Re: [asterisk-users] queues

2007-04-17 Thread Octavio Ruiz (Ta^3)
Is there anyway to setup a queue with only one agent (device) which is
always logged in. So when a call hits that queue the device will ring (if
not already on a call) or will be put in the queue if the call is already
in place?

Sure, in queues.conf you can add many type of members (not just agents) like SIP
or Local channels. So you don't need to use AgentLogin/CallBackLogin

ej.

[recepcion]
musicclass = default
monitor-format = wav49
strategy = ringall
timeout = 15
retry = 2 
autopause = no 
maxlen = 3
context = voicemail
setinterfacevar = yes
announce-frequency = 15
periodic-announce-frequency = 0
announce-holdtime = yes
announce-round-seconds = 10
joinempty = strict
leavewhenempty = strict
eventwhencalled = yes
eventmemberstatus = yes
ringinuse = yes
timeoutrestart = no

member = SIP/9001,1
member = SIP/9005,2



-- 
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] Re: [asterisk-dev] Find the name of queue

2007-03-29 Thread Octavio Ruiz (Ta^3)
 i´m trying find in the codes of asterisk as change the name of file created
 after that a extension dial for a queue.
 Has someone some sugestion for obtain this name of queue(150)?

Use Queue monitor rather than Agent monitor.

agents.conf
recordagentcalls=no

queues.conf
monitor-type = MixMonitor
monitor-format = gsm

And define MONITOR_FILENAME in your dialplan

(just an example)

[macro-queues]
exten s,1,MONITOR_FILENAME=${ARG1}
exten s,n,Queue(${ARG1})

-- 
I like your SNOOPY POSTER!!
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] deprecated ALERT_INFO var andAMI's Originate command

2007-03-12 Thread Octavio Ruiz (Ta^3)
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this
via AMI:

Action: Originate
Channel: Sip/1234
Application: AgentLogin
Data: 1234
Variable:  _ALERT_INFO=info=alert-autoanswer
Callerid: AutoLogin[1234]

In order to send an autologin and autoanswer call to the agent 1234 on an
Aastra phone at extension 1234. (just for example).

Now in * 1.4 with ALERT_INFO deprecated I don't have that option, rather
than calling a Local/ channel to SetSIPheaders() and Dial(). I don't want
to do it in that fashion 'cos I like (and have) to have completely separated
dial plan logic (extensions.conf) and external applications via AMI.

Regards,

-- 
I'm having a MID-WEEK CRISIS!
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Call load balancing

2007-03-09 Thread Octavio Ruiz (Ta^3)
 I've got a system I'm putting together to handle IVR calls with *
 I have one head system that terminates two PRIs. It routes the calls from
 the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
 The * boxes run AGI scripts to process the IVR calls. Can I load balance the
 routing if I have five calls each of the IVR * boxes gets two call and the
 next call would go to the system that currently has the lowest number of
 calls?

Another approach: what about load-balance (in terms of redundancy and
scalability)  the AGI app's and just the AGIs with FastAGI? So your
IVR application can be separated from your * boxes and they (the * boxes)
dont have to ve overloaded with your AGI apps.

Your head system receive the two PRIs and in dial-plan logic you can (maybe
using RANDOM() or something more deterministic like a counter)

[just an example]:

exten s,1,Answer
exten s,n,Random(50:next)
exten s,n,AGI(agi://asterisk1/${VAR1}|${VAR2})
exten s,n,Hangup
exten s,n,AGI(agi://asterisk2/${VAR1}|${VAR2})
exten s,n,Hangup

-- 
Honi soit la vache qui rit.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Asterisk queue and agents

2007-03-07 Thread Octavio Ruiz (Ta^3)
Have a question for the group
If I have an agent is on the phone outside of the queue should that person
still get queue calls ?
Doing a show agents online I see Available however show hints I see inuse.

There is a ringinuse feature for SIP devices on 1.4.X which is what you are 
looking for.

-- 
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] API: how to bridge originated call?

2007-01-14 Thread Octavio Ruiz (Ta^3)
 in my case I want a user to be on-line all the time - the system will
 dial and connect them and, when they're done, they select the next one.
  what I'm doing now is putting them into a loop with a g-option on the
 dial.  the number it dials is set thru the api.  if the number's not
 set it waits one second and loops again.

Why not use an agent channel AgentLogin()?  Or I missunderstood your need?

-- 
Honi soit la vache qui rit.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] play music while continue executing dial plan

2007-01-14 Thread Octavio Ruiz (Ta^3)
Perhaps MusicOnHold() app?

  Is there any application can let the dial plan to execute while
 playing music?  Say I have a lot of command to do in the dial plan but
 I don't want to keep silence while execution of dial plan.  I notice
 Background(file) can play music but it will hold until pressing a key.
 I want something like background and it plays music with continuing
 execute the rest of the command in dial plan.

-- 
YOW!!!  I am having fun!!!
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] CLI output to file

2006-12-17 Thread Octavio Ruiz (Ta^3)
 Hi all,
  How can I redirect the CLI output to file without viewing it on
 screen?  Is it possible.

Read and edit /etc/asterisk/logger.conf

You should have already that output at /var/log/asterisk/messages.

-- 
Oh, wow!  Look at the moon!
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] CLI output to file

2006-12-17 Thread Octavio Ruiz (Ta^3)
 Thanks!
 But the information in /var/log/asterisk/messages is much different
 from the messages in CLI.  I want to log the message in CLI to file
 for easy debugging.

It is the same, see the levels in logger.conf, copy the console config
to the messages config:

(just for example)

console = notice,warning,error,verbose,dtmf
messages = notice,warning,error,verbose,dtmf

And you will have the same output. :)

 On 12/18/06, Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote:
  Hi all,
   How can I redirect the CLI output to file without viewing it on
  screen?  Is it possible.
 Read and edit /etc/asterisk/logger.conf
 You should have already that output at /var/log/asterisk/messages.

-- 
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] AgentCallbackLogin deprecated?

2006-11-28 Thread Octavio Ruiz (Ta^3)
  Yes, AgentCallbackLogin is deprecated, but it will not be removed
  until after 1.4. 
 
 Is there an isolated example somewhere of how to use existing dialplan
 logic and dynamic queue membership to simulate the current behaviour?

http://svn.digium.com/view/asterisk/trunk/doc/queues-with-callback-members.txt

 What about generation of statistics for callcentre monitoring? If this
 is not taking place through chan_agent, won't it be reinventing the
 wheel to have to simulate this behaviour, too?

Why? Seems that reinventing the well was the agentcallbacklogin
implementation, when it could be happend in dialplan logic.


-- 
  May I ask a question?
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] queue management

2006-11-16 Thread Octavio Ruiz (Ta^3)
 That's pretty easy and included in the basic * implementation - you tell  
 the queue not to accept users and play a message after the queue command  
 terminates.
 l.

Don't forget to analyze the QUEUESTATUS variable. :)

-- 
Sign my PETITION.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Can AGI do this?

2006-11-14 Thread Octavio Ruiz (Ta^3)
Bret Schuhmacher, who happens to be smarter than you, thinks:
 Please pardon the absolute noob questions.  Someone has asked me to 
 interface with Asterisk and have it dial 4 numbers in succession

I dont understand this: 

 to have 
 it track down an on-call person.

 If not, is it possible to write an AGI program that gets all 4 numbers, 
 then somehow hands them one-by-one to Asterisk?  If so, how does 
 Asterisk manage the communication of failed to complete the call with 
 the AGI app?  Does the AGI just monitor stdin looking for status 
 messages and returns the next number?
 
 If Asterisk/AGI can do both, is the first method better than the 
 second?  It certainly seems easier.

The AGI script can die when it finished they job, probably I'm
missunderstanding what you want but maybe this examples helps:

exten = 123,1,Answer
exten = 123,n,AGI(somescript)
exten = 123,n,Dial(${TRUNK}/${FOURDIGITNUMBER})
exten = 123,n,Hangup

somescript: just sends AGI command SETVAR FOURDIGITNUMBER 2468

[context]
exten = s,1,Answer
exten = s,n... (set timeouts, et all).
exten = s,n,Background(audio)
exten = s,n,WaitExten

exten = _,1,Dial(${EXTEN})

exten = t,..
exten = h,...

-- 
Onward through the fog.
___
--Bandwidth and Colocation provided by Easynews.com --

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