Re: [asterisk-users] Queue Question

2009-08-21 Thread Jim Dickenson

Here is my dialplan for my support queue:

exten => 201,1,Verbose(2,Doing support call)
exten => 201,n,Answer()
exten => 201,n,Wait(0.5)
exten => 201,n,Set(qac=${QUEUE_MEMBER(support,free)})
exten => 201,n,GotoIf($[${qac} > 0]?HAVEAGNT)
exten => 201,n,Verbose(2,No agents free in support queue)
exten => 201,n,Playback(cfmc/support-no-agent)
exten => 201,n,Voicemail(2...@ourvm,u)
exten => 201,n,Playback(goodbye)
exten => 201,n,Hangup()
exten => 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten => 201,n,Verbose(2,Queuing caller for support agent)
exten => 201,n,Queue(support,nrt,,,120)
exten => 201,n,Verbose(2,Support agent did not answer call)
exten => 201,n,Voicemail(2...@ourvm,b)
exten => 201,n,Playback(goodbye)
exten => 201,n,Hangup()


;  Agent login logout 
exten => *20,1,Verbose(2,Doing agent login/logout)
exten => *20,n,Answer()
exten => *20,n,wait(.0.5)
exten => *20,n,Read(AgentNumber,agent-user)
exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER)
exten => *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten => *20,n,GotoIf($["${AgentStatus}"="1"]?VERIFY)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?VERIFY)
exten => *20,n(NOUSER),Playback(cfmc/bad-agent)
exten => *20,n,Playback(vm-goodbye)
exten => *20,n,Hangup()
exten => *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?AGENTOFF)
exten => *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten => *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten => *20,n,AddQueueMember(support,Local/Queue$ 
{agentnumb...@ansqueue${C$

;   AQMSTATUS can be  ADDED | MEMBERALREADY | NOSUCHQUEUE
exten => *20,n,Playback(agent-loginok)
exten => *20,n,Verbose(2,Agent ${AgentNumber} added ${DB(users/$ 
{UserID}/AgentD$

exten => *20,n,Hangup()
exten => *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten => *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten => *20,n,RemoveQueueMember(support,Local/Queue$ 
{agentnumb...@ansqueue)

exten => *20,n,Playback(agent-loggedoff)
exten => *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten => *20,n,Hangup()

[ansqueue]
exten => _Queue.,1,Set(AgentNumber=${EXTEN:5})
exten => _Queue.,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => _Queue.,n,Set(AgentDevice=${DB(users/${UserID}/AgentDevice)})
exten => _Queue.,n,Verbose(2,Agent ${AgentNumber} status is $ 
{DEVICE_STATE(${Ag$
exten => _Queue.,n,GotoIf($["${DEVICE_STATE($ 
{AgentDevice})}"="NOT_INUSE"]?DIAL$

exten => _Queue.,n,Busy()
exten => _Queue.,n,Hangup()
exten => _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten => _Queue.,n,Hangup()

--
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Aug 21, 2009, at 1:00 PM, James A. Shigley wrote:


First off this is not my work for extensions.conf it is modified from
http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbacklogin-to-standard-dialplan-methods-part-1/
So credit to Leif Madsen

But as to my question

[AgentLogin]
;A replaced version of AgentCallbackLogin() using a GoSub()
;
exten => XX,1,Verbose(2,Logging in agent)
exten => XX,n,WaitExten(5)
exten => XX,n,GoSub(AgentCallbackLogin,start,1)
exten => XX,n,Hangup()

[AgentLogOut]
exten => XX,1,RemoveQueueMember(9819930,DAHDI/g1/$ 
{CALLERID(num)})


; calling 'primary' queue
;
exten => XX,1,Verbose(2,Calling into the primary queue)
exten => XX,n,Playback(silence/1)
exten => XX,n,Queue(9819930)
exten => XX,n,Hangup()
;
[AgentCallbackLogin]
; conversion of AgentCallbackLogin() to using AddQueueMember()
;
exten => start,1,Verbose(2,Logging in agent)
exten => start,n,Playback(silence/1)
exten => start,n,Read(AGENT_USERID,agent-user)
exten => start,n,VMauthenticate(${agent_user...@default)
exten => start,n,AddQueueMember(Queue,DAHDI/g1/${CALLERID(num)})
exten => start,n,Playback(agent-loginok)
exten => start,n,Return()


Queue Context from Queues.conf

[Queue]
music=default
strategy=linear
timeout=5
retry=5
wrapuptime=0
maxlen = 0
announce-frequency = 0
announce-holdtime = no

Ok Here is how I would like this queue to work

First try and deliver the call to the Dynamic Agents who login using  
the above Setup from extensions. I of course want that to be linear  
in fashion with 5 seconds to try each one. BUT if none of the  
dynamic agents are available or if there are no dynamic agents I  
want to send that call to an interface/context. We are an answering  
service. If no agent is available in the queue I want to send it to  
the Interface which goes to my TAS equipment.


So DAHDI/g2/Exten

How do I accomplish that because I can’t figure it out from googling  
or http://www.voip-info.org/



James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,

CONFIDENTIALITY NOTICE: This email, including any attachments,  
c

[asterisk-users] Queue Question

2009-08-21 Thread James A. Shigley
First off this is not my work for extensions.conf it is modified from

http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl
ogin-to-standard-dialplan-methods-part-1/

So credit to Leif Madsen  

 

But as to my question

 

[AgentLogin]  

;A replaced version of AgentCallbackLogin() using a GoSub()   

; 

exten => XX,1,Verbose(2,Logging in agent)  

exten => XX,n,WaitExten(5) 

exten => XX,n,GoSub(AgentCallbackLogin,start,1)  

exten => XX,n,Hangup()

 

[AgentLogOut]

exten =>
XX,1,RemoveQueueMember(9819930,DAHDI/g1/${CALLERID(num)})   

 

; calling 'primary' queue  

;  

exten => XX,1,Verbose(2,Calling into the primary queue)  

exten => XX,n,Playback(silence/1)  

exten => XX,n,Queue(9819930)  

exten => XX,n,Hangup()

;

[AgentCallbackLogin]  

; conversion of AgentCallbackLogin() to using AddQueueMember()

;

exten => start,1,Verbose(2,Logging in agent)  

exten => start,n,Playback(silence/1)  

exten => start,n,Read(AGENT_USERID,agent-user)  

exten => start,n,VMauthenticate(${agent_user...@default)  

exten => start,n,AddQueueMember(Queue,DAHDI/g1/${CALLERID(num)})  

exten => start,n,Playback(agent-loginok)  

exten => start,n,Return()  

 

 

Queue Context from Queues.conf

 

[Queue]

music=default

strategy=linear

timeout=5

retry=5

wrapuptime=0

maxlen = 0

announce-frequency = 0

announce-holdtime = no

 

Ok Here is how I would like this queue to work

 

First try and deliver the call to the Dynamic Agents who login using the
above Setup from extensions. I of course want that to be linear in
fashion with 5 seconds to try each one. BUT if none of the dynamic
agents are available or if there are no dynamic agents I want to send
that call to an interface/context. We are an answering service. If no
agent is available in the queue I want to send it to the Interface which
goes to my TAS equipment.

 

So DAHDI/g2/Exten 

 

How do I accomplish that because I can't figure it out from googling or
http://www.voip-info.org/

 

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

<>___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] queue question

2008-12-22 Thread David fire
hi
this is the situation i have a queue and i am monitoring when an important
customer call i want to take it off  from the queue and send it to an agent
directly can i do that? how?
thanks
David


-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queue question

2008-12-18 Thread Giedrius Augys
2008/12/18 David fire 

> you can use the h extencion, when a call is hangup this extencions is
> executed
> (doble posting in action)
>
> 2008/12/18 Giedrius Augys 
>
>>  Hello,
>>
>>Is it possible, that after the call was established between client and
>> agent and one of the them hangups the call , the cmd queue executes cmd
>> (gosub, macro) or something...
>> Thanks for advance
>>
>>
>> --
>> Pagarbiai  / Best Regards,
>> Giedrius Augys
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> you can use the h extencion, when a call is hangup this extencions is
> executed
> (doble posting in action)
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi , David,

  I forgot to mention that I'm using Asterisk 1.6.0.1 version and ael. The
reason to execute some commands is: when the call is established , cmd queue
'launches' macro or 'gosub', so when I setup some variables or insert data
to database (agent interface, channel and so on). But when the call finishes
between agent and client, I want to remove information about this call. I
want to do, for example: recording on demand , not using dtmf , but agent on
the web clicks the button during conversation 'start to record'. Maybe where
is easiest way... ?

-- 
Pagarbiai  / Best Regards,
Giedrius Augys
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queue question

2008-12-18 Thread David fire
you can use the h extencion, when a call is hangup this extencions is
executed
(doble posting in action)

2008/12/18 Giedrius Augys 

> Hello,
>
>Is it possible, that after the call was established between client and
> agent and one of the them hangups the call , the cmd queue executes cmd
> (gosub, macro) or something...
> Thanks for advance
>
>
> --
> Pagarbiai  / Best Regards,
> Giedrius Augys
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

you can use the h extencion, when a call is hangup this extencions is
executed
(doble posting in action)

-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] queue question

2008-12-18 Thread Giedrius Augys
Hello,

   Is it possible, that after the call was established between client and
agent and one of the them hangups the call , the cmd queue executes cmd
(gosub, macro) or something...
Thanks for advance


-- 
Pagarbiai  / Best Regards,
Giedrius Augys
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Queue Question

2008-12-16 Thread Mark Michelson
Sebastian wrote:
> Is this going to be released in any 1.6 version soon??
> 
> Your branch (queue-reset) is supouse to be the same as trunk but with this
> functionality?
> Is this branch updated every time trunk is committed?? I checked the log and
> seems to have the latest commits of trunk, but I would like to be sure.
> 
> 
> Thanks
> 
> 
> Regards,
> 
> Sebastian

This branch is based off of Asterisk trunk and is automatically updated once an 
hour with the latest updates to trunk.

As far as when this will make it into 1.6, it is unknown. There are still a few 
minor tweaks that need to be made and I'll probably do some testing and code 
review. Then it will need to go through the peer review process and a version 
will be set for its release. If I had to make an estimate, it will be in 1.6.2 
at the earliest, but 1.6.3 seems more likely given that I haven't put a lot of 
work into this branch lately due to more pressing matters.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Queue Question

2008-12-15 Thread Sebastian
Is this going to be released in any 1.6 version soon??

Your branch (queue-reset) is supouse to be the same as trunk but with this
functionality?
Is this branch updated every time trunk is committed?? I checked the log and
seems to have the latest commits of trunk, but I would like to be sure.


Thanks


Regards,

Sebastian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Sent: lunes, 15 de diciembre de 2008 09:00 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Queue Question

Is this going to be realeased in any 1.6 version son??
Thanks


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: lunes, 15 de diciembre de 2008 08:14 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue Question

Sebastian wrote:
>  
> 
>  
> 
> Hi,
> 
>  
> 
> In queues realtime, when the queue start and when it ends.
> 
> I mean, for example to calculate service level, how many calls, etc.
> 
> If I want to start the queue from with 0 calls, etc, how do I do this? 
> And if I want to stop it, so I can start it again??
> 
>  
> 
> Thanks!!
> 
>  
> 
> Regards,
> 
>  
> 
> Sebastian
> 

Currently, there is not a way to do this with realtime queues. During a
reload, 
realtime queues are not touched at all. I have a development branch set up
which 
is supposed to help this as well as other rigidities present when it comes
to 
reloading and resetting queues. The branch is located at the following URL
if 
you wish to give it a test:

http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset

If you run the code there, you'll find that there is a command called "queue

reset stats" which should do what you want.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

__ Information from ESET Smart Security, version of virus signature
database 3693 (20081215) __

The message was checked by ESET Smart Security.

http://www.eset.com


 

__ Information from ESET Smart Security, version of virus signature
database 3693 (20081215) __

The message was checked by ESET Smart Security.

http://www.eset.com
 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

__ Information from ESET Smart Security, version of virus signature
database 3693 (20081215) __

The message was checked by ESET Smart Security.

http://www.eset.com


 

__ Information from ESET Smart Security, version of virus signature
database 3693 (20081215) __

The message was checked by ESET Smart Security.

http://www.eset.com
 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Queue Question

2008-12-15 Thread Sebastian
Is this going to be realeased in any 1.6 version son??
Thanks


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: lunes, 15 de diciembre de 2008 08:14 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue Question

Sebastian wrote:
>  
> 
>  
> 
> Hi,
> 
>  
> 
> In queues realtime, when the queue start and when it ends.
> 
> I mean, for example to calculate service level, how many calls, etc.
> 
> If I want to start the queue from with 0 calls, etc, how do I do this? 
> And if I want to stop it, so I can start it again??
> 
>  
> 
> Thanks!!
> 
>  
> 
> Regards,
> 
>  
> 
> Sebastian
> 

Currently, there is not a way to do this with realtime queues. During a
reload, 
realtime queues are not touched at all. I have a development branch set up
which 
is supposed to help this as well as other rigidities present when it comes
to 
reloading and resetting queues. The branch is located at the following URL
if 
you wish to give it a test:

http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset

If you run the code there, you'll find that there is a command called "queue

reset stats" which should do what you want.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

__ Information from ESET Smart Security, version of virus signature
database 3693 (20081215) __

The message was checked by ESET Smart Security.

http://www.eset.com


 

__ Information from ESET Smart Security, version of virus signature
database 3693 (20081215) __

The message was checked by ESET Smart Security.

http://www.eset.com
 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Queue Question

2008-12-15 Thread Mark Michelson
Sebastian wrote:
>  
> 
>  
> 
> Hi,
> 
>  
> 
> In queues realtime, when the queue start and when it ends.
> 
> I mean, for example to calculate service level, how many calls, etc.
> 
> If I want to start the queue from with 0 calls, etc, how do I do this? 
> And if I want to stop it, so I can start it again??
> 
>  
> 
> Thanks!!
> 
>  
> 
> Regards,
> 
>  
> 
> Sebastian
> 

Currently, there is not a way to do this with realtime queues. During a reload, 
realtime queues are not touched at all. I have a development branch set up 
which 
is supposed to help this as well as other rigidities present when it comes to 
reloading and resetting queues. The branch is located at the following URL if 
you wish to give it a test:

http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset

If you run the code there, you'll find that there is a command called "queue 
reset stats" which should do what you want.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Queue Question

2008-12-15 Thread Sebastian
 

 

Hi,

 

In queues realtime, when the queue start and when it ends.

I mean, for example to calculate service level, how many calls, etc.

If I want to start the queue from with 0 calls, etc, how do I do this? And
if I want to stop it, so I can start it again??

 

Thanks!!

 

Regards,

 

Sebastian

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Queue Question

2007-09-20 Thread Kevin Smith
Hi Jeremy,
A few thoughts that come to mind. We have a queue that is open between 
certain hours. I have a few checks in place before a caller enters, 
first it checks to see if there it is within the time window, then 
checks to see if there are any agents log into queue, if any fail they 
get our closed message. Sounds like you are trying to do something similar.
Not sure what you have for extension numbers numbers, but you will get 
the idea.

Your first friend:
GotoIfTime(|||?[[context|]extension|]pri)
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime

I don't know how your dial plan is structured. My guess is the "after 
hours" operation is in a separate part of the code from the other. Since 
we are just looking at after hours, I would use the reverse on your 
time. Because the command jumps when the statement is true. I do not 
know what will happen if you say go from 17:00-8:00, but you can try it.

Example:
exten => 800,1,GotoIfTime(8:00-17:00|mon-fri|*|*?NormalOp,900,1) ; Since 
this will fail if it is 9pm, it moves on to the next priority in this 
exten.

[NormalOp]
exten => 900,1,blah

Next, is your other test. Use the queue agent count function 
QUEUEAGENTCOUT(queuename)
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+queueagentcount

If the number is greater then 0, then you move them into the queue, if 
not, whatever you want.

Finally, in terms of your other questions about logging the agents in. 
You could do the database way. You also could create a log in extension 
where you can take their cell number ( caller id) and use the 
application AddQueueMember(queuename[|interface][|penalty])

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AddQueueMember

So you should be able to do something like
AddQueueMember(queueName|ZAP/${CALLID(num)})

Anyway hope that helps.

Kevin




Jeremy Mann wrote:
>
> I’m curious if anyone has implemented the following:
>
> Need to setup an on-call queue, that activates after 5PM and 
> de-activates at 8AM, also that activates/deactivates on demand(I’m 
> thinking a feature code here). The “agents” need to log in via cell 
> phones, and when calls come in from outside to the asterisk system, 
> it’ll need to call the cell phone agents that are active.
>
> I’m thinking that it’s a simple SQL query, to update the agents status 
> and number, and that asterisk will do a lookup and append that to the 
> ZAP channel to dial, but interested in any logic someone might be able 
> to come up with for the dialplan.
>
>
> 
> This e-mail, facsimile, or letter and any files or attachments 
> transmitted with it contains information that is confidential and 
> privileged. This information is intended only for the use of the 
> individual(s) and entity(ies) to whom it is addressed. If you are the 
> intended recipient, further disclosures are prohibited without proper 
> authorization. If you are not the intended recipient, any disclosure, 
> copying, printing, or use of this information is strictly prohibited 
> and possibly a violation of federal or state law and regulations. If 
> you have received this information in error, please notify Texas 
> Health Management Group immediately at 1-817-310-4999. Texas Health 
> Management Group, its subsidiaries, and affiliates hereby claim all 
> applicable privileges related to this information.
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* , and is
> believed to be clean.
> 
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 
>
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Queue Question

2007-09-20 Thread Jeremy Mann
I'm curious if anyone has implemented the following:

Need to setup an on-call queue, that activates after 5PM and de-activates at 
8AM, also that activates/deactivates on demand(I'm thinking a feature code 
here).  The "agents" need to log in via cell phones, and when calls come in 
from outside to the asterisk system, it'll need to call the cell phone agents 
that are active.

I'm thinking that it's a simple SQL query, to update the agents status and 
number, and that asterisk will do a lookup and append that to the ZAP channel 
to dial, but interested in any logic someone might be able to come up with for 
the dialplan.




This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Queue question

2005-05-16 Thread jason walker



 
 
I have a queue set 
up where SIP extensions and ZAP channels dump into. On the SIP calls, I can 
force the caller ID since I know the calling party and the called party. The SIP 
calls do not necessarily go through the queue, but are directly dialed between 
users.
 
The ZAP is a little 
more difficult since within the queue I do not know which SIP user is going to 
receive the call from the group.
 
The main thing I 
need to figure out is how to grab the IP address of the SIP user that receives 
the call so I can send caller ID via a tcp/ip call, basically a screen pop. 

 
Between SIP users I 
can do a DBget from the SIP portion of the database. But how do I determine, via 
the dialplan, where the call ends up? Also, once I submit a call to the queue 
from the dialplan, I seem to loose control of the call within the dialplan. Is 
this by design?
 
Thank you in 
advance.
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.11 - Release Date: 5/16/2005
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Queue Question

2005-02-26 Thread Anton Krall
Found it! Interesting.. Local helps on not having to make 2 contexts, one
with all extension and their dialouts and privs and one with just the local
extensions and their voicemails without privs so callers wont be able to
abuse... Am I right on this? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Sábado, 26 de Febrero de 2005 05:38 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Queue Question

On Sat, 26 Feb 2005, Anton Krall wrote:

> I have a quick question.. reading the wiki, I found this:
>  
> "Use the 'Local' channel construct to point to an appropriate dial-out 
> extension in the dialplan if you'd like to add remote agents using 
> AgentCallbackLogin()"
> 
> That's exactly what Im trying to do, so fasr I needed to make a new 
> context that only has Dial cmds so that the call would not get routed 
> to the voicemails... But, what is this local channel construct?

Use google: "asterisk local channel".

Peter


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Queue Question

2005-02-26 Thread Peter Svensson
On Sat, 26 Feb 2005, Anton Krall wrote:

> I have a quick question.. reading the wiki, I found this:
>  
> "Use the 'Local' channel construct to point to an appropriate dial-out
> extension in the dialplan if you'd like to add remote agents using
> AgentCallbackLogin()"
> 
> That's exactly what Im trying to do, so fasr I needed to make a new context
> that only has Dial cmds so that the call would not get routed to the
> voicemails... But, what is this local channel construct?

Use google: "asterisk local channel".

Peter


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Queue Question

2005-02-26 Thread Anton Krall
I have a quick question.. reading the wiki, I found this:
 
"Use the 'Local' channel construct to point to an appropriate dial-out
extension in the dialplan if you'd like to add remote agents using
AgentCallbackLogin()"

That's exactly what Im trying to do, so fasr I needed to make a new context
that only has Dial cmds so that the call would not get routed to the
voicemails... But, what is this local channel construct?
 
__
Anton Krall

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Queue Question

2005-02-21 Thread Peter Svensson
On Mon, 21 Feb 2005, Shaun Tierney wrote:

> Is there a way to prioritize calls in multiple queues based on hold time?  I
> have three queues set up on my Asterisk PBX with agents logged into all
> three queues.  I've noticed that sometimes calls in one queue will make it
> through in a couple minutes while another queue will be backed up with
> people having been on hold for 30+ minutes.  Is it possibly the fact that I
> am set for the rrmemory ring strategy?

We have solved the same problem by creating a single queue that all the 
calls are dumped in. Callers and agents are matched through various 
criteras. This treats all callers alike - the longers caller in the queue 
gets first dibs on any agents that become available if there is a match. 

Our queues are based on the "icd" queue system for Asterisk. I expect
we will submit the changes to the icd system to the icd team once we have 
cleaned up the implementation a bit. 

Peter


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Queue Question

2005-02-21 Thread Shaun Tierney
Is there a way to prioritize calls in multiple queues based on hold time?  I
have three queues set up on my Asterisk PBX with agents logged into all
three queues.  I've noticed that sometimes calls in one queue will make it
through in a couple minutes while another queue will be backed up with
people having been on hold for 30+ minutes.  Is it possibly the fact that I
am set for the rrmemory ring strategy?

Thanks,

Shaun

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] Queue question

2004-10-28 Thread Robert Jackson


> -Original Message-
> From: Jeremy Rusnak [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, October 28, 2004 2:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Queue question
> 
> Our agents are played around a five second message when they 
> pick up a call, announcing the queue it is from.  Is there a 
> way to play a sound to the remote caller during this period?
> 

Have you tried the ringing application?  I am not sure if it 
will work, but it could be worth a try.  In the context that 
the agents are called in put something like:

exten => _1XXX,1,Ringing
exten => _1XXX,2,Dial(SIP/${EXTEN},15,t)

Or something like it.  The only problem that I can see with that 
is if your agent doesn't answer.  The caller will hear ringing, 
but it will just go back to music on hold.

Just an idea,

Robert Jackson
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Queue question

2004-10-28 Thread Jeremy Rusnak
Hi all,

Right now our queue system is working quite well.  One thing that I
would like to have the option of doing would be to play a "ring" for
the customer when they are connected to an agent.   Right now they go
from music on hold to an agent with no indication that their call is
"going through."

Our agents are played around a five second message when they pick up a
call, announcing the queue it is from.  Is there a way to play a sound
to the remote caller during this period?

I'd love for the customer to hear a standard "ring" before they are
actually connected to the agent.  Any pointers?

Thanks,
Jeremy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users