Displaying 20 results from an estimated 2000 matches similar to: "Polycom - Buddy Watch not a choice when adding Speed Dial"
2008 Mar 14
3
Anyone know of a pass through ATA
Anyone know of a company that makes a pass through ATA?
By pass through I mean have an Ethernet switch built into the ATA, like most
desktop phones have.
All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN
ports.
I fooled around with DMZ etc...but it just doesn't work as well.
Thermal
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2009 May 12
2
Is anyone keeping up with the versions?
We are still using 1.4 and were going to start testing with 1.6.0, but then
1.6.1 was released and now 1.6.2 is already in beta 2.
That seems like a lot of independent releases to maintain. I read about all
the regressions ans hurried dot releases, makes us nervous.
How is everyone doing their testing?
-Matt
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2006 May 02
3
Queue reporting seems broken.
I am trying to figure out which one of our agents is answering the calls.
According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the
only time the queue_log puts the channel (agent) is during logoff & logon.
There is the connect & completeagent message, but it doesn't show which
channel (agent) answered the phone.
I can't even figure it our cross referencing the
2008 Mar 14
1
Group Listen on SIP Phone
Anyone know of a SIP phone that supports group listen?
Group listen allow you use the handset but what the far end says comes out
the speaker...it is F802 on a Norstar.
Thermal
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2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP trunk.
Something like this in sip.conf [general] would be in effect for all SIP
clients:
jbenable = yes
jbmaxsize = 150
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes
I only want
2010 Apr 13
1
Interesting One Way Audio
I have an Asterisk box, 1.4.30 with a PRI.
A Mitel 3300 is connected to the Asterisk box via SIP trunking.
When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.
But - when I route the Mitel user to echo() it works, send and receive. The
Mitel user also can record and playback greetings.
One thing I have noticed is that when the Mitel user
2007 Aug 08
1
Buddy watch and the hint priority - brain teaser
Apologies if this is a resend, but I've sent this 12 hours ago and still
can't see it on the list.
Hi,
I've just started to setup my phones with Buddy watch. Basically, it all
works fine when using the simple example on the wiki:
exten => 123,hint, SIP/some_sip_reg
exten => 123,1,SIP/some_sip_reg
BUT, what I need to do is dynamically decide where the hint checks for buddy
2006 Mar 06
1
Buddy watch?
Hi,
I am using Polycom 501 and I came across a problem. As soon as I have
incominglimit=1 in sip.conf, which is necessary for buddy watching, I
cannot transfer calls. On the console it tells me:
Call from user '3052' rejected due to usage limit of 1. Can someone
please tell me how to get around this problem?
(I don't know if this is relevant, but in the phone.cfg file, I have
2006 Jun 14
1
Please Help - Polycom IP 601 Buddy Watch problems
Hi,
I found your post on
http://threebit.net/mail-archive/asterisk-users/msg04580.html
I am having the exact same issue with the Polycom IP601 (SIP version
1.6.6.0036) with Asterisk 1.2.7.1.
I was wondering if you found any solution to it. I would really appreciate
if you could share your solution.
Thanks,
Khairul.
BELOW, THIS WAS YOUR POST
Polycom IP 601 Buddy Watch problems
2006 Feb 24
1
Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
Hi,
I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured.
Before the reload in the CLI appears:
-= Registered Asterisk Dial Plan Hints =-
3002 : SIP/3002 State:
2007 Jun 06
0
SIP buddy watch
Hi all,
Have some one advice me on the following needs?
1) Clients need to use 50 Hard IP Phone, and 20 Soft-Phone.
2) Clients have 20 Hard IP phone and 10 Soft-phone within Operational
Department.
3) The Soft-phone mostly is division head, and they wish to monitor
the online status of the existing hard-phone and soft-phone through the
soft-phone. The status like
2006 Oct 11
4
Multiple TE110P cards in one chassis
Does anyone know if you can have multiple TE110P cards in one chassis?
-Thermal
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2006 Feb 22
4
Polycom IP 601 Buddy Watch problems
Hi,
I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function.
Is this a specific problem of asterisk-1.2.3? How can I solve it?
Thank in advance, regards,
Marco.
2008 May 08
3
Looking for a Snom expert
I would like to hire someone to help us tweak our asterisk system for Snom
phones.
We would like to enable things like:
One touch recording
One touch park orbits
Presence
Please contact off-list if you will be able to help.
Thermal
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2010 Oct 08
2
Polycom getting DCHP address from wrong VLAN
Hello,
I have been tearing my hair out on this issue for 2 days, any help
would be appreciated.
We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch
There are two VLANs, 1(data) & 50(VoIP). When Polycoms are connected
to the switch with VLAN 50 hard coded in the config they grab a DHCP
address from VLAN 1, the PVID for the switch port.
The ports have membership in
2009 Feb 11
2
problem with 'which' and strings
I have written a function that goes through a database and calculates various
metric scores and allocates them to a data set. For 400 of the 500 sites
that I calculate these metrics for works fine and allocates the scores into
the appropriate column. For some reason, some sites I run into the below
problem:
Here is what I am doing:
> names(orig.metric)
[1] "BenInsect"
2009 Jul 15
1
Error in simulation R-code
Dear List,
I have got error message when I run the R-code. Can anyone has a suggestion?
v.code <- df.bm7[,c(10:31)]; v.code[1:3,]
names(v.code)
CM = v.code # variable binomial code
sim.sp <- function(data,CM,n,N)
{
C <- matrix(rep(NA,N),ncol=1)
for(i in 1:N)
{
j <- n
xx <- which(colSums(CM[j,])==1)
V <- names(xx)
V <- paste(V,
2009 Jun 30
1
How to wrap my (working) code in a loop or function? (loop/function newbie alert)
Dear R-helpers,
I have split a dataframe into a list with five elements, with the following
code:
> datalist<-split(data,data$UNIT)
I would now like to run some code (below) on each element of the list to
extract rows from the list elements; then I would like to rbind the
extracted rows into a new dataframe containing all of the extracted rows
from all of the list elements.
I don't
2010 Apr 20
1
Put a call on hold with Manager
I would like to be able to place a call on hold via the manager interface
and be able to retrieve it.
The user can click a button in the Order entry form to put the caller on
hold when they are looking up information. It saves them from having their
hands leave the keyboard and press hold on the phone.
I don't see 'hold' & 'retrieve' commands for the manager interface.
2010 Jul 14
1
Can't compile DAHDI - wrong kernel source
I have a virtual server with godaddy but can not compile DAHDI as it
complains that I do not have the correct kernel source.
The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest
version
Nothing to do
uname -a returns:
Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed
Aug 26 13:11:07