Join a queue with Aastra phones?

KodaK
Posts: 1885
Member Since:
2006-06-14

I've been searching all over for this, it has to be a simple addition to the Aastra dial plan:

I can't join a queue with Aastra phones. I dial 200* (my test queue is 200) and I just get a dial tone. If I dial the same thing from x-lite I get the queue login message from Asterisk like I'm supposed to.

I get the same results with both my 480i CT and my 9133i.

Any suggestions?

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



JamesDW
Posts: 518
Member Since:
2006-06-01
Re: Join a queue with Aastra phones?

Do you mean log on to a queue? I believe *11 is log in and *12 is log out.



KodaK
Posts: 1885
Member Since:
2006-06-14
Re: Join a queue with Aastra phones?

No.

I think my terminology may be off. I'm trying to dynamicaly add agents to the queue.

When you create a queue in the queue manager you log into that queue as an agent by dialing * and leave it by dialing **

For example, I created the queue 200.

If I dial 200* from x-lite, I can join the queue.

If I dial 200** from x-lite, I can leave the queue.

If I dial either from any of my Aastra phones it just gives me a dial tone again, and nothing shows up in the asterisk console (leading me to believe it's a problem with the dial plan on the phone -- it's never hitting asterisk.)

For shits and giggles I tried *11 and *12, they just give errors

*11:

-- Executing Macro("SIP/6696-098d29c8", "user-logon|") in new stack
-- Executing Set("SIP/6696-098d29c8", "DEVICETYPE=fixed") in new stack
-- Executing GotoIf("SIP/6696-098d29c8", "1?s-FIXED|1") in new stack
-- Goto (macro-user-logon,s-FIXED,1)
-- Executing NoOp("SIP/6696-098d29c8", "Device is FIXED and cannot be logged into") in new stack
-- Executing Playback("SIP/6696-098d29c8", "ha/phone") in new stack
-- Playing 'ha/phone' (language 'en')
-- Executing SayDigits("SIP/6696-098d29c8", "6696") in new stack
-- Playing 'digits/6' (language 'en')
-- Playing 'digits/6' (language 'en')
-- Playing 'digits/9' (language 'en')
-- Playing 'digits/6' (language 'en')
-- Executing Playback("SIP/6696-098d29c8", "is-curntly-unavail") in new stack
-- Playing 'is-curntly-unavail' (language 'en')
-- Executing Playback("SIP/6696-098d29c8", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing Hangup("SIP/6696-098d29c8", "") in new stack
== Spawn extension (macro-user-logon, s-FIXED, 6) exited non-zero on 'SIP/6696-098d29c8' in macro 'user-logon'
== Spawn extension (macro-user-logon, s-FIXED, 6) exited non-zero on 'SIP/6696-098d29c8'

So it tries to log into something, but I don't know what.

For comparison, here's what happens when I dial 200* from x-lite:

-- Executing Macro("SIP/6693-098d29c8", "agent-add|200|") in new stack
-- Executing Wait("SIP/6693-098d29c8", "1") in new stack
-- Executing Macro("SIP/6693-098d29c8", "user-callerid") in new stack
-- Executing GotoIf("SIP/6693-098d29c8", "0?report") in new stack
-- Executing GotoIf("SIP/6693-098d29c8", "0?start") in new stack
-- Executing Set("SIP/6693-098d29c8", "REALCALLERIDNUM=6693") in new stack
-- Executing NoOp("SIP/6693-098d29c8", "REALCALLERIDNUM is 6693") in new stack
-- Executing Set("SIP/6693-098d29c8", "AMPUSER=6693") in new stack
-- Executing Set("SIP/6693-098d29c8", "AMPUSERCIDNAME=X-lite") in new stack
-- Executing GotoIf("SIP/6693-098d29c8", "0?report") in new stack
-- Executing Set("SIP/6693-098d29c8", "CALLERID(all)=X-lite ") in new stack
-- Executing NoOp("SIP/6693-098d29c8", "Using CallerID "X-lite" ") in new stack
-- Executing Read("SIP/6693-098d29c8", "CALLBACKNUM|agent-user") in new stack
-- Playing 'agent-user' (language 'en')
-- User entered '6693'
-- Executing GotoIf("SIP/6693-098d29c8", "0?5:7)") in new stack
-- Goto (macro-agent-add,s,7)
-- Executing GotoIf("SIP/6693-098d29c8", "1?9:8)") in new stack
-- Goto (macro-agent-add,s,9)
-- Executing AddQueueMember("SIP/6693-098d29c8", "200|Local/6693@from-internal/n") in new stack
-- Executing UserEvent("SIP/6693-098d29c8", "Agentlogin|Agent: 6693") in new stack
-- Executing Wait("SIP/6693-098d29c8", "1") in new stack
-- Executing Playback("SIP/6693-098d29c8", "agent-loginok") in new stack
-- Playing 'agent-loginok' (language 'en')
-- Executing Hangup("SIP/6693-098d29c8", "") in new stack
== Spawn extension (macro-agent-add, s, 13) exited non-zero on 'SIP/6693-098d29c8' in macro 'agent-add'
== Spawn extension (macro-agent-add, s, 13) exited non-zero on 'SIP/6693-098d29c8'

and "show queues" lists:

asterisk1*CLI> show queues
200 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s
Members:
Local/6693@from-internal/n (dynamic) (Unknown) has taken no calls yet
No Callers

But I can't do this with the aastra phones. Nothing happens in the console when I dial 200* from them, and they just give me a dial tone back.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



KodaK
Posts: 1885
Member Since:
2006-06-14
Re: Join a queue with Aastra phones?

I've found a work around, but it's not ideal. Please read that as: I'd still like help with this.

Since ext-queues-custom is included in ext-queues and ext-queues does not exist, I simply created it, and made the login and logout commands similar, but without a * character:

in extensions_custom.conf:

[ext-queues-custom]
exten => 2001,1,Macro(agent-add,200,)
exten => 2002,1,Macro(agent-del,200,200)

This works, but it has significant drawbacks:

1) if I change, add or delete any queues I have to edit this file manually. Not exactly difficult, but a PITA, especially considering how scatterbrained I am.

2) I can't have, in this case, a 2001 or 2002 extension. That's ok, because my plans don't include that, but it's still a drawback. If my bosses want a queue number starting with 66 I'm SOL.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



KodaK
Posts: 1885
Member Since:
2006-06-14
Re: Join a queue with Aastra phones?

From Aastra support:

Try this

x+#|xxx*#

---------------

It works. You can't dial by IP, but who cares?

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



TROFFASKY
Posts: 78
Member Since:
2006-06-03
Re: Join a queue with Aastra phones?

When I set my system up about 5 months ago, I messed around with dynamic agents, but in the end I just made everyone a static agent, and they go on DND when they're not taking calls. The main reason for this is that we have more queues than agents, so dialling all those numbers to log in and log out is a real pain.



KodaK
Posts: 1885
Member Since:
2006-06-14
Re: Join a queue with Aastra phones?

Yeah, I'm probably going to do the same thing for different reasons. My users apparently have fingers made of the most delicate and precious flowers known to man and dialing all those digits is just too much for them.

I fear they may faint.

Anyway, yeah, static agents. :/

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



bdis
Posts: 111
Member Since:
2007-01-04
From Aastra support: Try

From Aastra support:

Try this

x+#|xxx*#

---------------

It works. You can't dial by IP, but who cares?

Any clue how to place this in the main config file on the ftp server?

I don;t want to hand edit 148 phones!!



KodaK
Posts: 1885
Member Since:
2006-06-14
If you are already using

If you are already using tftp all you need to do is edit the sip dial plan directive in aastra.cfg.

sip dial plan: "x+#|xxx*#"

If you are not using tftp to configure your phones, you should start. There's plenty of information on this site explaining how to set that up, but if you have specific questions go ahead and ask in this thread.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



gcc
Posts: 75
Member Since:
2007-02-24
Can this queue login and

Can this queue login and logout be used to track employee schedualing? Is there any way to harvest the information and to when an employee logs in and logs out and to add up the hours to use it for HR purpose and for pay?

Thanks



bdis
Posts: 111
Member Since:
2007-01-04
It's possible. Heck

It's possible. Heck anything is possible if you know how to program it. Is there someone out their who has already written code to do what you ask? queuemetrics might. you will have to look and see. They weren't that great to start out with but they have come a long way.



CustomGT
Posts: 2
Member Since:
2008-12-29
gcc and

gcc and bdis

http://lemens-ts.com/node/2

Checkout the timeclock program I wrote.



gcc
Posts: 75
Member Since:
2007-02-24
Oh, Amazing. I am going to

Oh, Amazing. I am going to have a look at this. I would have to somehow add this to Aastra startup.php file or dynamic agent login and log off (agents.conf) to get this working as a payroll clock. I am amazed and thanks for pointing this out CustomGT. I am just checking my elastix mysql password which should be eLaStIx.2007 by default but it doesn't seem to work. If I reset it to something else would anything else be broken? I am afraid that if I change this password then I break other things. Any feedback?

Thanks,



CustomGT
Posts: 2
Member Since:
2008-12-29
I wrote it for Asterisk 1.2,

I wrote it for Asterisk 1.2, if your on 1.4 or 1.6 couple of the cmds are different like "Wait" I believe.
If you wanted this to work when the agents login and off, you would need to modify it where it just inserts a record when they logon and off without asking any of the questions.

Im not sure if agents logon/off are stored in a db somewhere or not....



donhwyo
Posts: 28
Member Since:
2008-04-12
GCC its not eLaStIx.2007its

GCC its not eLaStIx.2007 its eLaStIx.2oo7. Too close for me to lol

Don



mst
Posts: 313
Member Since:
2008-05-02
I think the link

I think the link http://lemens-ts.com/node/2 is broken

Does anyone has working link?

Thank You



mst
Posts: 313
Member Since:
2008-05-02
I have new link from Matt

I have new link from Matt http://sourceforge.net/projects/asteriskphptime/



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