Displaying 20 results from an estimated 10000 matches similar to: "How does holdtime get calculated for queues"
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2016 May 11
2
How is Queue avg holdtime and avg talktime calculated
2011 Jun 25
1
Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"
Hi All;
Again, the Cisco IP Phones 7942G and using Skinny:
I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file.
The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination
2011 Jan 10
0
No subject
takes precedence over a queue's defined moh class.
--=20
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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<div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas =
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2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
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2011 Apr 12
0
No subject
be able to setup a SIP trunk. I've been able to successfully integrate a
Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine
you should be able to do the same here.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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<div
2013 May 01
1
multiple provider for incoming
Matt,
At some point you need to consider how much is too much...
I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely
2011 Jan 04
0
Queues, priorities and (miscalculated) holdtimes
Anyone ever noticed that the reported holdtime is wrong when there are
different priorities? Also talktime is 0, but for the moment I don't
care.
"queue show test" reports:
test has 23 calls (max unlimited) in 'ringall' strategy (193s holdtime,
0s talktime)
[...]
Callers:
1. Local/351 at default-8828;2 (wait: 3:32, prio: 15)
2. Local/351 at default-8361;2
2006 Mar 16
2
Queues Not Reporting Estimated Hold Time
I am running 1.2.5 with a simple queue and have announce-holdtime = yes
in queues.conf for that queue. The person is being told their posistion
in the queue and the CLI says the estimated hold time, but it never
plays it for the caller. It worked previously, i am not sure when it
stopped, i think after 1.2.1. Is this a known bug? I dont want to
report it to the bug tracker if its already been
2006 Jan 17
1
ACD announce-holdtime
Has anyone gotten announce-holdtime in queues.conf to work? Doesn't seem to matter what combination of options I use, I can't get this particular setting to do what the docs say.
Thanks.
Doug.
2011 Apr 11
3
changing port 5060 to 5061
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a way..........................
On 4/10/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
> Send asterisk-users mailing
2010 Jul 23
1
Attended Transfer question
I've been asked to implement the following transfer workflow in an asterisk
system, and I'm not seeing an easy way to do the bolded steps below (steps 4
and 5 for those with a text-only email client):
1 - Put the call on hold
2 - Call the extension for the staff member needed
3 - Give them a rundown of the caller and situation
*4 - Bring the caller on with the staff member the call will
2009 Mar 20
3
Queues Announce help request.
I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some "comforting" voice announcements would be nice.
Below is the queues.conf file relevant portions.
Member phone number is munged to protect the guilty.
We shouldn't need the announcement source info, but I have been trying
everything.
The problem is with the member busy, we get no
2007 Mar 08
2
Queue announcing hold sequence instead of hold time
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian
Sarge) and the behaviour of our Call Centre queues has changed slightly.
Before the upgrade, when a caller was waiting in the queue, the
estimated hold time was announced as expected ("estimated hold time is
less than 2 minutes ...").
Now the caller gets an announcement of their sequence in the queue
2006 Feb 28
0
playing hold time announcement without queue position announcement
Greetings fellow list members,
I have what I think is a relatively simple question, but it did not
appear to be addressed on the wiki. I am trying to setup a queue so
that it plays an estimated holdtime announcement, but not a queue
position announcement. Currently my dialplan does both, and while I
know how to take out the estimated holdtime without affecting the queue
position
2010 Jul 13
3
STRFTIME function declared in globals context
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
If I evaluate these variables in the dialplan later, using
exten
2009 Nov 30
1
Polycom 500 format file system on every reboot
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone, then they
receive an error about not being able to find the config file and then the
phone will not boot up. Has
2010 Jul 26
1
VPMADT032 Failed! Unable to ping the DSP (2)!
Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a
week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1
VPMADT032 Module, hooked up to 5 analog lines. I get the error message
referenced in the subject in my dmesg output everytime I load / reload DAHDI
using the command "system dahdi start/restart". When I make an outbound
call over
2010 May 26
1
Error compiling DAHDI...
I was at a client site tonight to install OSLEC on his machine running
asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and
DAHDI, downloaded the latest version of DAHDI 2.2.1
(dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to
compile OSLEC with DAHDI, but I ran into compilation issues that I had never
seen before. So as a test I deleted my
2013 Jan 16
1
Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
I'm trying to decide if I need to open an issue for this or if it's just a
misconfiguration issue of some sort. Here's the situation - yesterday
morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS
5.8 installation and got a shell of a basic asterisk install setup (minimum
required configuration files, etc, with no dialplan or sip peers setup
yet). In the