Displaying 20 results from an estimated 1000 matches similar to: "Polycom IP601 Question"
2007 Nov 08
0
Polycom IP601 (mac)-directory.xml changes don't update phone
Hi Polycom experts,
I'm having a problem getting changes to the Polycom IP 601's
(mac)-directory.xml file to update the button list on the phone. If
the phone is newly provisioned (i.e. if I "Format File System" on the
phone) then the new list will show up on the buttons, but of course
this is pretty drastic way to do it.
- Environment: Asterisk test setup with 7 phones,
2006 Mar 24
3
Polycom 601 Message Center
While I know this is not a true asterisk problem, I figure someone where
may know. When you click on Messages and it gives you the count of
Urgent, New, etc. How can you make the phone gather that information?
For example, my phone shows me there is an e-mail. It also sends an
e-mail. Yet, when I click on message before I connect to the contact
center, it doesn't have any counts.
Here is
2019 Jun 25
2
Emails not visible after renaming folders
Hello,
I have strange problem with "losing" emails after rename mail
folder(s) (via imap client: thunderbird, roundcude, etc..)
How to reproduce:
1. Create some folder name, like TEST
2. Create sub-folder under TEST (like SUBTEST)
Structure:
TEST
|--SUBTEST
# doveadm mailbox list -u postmaster at testmailbox
Spam
Trash
Sent
Drafts
INBOX
TEST
TEST/SUBTEST
3. Move (or copy)
2019 Jun 26
2
Emails not visible after renaming folders
Copying or moving with email client: thunderbird, roundcube (webmail), mutt or any other email client via imap protocol.
25.06.2019 22:10, Germ?n Herrera ?????:
> Are you copying/moving the emails with {cp|mv} or with "doveadm {copy|move}"?
>
> On 2019-06-25 12:00, Aleksandr via dovecot wrote:
>> Hello,
>>
>> I have strange problem with "losing"
2006 Oct 27
1
Taking a Polycom IP601 home
Make sure you set nat=yes for the sip user. Asterisk will then send replies back to the source IP address, rather than what's in the Via: header.
> -----Original Message-----
> From: Warren (mailing lists) [mailto:warren-lists@icruise.com]
> Sent: Friday, October 27, 2006 5:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Taking a
2008 Mar 11
1
Newbie Polycom: IP601 console with expansion module
I was reading a Polycom brochure and it appears that there is really no
special receptionist console and the console is basically a IP601. Is
this correct?
The only difference is to purchase an expansion module in order to have
more shortcut keys for the girls.
So, apart from the hardware, as far as the dialplan is concerned, do I
just treat the receptionist console as any other extension?
Are
2008 Apr 04
0
discrepancy between CDR clid and Polycom IP601 clid
Hi,
Returning to my office I find two "missed calls" (from autodialers) that
my IP601 displays as originating from 01111111111. However the CDR
database recorded the call this way:
calldate: 2008-04-04 14:18:16+02
clid: 0172752780
src: 0172752780
dst: 2131
dcontext: default
channel: Zap/1-1
dstchannel: SIP/0146472131-007a7e80
lastapp:
2007 Nov 08
0
Polycom IP601 call parking
One more Polycom IP601 question please (sorry for the long intro here
to document) ...
In order to closely approximate the behavior of the previous telephone
system that many of the users are familiar with, I have set up call
parking like this:
- features.conf [general] section contains:
parkext => ** ; What extension to dial to park
parkpos => 10-11 ; What
2007 May 09
1
Boost Polycom IP601 headset volume
Hi everyone, I have a user that needs a little extra volume on his
Polycom IP 601 phone set for all calls (beyond what the volume control
currently offers). Is there a provisioning setting for this anywhere?
(I'd like to avoid a separate amplifier between the phone and handset if
possible.)
Alternatively, is there a way to have Asterisk 1.4.x boost the volume to
a particular SIP device
2007 Jan 16
2
Polycom IP601 - some hints working, not others?
Are all of the sip phones in the same context?
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Robert Jenkins
> Sent: Tuesday, January 16, 2007 1:44 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Polycom IP601 - some hints working,
2006 Nov 21
4
IP601 Expansion Module HELP!!!
Hey list,
Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion
modules to work. I need the XML config files I guess. Does anyone have these
I can have? Im trying to get this phone up and running, and haveing MUCHO
problems, can someone help me out!! Im sure if I see the configs I can see
how it works, just need those XML files!! The ones from the 501 that I have
dont seem to
2010 Jan 28
3
Dial cellphone from one PBX1 to PBX2? is it possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
2009 Nov 27
1
problem with "dynformula" from "plm" package [RE-POST]
Hello list,
I'm following the paper (http://www.jstatsoft.org/v27/i02/paper) on
how to use "plm" to run panel regressions, and am having trouble with
what I believe should be something very basic.
When I run the command (p.9 in the paper):
R>
dynformula(emp~wage+capital,log=list(capital=FALSE,TRUE),lag=list(emp=2,c(2,3)),diff=list(FALSE,capital=TRUE))
I see:
emp ~ wage +
2010 Jan 25
3
Issue using tapply
Hello all,
I am trying to use the tapply function to sum some values and change the
column names of the resulting vector.
I input
Emp Et
1 10565 ACC
2 7515 ADM
3 625 AGF
4 6243 CNS
5 12721 EDU
6 3924 FIN
7 18140 HLH
8 3686 INF
9 15841 MFG
10 243 MIN
11 1864 MNG
12 4664 OSV
13 5496 PRF
14 4988 PUB
15 2166 REC
16 2153 REL
17 16082 RTL
18 3582 TRN
19 757 UTL
20
2010 May 06
1
question about rolling regressions
Hi All,
I am using R 2.11.0 on a Ubuntu machine. I have a time series data set and
want to run rolling regressions with it. Any suggestions would be useful.
Here are the details:
(1) I convert relevant variables into time series objects and compute first
differences:
vad <- ts(data$ALLGVA/data$GDPDEF, start=1948, frequency=1)
emp <- ts(data$ALLEMP, start=1948, frequency=1)
vad.dif1 <-
2006 Jul 06
4
Oracle HR on Rails
Interesting read...apologies if it has been posted already.
http://www.oracle.com/technology/pub/articles/saternos-rails.html
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2009 Mar 18
2
geometric mean of probability density functions
Hi,
This is my first time posting to the mailing list, so if I'm doing something
wrong, just let me know. I've taken ~1000 samples from 8 biological
replicates, and I want to somehow combine the density functions of the
replicates. Currently, I can plot the density function for each biological
replicate, and I'd like to see how pool of replicates compares to a
simulation I conducted
2005 Mar 08
1
To convert an adjacency list model into a nested set model
Dear R-help
I am wondering if somebody wrote some code to convert an adjacency list
model into a nested set model.
In principal I want to do the same as John Celko mentioned it here with
SQL:
http://groups.google.co.uk/groups?hl=en&lr=lang_en&selm=8j0n05%24n31%241
%40nnrp1.deja.com
Assume you have a tree structure like this
Albert
/ \
/
2006 Mar 22
5
Double Call Progress tones
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls
I get a double ring tone (UK style + US style). I also have a DECT phone
on a Sipura SPA-3000 configured with UK tones. This gives me a double
ring of UK + UK, so this
2016 May 03
2
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Whoops, email client auto-filled dev previously. Let's try this again.
Michael Maier wrote:
<snip>
> Ok - but this doesn't seem to answer my main question:
>
> Why must
>
> progressinband=never
>
> be applied especially if asterisk uses it by default? The big difference
> between w/ and w/o it is:
The default in 13 is "no" which still