[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 [4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new stack
-- Executing [4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num 90015103) in
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1, SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk) in new stack
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue
position (which was 1)
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal, 4767209611, 4) exited non-zero on
'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two extensions/agents
agent = 301,301
agent = 302,302
--
[r...@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=xx
host=dynamic
context=phones
mailbox=...@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
ak...@abacus-it.nomailto:ak...@abacus-it.no
--
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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/301member=SIP/302instead 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: ak...@abacus-it.no
To: asterisk-users@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
[4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new
stack
-- Executing
[4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num
90015103) in new stack
Callerid num 90015103
-- Executing
[4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1,
SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is
ringing
-- Nobody picked up in
5000 ms
-- Executing
[4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk)
in new stack
-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1
--
SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told
SIP/odin.service.ipallover.net-00d1 in teknisk their queue position (which
was 1)
--
SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal,
4767209611, 4) exited non-zero on 'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two
extensions/agents
agent = 301,301
agent = 302,302
--
[r...@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=xx
host=dynamic
context=phones
mailbox=...@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
ak...@abacus-it.no
_
The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with
Hotmail.
http://www.windowslive.com/campaign/thenewbusy?tile=multicalendarocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_5--
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New to Asterisk? Join us for a live
Re: [asterisk-users] Call queues - issues, can't make it work.
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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: ak...@abacus-it.no
To: asterisk-users@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 [4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new stack
-- Executing [4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num 90015103) in
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1, SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk) in new stack
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue
position (which was 1)
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal, 4767209611, 4) exited non-zero on
'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two extensions/agents
agent = 301,301
agent = 302,302
--
[r...@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=xx
host=dynamic
context=phones
mailbox=...@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
ak...@abacus-it.nomailto:ak...@abacus-it.no
The New Busy think 9
