asterisk users - Sep 2008 - AgentCallbackLogin AddQueueMember

Hi 

i have problem with AddQueueMember logic. 

I need login Agent(Member) in asterisk. 
use this option: 
for example: 
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)

and now i want to call to this Agent: 

exten => _1XX,1,Dial(Agent/${EXTEN:1})

call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. 

This doesn't work, How can i do this on Asterisk 1.4(not using
AgentCallbackLogin).

Chris

On Tue, Sep 2, 2008 at 6:07 PM, Krzysztof Zimnicki <krzysiez at go2.pl>
wrote:
> Hi
>
> i have problem with AddQueueMember logic.
>
> I need login Agent(Member) in asterisk.
> use this option:
> for example:
> AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
>
> and now i want to call to this Agent:
>
> exten => _1XX,1,Dial(Agent/${EXTEN:1})
>
> call to 113 and asterisk should call to Agent => 13 on interface
SIP/ekiga.
>
> This doesn't work, How can i do this on Asterisk 1.4(not using
AgentCallbackLogin).
>
You can't dial Agent/ as it's not channel anymore (not sure that it
was even possible in 1.2), but just queue member name (displayed in
"queue show"). If you use Local channels for dialing members, you can
enable setinterfacevar=yes in queues.conf and then get MEMBERINTERFACE
variable to dial actual SIP device. The rest of mapping is up to you,
keep it in asterisk db or SQL.

However I must warn you that there is really not much use of Local
channels in queue for 1.4 as there's no state_interface. Queue will
always treat your members as available, and limiting calls with
GROUP_COUNT as in sample docs will create 1 channel for each queue
call per each member, just to find out that it's currently busy.

I would suggest to jump directly to 1.6 if it's new setup, you should
test stability anyway.. otherwise you may try out backport of
state_interface to 1.4 -
http://ftp.iq-labs.net/state_interface-1.4/asterisk-svn-1.4.19-state_interface_101578.patch

Btw, there is currently one very rare deadlock problem in
state_interface, however that shouldn't keep you away, as we have 2000
calls per day and we've seen it only once for half year. I hope it
will be fixed soon.. (putnopvut?)

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

> I need login Agent(Member) in asterisk.
> use this option:
> for example:
> AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
Just out of curiosity, where do you get this AddQueueMember syntax from?
http://www.voip-info.org/wiki/view/Asterisk+cmd+AddQueueMember

Description: 
  AddQueueMember(queuename[|interface][|penalty]):

> Just out of curiosity, where do you get this AddQueueMember syntax from?
Here:
http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
page: 367

> > Just out of curiosity, where do you get this AddQueueMember syntax
from?
> 
> Here:
>
http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.c
om
> /books/9780596510480.pdf
> page: 367
Oh so the VOIP Wiki is out of date!
Now, where should we go to for reliable Asterisk info then?