
On Apr 1, 2005, at 2:34 PM, Noah Miller wrote:
Just so you know, I don't think anybody has gotten the Polycom
presence features to work properly with Asterisk. Am I wrong,
anybody?
If we went that way, I was going to handle the presence directly (i.e.,
a separate python script that subscribes t
On Apr 1, 2005, at 1:15 PM, Noah Miller wrote:
The vast majority of our handsets are Polycoms. I know that they do
this correctly (with a little help from CheckGroup/SetGroup and
multiple SIP registrations). Of course, you can't get the nifty
sidecar for the Polycoms like you can for the 220.
On Apr 1, 2005, at 12:20 PM, Noah Miller wrote:
This is just the way that Snom phones work. They say they want this
behavior to keep things simple. Here is an old message from this list
(Christian works for Snom):
http://lists.digium.com/pipermail/asterisk-users/2004-April/044273.html
- Noah
P
On Apr 1, 2005, at 10:52 AM, Noah Miller wrote:
I don't have a 220, and I haven't really tested the 360, but on our
190's I just register each line appearance to the same sip device, and
multiple simultaneous calls automatically roll from line 1 to line 2
to line 3, etc. Are you using any Check
I've got an issue on the snoms, and I'm wondering if anyone has some
recent experience with it; I've contacted the one specific reference I
found to it in the list archives, and the person in question didn't
seem to find an answer (and snom doesn't appear to be finished moving
their offices yet
I'm working through my list of features people will expect, and Hold
Pickup is at the top at the moment -- has anyone done any work on this?
We've had some unpleasant experiences with call parking, and everyone
seems to like the Hold Pickup model. If you don't know what I mean by
Hold Pickup,
On Mar 18, 2005, at 8:27 AM, Ben Ruset wrote:
3. They don't realy support their phones, unless there is a hardware
problem.
They don't support them with Asterisk, but if you don't tell them
about it, they tend to be very good at working to resolve issues.
You have to know what issues they conside
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended to the To: header as per requirements.
The "intercom=true" needs to be appended to the request URI, not to the
header as a whole -- your To: header should be:
To:
Mind you, I didn't get the phon
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended to the To: header as per requirements.
The "intercom=true" needs to be appended to the request URI, not to the
header as a whole -- your To: header should be:
To:
Mind you, I didn't get the phon
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