In this book example there is a printing issue at Unpaused section. it should
be like following
same => n,GotoIf($[${UPQMSTATUS} = UNPAUSED]?agent_unpaused,1:agent_not_found,1)
From: [email protected]
To: [email protected]
Date: Fri, 27 May 2011 18:41:18 +0000
Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
Oh! wait i got following error when i trying to Unpause my queue. do you have
any idea ?
holler*CLI>
== Using SIP RTP CoS mark 5
-- Executing [*99@from-sip:1] Verbose("SIP/7102-0000000e", "2,UnPausing
member in all queues") in new stack
== UnPausing member in all queues
-- Executing [*99@from-sip:2] Gosub("SIP/7102-0000000e",
"subSetupAvailableQueues,start,1()") in new stack
-- Executing [start@subSetupAvailableQueues:1] Verbose("SIP/7102-0000000e",
"2,Checking for available queues") in new stack
== Checking for available queues
-- Executing [start@subSetupAvailableQueues:2] Set("SIP/7102-0000000e",
"MemberChannel=7102") in new stack
-- Executing [start@subSetupAvailableQueues:3] Set("SIP/7102-0000000e",
"MemberChanType=SIP") in new stack
-- Executing [start@subSetupAvailableQueues:4] Set("SIP/7102-0000000e",
"AvailableQueues=booktech1^booktech2") in new stack
-- Executing [start@subSetupAvailableQueues:5] GotoIf("SIP/7102-0000000e",
"0?no_queues_available,1") in new stack
-- Executing [start@subSetupAvailableQueues:6] Return("SIP/7102-0000000e",
"") in new stack
-- Executing [*99@from-sip:3] UnpauseQueueMember("SIP/7102-0000000e",
",SIP/7102") in new stack
[May 27 11:40:19] WARNING[2358]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():
syntax error: syntax error, unexpected '=', expecting $end; Input:
= PAUSED
^
[May 27 11:40:19] WARNING[2358]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [*99@from-sip:4] GotoIf("SIP/7102-0000000e",
"?agent_unpaused,1:agent_not_found,1") in new stack
-- Goto (from-sip,agent_not_found,1)
-- Executing [agent_not_found@from-sip:1] Verbose("SIP/7102-0000000e",
"2,Agent was not found") in new stack
== Agent was not found
-- Executing [agent_not_found@from-sip:2] Playback("SIP/7102-0000000e",
"silence/1&cannot-complete-as-dialed") in new stack
-- <SIP/7102-0000000e> Playing 'silence/1.ulaw' (language 'en')
-- <SIP/7102-0000000e> Playing 'cannot-complete-as-dialed.ulaw' (language
'en')
-- Auto fallthrough, channel 'SIP/7102-0000000e' status is 'UNKNOWN'
From: [email protected]
To: [email protected]
Date: Fri, 27 May 2011 18:03:02 +0000
Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
This is working great! Thanks a lot paul.
One more question before we have Agent/XXXX configured in queueMetrics so i
need to change them in queueMetrics with SIP/XXXX right ?
> Date: Fri, 27 May 2011 10:18:39 +0100
> From: [email protected]
> To: [email protected]
> Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
>
> On 26/05/11 23:18, Satish Patel wrote:
> > Thanks,
> >
> > I went through this example before. I was confuse and wondering how
> > should I add third queue in this picture?
> >
>
> From the example:
>
> *CLI> database put queue_agent 0000FFFF0001/available_queues support^sales
>
> "support^sales" is a list of queues. Put as many in the list as you
> need. E.G. sales^support^tech
>
> cheers,
> Paul.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users