RE: [Asterisk-Users] Why 's' doesn't take over unknown extension incontext ?

[Steve Mann]

I think it is "i" you want, "s" is the start for a context, meaning anything
coming in through that context will start there, "i" is invalid, and fires
if an invalid extension is keyed in that context.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Rozman
Sent: Monday, April 11, 2005 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Why 's' doesn't take over unknown extension
incontext ?


Hi,

I always thought that if there is no called extension in context, then 's'
extension is started (I use latest bristuffed Asterisk) 

I have context 'from-isdn' :
[from-isdn]
exten => s,1,Wait,2
exten => s,2,NoOp(ISDN call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number: ${CALLERIDNUM})
exten => s,3,SetCIDName(From ISDN: ${CALLERIDNUM})
exten => s,4,SetCIDNum(0${CALLERIDNUM})
exten => s,5,AGI,callerid_lookup.agi
exten => s,6,NoOp(After callerid_lookup.agi: ${CALLERID}: Name:
${CALLERIDNAME}, Number: ${CALLERIDNUM})
exten => s,7,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten => s,8,DBdel(DYNAMIC/${CALLERIDNUM})
exten => s,9,Background(custom/aa_1)
exten => s,10,Wait,5
exten => s,11,Dial(Local/[EMAIL PROTECTED]/n)

exten => s,108,Goto(from-pstn,s,1)   ;

exten => 99,1,Goto(s,1)   ;


Now if there is no line 99 on incoming call I get :
-- Extension '99' in context 'isdn-incoming' from '041461620' does
not exist.  Rejecting call on channel 0/1, span 1

Why doesn't extension 's' get started if extension 99 is unknown in
context from-isdn?

Thanks in advance,

regards,

Rob.



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RE: [Asterisk-Users] Getting a good deal on a PRI

[Steve Mann]

I was quoted about $700/month if I was within my downtown area for ISDN PRI.
So your price is in the right ball park.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of snacktime
Sent: Thursday, April 07, 2005 6:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Getting a good deal on a PRI


We have 10 incoming POTS lines to our offices, and a nortel norstar
pbx.  I've been looking at replacing it with * at some point in the
future, and the point that looks most cost effective is when we move
to PRI.

Problem is, I'm not really sure how to go about getting a good deal,
or what questions to ask.  90% of calls will be inbound.  I called up
Qwest and they quoted me $800 month.  I haven't called up any CLEC's
yet to see what they can do.

Any suggestions?  We are in Seattle, Washington.

Chris
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RE: Out of Office AutoReply: [Asterisk-Users] Voice controlledcalling?

[Steve Mann]

I have added him, and Joshua Chessman to my filter-to-trash rules. :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of dean
collins
Sent: Thursday, April 07, 2005 4:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: FW: Out of Office AutoReply: [Asterisk-Users] Voice
controlledcalling?


Thanks Robert, that information is really handy to know - now turn the
freaking out of office message off!

-Original Message-
From: Huddleston, Robert [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 07, 2005 4:57 PM
To: dean collins
Subject: Out of Office AutoReply: [Asterisk-Users] Voice controlled
calling?

Out on medical leave - I will return Monday 4/11/05


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RE: [Asterisk-Users] Zap (analog line) and volume

[Steve Mann]

Haha, this is trueit reminds me of my loud obnoxious car audio days,
competing in the dB drag events
I remember people telling me how they had to spend thousands of dollars to
double their amplifier wattage just to get 3 more dB on the competition.

It's funny how now when some guy pulls up with his stereo thumping, I feel
annoyedwhere did my youthful carelessness for my hearing go? :)

anyway, off topic, just feeling nostalgic



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David
Brodbeck
Sent: Thursday, April 07, 2005 3:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Zap (analog line) and volume


> -Original Message-
> From: Rich Adamson [mailto:[EMAIL PROTECTED]

> 3.0db of gain is roughly equal to doubling the volume to the
> human ear.

Actually, that's not true.  Each increase of 3.0 dB doubles the *power*.
But the human ear's response is logarithmic, and the decibel scale is also
logarithmic to reflect that.  1 dB is the smallest change in volume most
people can perceive.
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RE: [Asterisk-Users] Database lookups?

[Steve Mann]

Similar to my post about using a DB for authentication, you can use the Read
app to
1 - Play a recoding which asks for digits
2 - Indicate which variable to assign entered digits to
3 - Set the max length for the digits.
 
Once this function is called, the digits entered by the caller are stored in
a variable, the next line of your IVR can pass that variable to an AGI app.
Using any executable app such as Perl, PHP, etc. you can write a script that
reads in the variable passed, dive into a database, pull out the status,
then have asterisk play a different sound file based on the status using
either a specific AGI command, or using the AGI command "exec" to run an
asterisk command.
 
See:
 
http://www.voip-info.org/wiki-Asterisk+AGI
 
 
for more info.
 
Hope that points you in the right direction.
 
Steve

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jan Johansson
Sent: Thursday, April 07, 2005 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Database lookups?



Is it possible (How complicated is it?) to do this;

 

Call comes in, connects to IVR

IVR plays the usual "please type your order number, finish with pound"

 

Then I would like to query a MSSQL database server, looking up the "Status"
column from a row where ordernr = the entered order number.

 

Depending on the result of the lookup, play one of two messages  ("Yes,
ready for pickup" or "No, your order is not ready").

 

Can someone clue me in on which docs I should start with? Or is there an
example of this somewhere?

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RE: [Asterisk-Users] Can asterisk get code for cmd Authenticate fromDatabase

[Steve Mann]

You could use the Read App
(Read(variable[|filename][|maxdigits][|option])) in a dialplan to request
the user enter their code.
Then use AGI to launch an executeable script which is passed the variable
from the Read command, using this scripts native code to connect into the DB
and check if the password is correct, if not you can use AGI to have a
message played back to the user then hangup on them.

my 2 cents worth.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bartosz
Wegrzyn - asterisk
Sent: Thursday, April 07, 2005 11:41 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can asterisk get code for cmd Authenticate
fromDatabase


Hello,

I would like to store the account code for the user in mysql database.
Is the any way to retrieve that value and set for Authenticate cmd.
Is there any interface for asterisk to access database data?

Thanks

Bart,

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RE: [Asterisk-Users] ser <-> asterisk configs anyone?

[Steve Mann]

This may help, I just happen to be a google searching master :)

http://www.voip-info.org/wiki-Asterisk+at+large

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of G.Marshall
Sent: Wednesday, April 06, 2005 11:06 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ser <-> asterisk configs anyone?



I have searched high and low for these, but to no avail, nothing useful
back from google, nothing I could find on this mailing list, or
voip-user.org.

Does anyone have any good urls and or pointers which will assist in
configuring SIP Express Router and Asterisk talking to each other on the
same machine?

Many thanks,

Spencer

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RE: [Asterisk-Users] asterisk on UML

[Steve Mann]

I was already wrong about this with Music on Hold, but as far as any
documentation I can find, a timer of one flavour or another is still
required for MeetMe conferencing. Docs also state that a timer can HELP with
Music on hold quality

But as I said, I was already informed I was wrong about this once, so take
this with a grain of salt.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Cameron
Beattie
Sent: Tuesday, April 05, 2005 3:07 PM
To: snacktime; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk on UML


This may be a stupid quesion but why do you need ztdummy? If it's for timing
I thought that ztdummy is no longer required?

Regards

Cameron
- Original Message -
From: "snacktime" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, April 05, 2005 5:04 PM
Subject: [Asterisk-Users] asterisk on UML


>I just got a linode account and got * up and running without any
> problems.  I was going to ask them to load zaptel/ztdummy, but I was
> wondering if anyone else was interested in an * friendly UML hosting
> provider?  I might have more luck with getting them to load the kernel
> mods if there was more than just me interested in it.  I almost hate
> to ask them to do this just for me anyways.
>
>
> Also, any dangers/performance issues from an isp point of view for
> running ztdummy?
>
>
> Chris
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RE: [Asterisk-Users] asterisk sounds

[Steve Mann]

This does not contain a transcription of the sound files, but here is a link
to the sound files contained in the CVS:
http://asterisk.espia-net.net/horde/chora/cvs.php/asterisk-sounds/sounds?log
in=2

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Carlos
Rojas
Sent: Tuesday, April 05, 2005 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk sounds


Hi
here there are some of them

http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files%20inte
rnational

Carlos Rojas
Lima - Peru
On Apr 5, 2005 12:24 PM, Dov Bigio <[EMAIL PROTECTED]> wrote:
>
> Hello all,
>
> I am looking for a list of all available sound files for asterisk and a
> transcription of their content, so that I can have someone translate them
> into portuguese.
>
> Does anybody have a list of these files?
>
>
> Thank you
> Dov
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RE: [Asterisk-Users] "Multiplexing" (or what ever the term is) FXOports into a "Trunk"

[Steve Mann]

In the zapata.conf where you define your channels, you would also define
them as part of a group.
Then in your dial plan, when you execute the dial command, you would pass it
the ZAP/group_name

This will tell the dial command to use the first available channel within
the group you have defined.

see: http://www.voip-info.org/tiki-index.php?page=Channels%20and%20Groups
for more info.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David John
Walsh
Sent: Tuesday, April 05, 2005 1:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] "Multiplexing" (or what ever the term is)
FXOports into a "Trunk"


Hi all,

For an event we are doing, we have been donated several analogue PSTN
lines and an 8 port FXO bridge.

On the bridge, we have set up each of the ports to work on the SIP
protocol, and have referenced them, line1, line2, line3 etc for their
username / password.

I have placed the config in sip.conf, and they all work fine, inbound
and out - for testing anyway!

How do I get asterisk, to treat these 8 lines as one 8 call limit
trunk?  From a users perspective, all he/she needs to dial is
9 (where x's the number) to get any of the 8 outside lines?

Sure I could "hardcode" somthing in each part of the extensions.conf,
but if this trial is sucsessful, the number of lines may increase, and
it would be nice to define the array once as it were.

(I am aware that most of my troubles would go away if I used a more
intelligent termination such as ISDN, but for several issues, its not
possible)

Thank you for your time on this matter.

David
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RE: [Asterisk-Users] broadvoice

[Steve Mann]

Maybe try type=friend as opposed to type=peer

Still a newbie, but my understanding from what I read is that a peer is call
out only.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of hugolivude
Sent: Monday, April 04, 2005 3:11 PM
To: Matt; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] broadvoice


Woops forgot to include my config files...

;*
;/etc/hosts
# Do not remove the following line, or various programs
# that require network functionality will fail.
127.0.0.1   localhost.localdomain   localhost
# proxy.dca.broadvoice.com
147.135.0.128   sip.broadvoice.com
;
;*
;
;/etc/asterisk/sip.conf
;
[general]
port=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0  ; Address to bind to (all addresses on machine)
context=from-sip-external ; Send unknown SIP callers to this context
pedantic=no
register =>
[EMAIL PROTECTED]::[EMAIL PROTECTED];/3003
;
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8145551212
secret=
username=8145551212
insecure=very
context=from-broadvoice
authname=8145551212
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
;
;*
;
/etc/asterisk/extensions.conf
[general]
static=yes   ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
;
;
[from-broadvoice]
exten => s,1,Dial(ZAP/1,30)
exten => s,2,Hangup

[from_FXS]
exten => _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30)
exten => _1NXXNXX, 2, congestion()
exten => _1NXXNXX, 102, busy()
;
;*
;
;/etc/asterisk/zapata.conf
;
;
[channels]
language=en
context=from-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
channel => 4
;
language=en
context=from_FXS
signalling=fxo_ks
channel=>1
;
language=en
context=from-ILS-FXS
signalling=fxo_ksFailed to authenticate on INVITE to '"asterisk"
;tag=as4a325b3a'
channel=>2
;
;*
;
;/Asterisk Console Output
;
Asterisk Ready.
*CLI> sip show registry
Host  Username Refresh State
147.135.0.128:50608145551212   120 Registered
*CLI> -- Starting simple switch on 'Zap/1-1'
   -- Executing Dial("Zap/1-1",
"SIP/[EMAIL PROTECTED]|30") in new stack
   -- Called [EMAIL PROTECTED]
Mar 27 20:55:26 NOTICE[1116941248]: chan_sip.c:5047 handle_response:
Failed to authenticate on INVITE to '"asterisk"
;tag=as4a325b3a'
Mar 27 20:55:26 WARNING[1209214528]: app_dial.c:347 wait_for_answer:
Unable to forward voice
 == Spawn extension (from_FXS, 13035551212, 1) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'

On Apr 4, 2005 2:29 PM, Matt <[EMAIL PROTECTED]> wrote:
> Hi,
> I'm currently routing my asterisk server out over broadvoice.. it
> seems I can do multiple outgoing and incoming calls does anyone
> know if broadvoice actually allows this or not?
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RE: [Asterisk-Users] monmp3thread: Request to schedule in the past?!?!

[Steve Mann]

>From what I have read, you made a small mistake, if you are not using Digium
hardware, but want to use MeetMe of Music on Hold, you still require a
timing source, regardless of kernel.

Kernels earlier then 2.6 without Digium must have either ztdummy or zaprtc
running.
In this case you can only use ztdummy if you have a usb-uhci device, ohci
will not work, only USB-UHCI, if you do not have usb-uhci listed when using
lsprob, then you will need to use zaprtc.

For Kernels 2.6.x and up without digium, you still require a timing source,
in this case though, ztdummy will be able to use the processor as a timing
source, so as of 2.6, you only need to use ztdummy.

So, either way, you need a timing source, which one depends on the above
mentioned.

If you have digium hardware, then you have the timing source needed and
should not be loading ztdummy or zaprtc.

Steve

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kanuri,
Seshu (Company IT)
Sent: Monday, April 04, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] monmp3thread: Request to schedule in the
past?!?!


Give us the full story. Which Linux core 2.4 or 2.6 you are on? 2.6 does
not need Ztdummy, if you don't have a Zaptel card.

1) Do you have a Zap device?
2) or are you using Ztdummy?
3) your 'lsdev' out put from linux commandline.
4) Your sip.conf entry.
5) your extensions.conf entry which is causing the error.
6) your musiconhold.conf entry

Seshu
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Glenn
Powers
Sent: Monday, April 04, 2005 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] monmp3thread: Request to schedule in the
past?!?!


I keep getting this error every five minutes:

Apr  4 13:35:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!

Apr  4 13:40:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:40:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:40:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:40:02 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!


I'm running  CVS-v1-0-03/08/05-09:27:38. How can I fix this?

thanks,
glenn

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RE: [Asterisk-Users] broadvoice

[Steve Mann]

I think I remember seeing someone talking about this, and they stated that
after a while broadvoice caught on and back-billed them for x number of
months worth of Business Level service. So, basically, I don't think I would
tempt fate without a decent bankroll just in case.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kerry
Garrison
Sent: Monday, April 04, 2005 1:48 PM
To: 'Matt'; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] broadvoice


I called them about this and the vauge answer I got was that you get 2
connections per account in order to allow the equivilant of a line with call
waiting. While there is no hard-wired limitation that I know of, it is best
not to abuse it so as to prevent them from enforcing one.
-Kerry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, April 04, 2005 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] broadvoice

Hi,
I'm currently routing my asterisk server out over broadvoice.. it seems I
can do multiple outgoing and incoming calls does anyone know if
broadvoice actually allows this or not?
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RE: [Asterisk-Users] hardware requirements

[Steve Mann]

This is because, depending on what you are using it for, Asterisk will scale
differently.
VOIP and multiple codes require more resources then a vanilla FXO/FXS PBX,
etc. and so on.
 
2 links that may help you:
http://www.voip-info.org/wiki-Asterisk+dimensioning
 
http://www.voip-info.org/wiki-Asterisk+hardware+recommendations
 
 
Steve

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Antonio Airoso
Sent: Monday, April 04, 2005 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] hardware requirements


I have looked around and can't seem to find a good page that has information
for how many users to a server and how much RAM and CPU speed to a user.
thank you

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[Asterisk-Users] Does the agent queue app support Aftercall and AUX agent status?

[Steve Mann]

In most call centers 
I have worked in, the agents had the ability to change their status from "auto 
ready" or "available" into an AUX of After call state, Aftercall basically works 
like wrap time, in that the agent would not receive another call in the queue 
until their status was manually changed back to auto ready by a specific key 
combination on the dial pad. Aux worked in that the agent could change their 
state from auto ready into an AUX state where the press a code that indicates 
what type of AUX state they are in, an example would be Aux-Break, or 
Aux-Supervisor feedback (for tracking of time, etc.)
 
Example, I am an 
agent, I receive a call, while on the call, I dial a special key code, and then 
when the call disconnects, instead of going right back to the queue and 
receiving another call, I go into an After Call state, where I can write notes, 
and log the call, etc. Then I would dial another sequence to put me back into a 
ready state.
 
An example of Aux, I 
am an agent, I am in a queue, but not on a call, my supervisor calls my 
extension and says they need to discuss one of my previous calls with me. I dial 
a code, and my state is changed from auto ready into an Aux state, where I then 
dial an additional digit to indicate why I am in Aux, example: Feedback, Break, 
etc. I then get back to my desk, and dial a new code to place me back into a 
ready state.
 
I know that with 
Asterisk, you can program a wrap time option to allow the CSR X number of 
seconds or minutes of wrap time before receiving another call, but I am looking 
for the above functionality over and above the simple implementation of wrap 
time.
 
I do not want to 
just have the agents log in, and out when they don't want a call, but instead 
use the functionality I described above as a time keeping system for payroll, 
reporting, and agent tracking purposes.
 
I sent an email, 
with a more ambiguous subject line about this subject, and received no response, 
so I am hoping with a better subject line, someone may open the 
email.
 
In that previous 
email I mentioned that on the digium homepage's FAQ it listed some call center 
terminology that detailed the above mentioned functionality, but I can not find 
and documentation on it, so I am hoping it exists, but has not been documented 
yet, and that someone out there has used it, or knows if it truly does exist, or 
if I am out of luck.
 
The link to the FAQ 
section:
http://www.digium.com/index.php?menu=faq#General_7
 
If I am asking the 
wrong list, if someone knows a better place to ask this question, please let me 
know.
 
Thanks,
 
Steve MannNetwork 
AdministratorFineLine Solutions
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[Asterisk-Users] ACD Functionality

[Steve Mann]

Hello 
All,
 
I have been trying 
to find a definitive yes or no on this, and have decided to post it to the list 
since I can not find and documentation on it.
 
On the Digium FAQ, 
there is a section on ACD terminology which lists the 
following:
 
ACD_IN (status) - Agent is on an ACD 
CALLEXT_OUT (status) - Agent in on an outbound non-ACD call.EXT_IN 
(status) - Agent is on a non-acd inbound or internal call.HOLD (status) - 
Agent has placed a call on hold.ACW (status) - Agent is in "After Call Work" 
mode.AUX (status) - Agent has selected an aux work mode to avoidcalls 
while remaining logged into the system. Typical AUX codes might include one 
for "supervisor assistance", breaks, answering customer email, performing 
callbacks, as well a a "default" aux statusthat agents logging into the 
system are automatically placed in until they indicate thatthey are ready to 
take calls. This status typically is used also aftera RONA event (roll over 
no answer). RONA (event) - roll over no answer - a call was routed to the agent 
but the call was not picked up. (A flogging offense in most callcenters 
;)
 
Since it is on the 
FAQ, this may have mistakenly lead me to believe that there is functionality for 
an agent to place themselves into an AUX or ACW state.
However, I can not 
find any documents on the cmd's or configuration for an agent to be able to do 
this.
 
Can anyone comment 
as to whether the agent and queue features of the switch actually have the 
functionality to dial a code, or press a button on an ADSI phone, and have their 
state changed into ACW or AUX, etc. so they stop receiving calls, but stay 
logged in?
 
Thanks,
 Steve
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