[asterisk-users] Call queues - issues, can't make it work.

[Aksel Celasun]

Thank You Tarek!

That was the case, and i saw now i had a typo in the extension further down, but, you solved it.
Now I faced a couple of other problems, alle the announcements and MOH didn’t play, the settings are default.
Maybe i'll figure it out.

Thank you


Regards 

Aksel


Fra: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] På vegne av Tarek Sawah
Sendt: 14. juni 2010 15:00
Til: Asterisk Users
Emne: Re: [asterisk-users] Call queues - issues, can't make it work.

when you add an agent to a queue the agent should log in
try adding
member=SIP/301
member=SIP/302
instead of agent directives.
this will ring both phones.. from your output it doesn't seem to be ringing the agents at all.

-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308


________________________________
From: aksel at abacus-it.no
To: asterisk-users at lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there


I have been struggling with queues, because i think this is the right module for our business.
My main goal, is when we receive external calls, the receptionist should be able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 seconds, and then transferred to queue “teknisk”
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings…and rings…until it hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [4767209611 at internal:1] NoOp("SIP/odin.service.ipallover.net-000000d1", "") in new stack
    -- Executing [4767209611 at internal:2] Verbose("SIP/odin.service.ipallover.net-000000d1", "Callerid num 90015103") in new stack
Callerid num 90015103
    -- Executing [4767209611 at internal:3] Dial("SIP/odin.service.ipallover.net-000000d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
    -- Called 301
    -- SIP/301-000000d2 is ringing
    -- Nobody picked up in 5000 ms
    -- Executing [4767209611 at internal:4] Queue("SIP/odin.service.ipallover.net-000000d1", "teknisk") in new stack
    -- Started music on hold, class 'default', on channel 'SIP/odin.service.ipallover.net-000000d1'
    -- Stopped music on hold on SIP/odin.service.ipallover.net-000000d1
    -- <SIP/odin.service.ipallover.net-000000d1> Playing 'queue-youarenext.gsm' (language 'en')
    -- Told SIP/odin.service.ipallover.net-000000d1 in teknisk their queue position (which was 1)
    -- <SIP/odin.service.ipallover.net-000000d1> Playing 'queue-thankyou.gsm' (language 'en')
    -- Started music on hold, class 'default', on channel 'SIP/odin.service.ipallover.net-000000d1'
    -- Stopped music on hold on SIP/odin.service.ipallover.net-000000d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 'SIP/odin.service.ipallover.net-000000d1'

asterisk*CLI>

-----------------------------------------------------------------------------------------------------------------------
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


----------------------------------------------------------------------------------------------------------------------
[root at asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-----------------------------------------------------------------------------------------------------------------
Sip.conf
[301]
type=friend
secret=xxxxxxxxxx
host=dynamic
context=phones
mailbox=301 at default
qualify=yes
callgroup=teknisk
---------------------------------------------------------------------------------------------------------------------
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);           ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
aksel at abacus-it.no<mailto:aksel at abacus-it.no>


________________________________
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