Displaying 20 results from an estimated 3000 matches similar to: "Asterisk Queues problem"
2008 Aug 05
2
Queue Penalties not working properly
Hi,
I am using Asterisk 1.4.18. I am implementing Penalties for my agents.
What is happening: two agents configuired one agent with penalty 1 and
the other with penalty 2. All the calls must go first to Agent 1 and if
his line is busy then only then agent 2 will get the call. However my
queues are not behaving in this manner. I have impmemnted ringall
strategy. Now when first call comes it
2010 May 19
2
a2billing DID and Queues
Hi all,
I have configured asterisk and a2billing.for inbound i have also configured
did and its forwarded to sip extensions.
But i want to enable queues with inbound numbers(DID).But i could not find a
way to do this in a2billing.
I want enable that if some did comes to asterisk/a2billing it should be
forwarded to queues not sip extensions and
their i want to enable hunting so if one
2015 Jul 28
2
Queues don't follow dialplan if no members are registered
Hello,
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as
follows in extensions.conf:
exten => s,1,Queue(myqueue,rtnC,18)
same => n,Background(user_unavail)
same => n,WaitExten(10)
exten => 1,1,Voicemail(1111 at my-vm,s)
This rings the phones in the queue for 18 seconds. If no queue members answer,
the caller is then prompted to press 1 and leave a
2015 Jul 29
2
Queues don't follow dialplan if no members are registered
----- Original Message -----
> From: "John Kiniston" <johnkiniston at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Tuesday, July 28, 2015 12:12:05 PM
> Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered
>
> In your queues.conf do you
2008 Oct 10
9
How to enable inbound CLI for X-Lite/Asterisk phone.
Hi,
I am using asterisk 1.4.18. I am using it for inbound only call center.
The SIP phones are X-Lite. Right now when a call is proxied by Asterisk
to X-Lite the agent only sees asterisk written on its CLI screen. I want
the agents to be able to view the callees number. Is there any way to
make this happen.
Regards
Syed Nasruddin
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An HTML
2009 May 07
1
Macro arguments on app_queue
hi list, i have a question about the args of queue:
when we use Queue() app, there are some arguments than can use. help from
CLI:
Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]])
well.. i'm trying to identify who has taken the call on a queue, and, when
agent conected, launch a macro with some args based on who takes the call
i do:
exten =>
2007 Apr 09
4
incoming zaptel calls fail
Using the latest SVN of zaptel and asterisk, I can no longer receive
incoming analog calls. The caller just hears it ringing but Asterisk
doesn't pick up.
I am seeing these error messages:
[Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID
returned with error on channel 'Zap/2-1'
[Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID
2007 Aug 02
1
A simple IVR extension problem
Hi list,
I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS
5.
I am having trouble to make my simple IVR extension work, here is relevant
config:
zapata.conf
----
context=incoming
signalling=fxs_ks
channel => 4
context=internal
signalling=fxo_ks
channel => 1
-----
extensions.conf:
----
[office]
exten => s,1,Dial(Zap/1,30)
[home]
exten =>
2005 Aug 01
1
X100P/Caller ID: clidtest works, * complains [repost]
Hi,
I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
having problems with Caller ID. I have run clidtest, and it seems happy
enough, returning:-
server clidtest # ./clidtest /dev/zap/1
Number: 0412222222, Name: MOBILE
(that number's fake.) However, I'm not getting the caller ID passed
through with *. Sometimes I get a "failed checksum" like
2005 Jan 17
1
TDM400 answers the line all the time!
hi all,
We have a TDM400 card with 4 wfo modules. now the modules load fine
and when i start asterisk with on phone line connected it just starts
spewing these messages:
-- Starting simple switch on 'Zap/4-1'
Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
(Ring/Answered)...
Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
(Ring/Answered)...
2004 Sep 28
7
UK (British Telecom) Caller ID again
I've followed the recent thread on caller id with UK British Telecom
networks (where the caller id data is delivered before the first ring).
My understanding is that if I use a recent CVS head (e.g.
CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to
do is include the line:
usecallerid=uk
In my zapata.conf (in the [channels] section)
I've done this, but I get:
Sep
2018 Dec 04
2
asterisk is not seeing my queues in database
I enabled the logs on the mysql database and ran :
realtime load queues name cou0002-test
in the mysql log I can see that the proper select statement is being
executed:
2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION
ISOLATION LEVEL READ COMMITTED
2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE
name = ?
2018-12-04T16:29:27.254902Z 229
2004 Mar 03
3
Ringing Delay
Sorry if this is a daft question but when a PSTN call comes in on my
X100P the console shows the following;
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
2005 Sep 12
2
Callerid fails in any release after beta1 fails
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead
releases fail with a combination of checksum and ss_thread errors?
I'm concerned when beta2 or the 1.2 release comes out it will not work.
I have been through the configs I can't find and changes that need to be
made to get CVSHEAD to work.
Thanks
John Hill
2007 Jan 22
1
2 ring delay before asterisk answer
I am a little green when it comes to all this but I am trying to connect
our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able
to dial an extension on my PBX handset and I get a dialtone from the PBX.
After 2 rings I then hear the asterisk server connect and I get a dialtone
from asterisk. I am then able to dial an extension on another asterisk
server.
My question is: How do
2007 Apr 17
2
queues
Is there anyway to setup a queue with only one agent (device) which is
always logged in. So when a call hits that queue the device will ring (if
not already on a call) or will be put in the queue if the call is already in
place?
Thanks
Miles
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2006 Mar 21
2
TDM400 FXO module not answering or dialing out.
Hi all,
I have hit a wall configuring a TDM400, I have set these up before without
issue but today I just can't seem to figure out what I am doing wrong.
On an incoming call the following is produced in the Asterisk console with
verbose 4
-- Starting simple switch on 'Zap/2-1'
Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring
Begin)...
Mar 22
2004 Aug 28
5
Distinctive ring detection problem
I am trying to get distinctive ring to work on my PSTN with no luck.
I can get 2 different ring codes but it skips the context assigned...
here is my complete zapata.conf:
[channels]
signalling=fxs_ks
usecallerid=yes
rxgain=1.0
txgain=1.0
language=en
context=default
usedistinctiveringdetection=yes
dring1=134,0,0
dring2=137,0,0
dring1context=internal2
dring2context=default
2006 Jan 31
3
ZAP <--> sip(polycom301) can not hear each other
please help!!!
I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong?
thanks
sip.conf:
[general]
context=local-access ; Default context for incoming calls
bindport=5060
2008 Oct 10
3
Got event 17 (Polarity Reversal)...
Can anyone tell me what this message means?
Got event 17 (Polarity Reversal)...
I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0.
It appears that I get this Polarity Reversal each time an inbound call
hangs up. This results in another ring, but no one is there. It appears
as an unknown caller, but I believe its a phantom.
Thanks,
Jim
[Oct 10 12:47:54] NOTICE[6669]: