Displaying 20 results from an estimated 4000 matches similar to: "Queues - how to add back a agent without all other calls to agents stoping and re-starting"
2018 Nov 29
2
Queues and penalties
Hi John
This works fine providing extensions 1001,1002 and 1003 are "Incall" or
"Paused" - the problem appears to be that is a handset say 1002 is "ringing"
then the 2xxx then the penalty is not honoured.
This is well described in the History section of the following link
https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
As I say this seems to
2018 Nov 28
2
Queues and penalties
Hi All
I have been looking at this problem for a few days/weeks now and after some
advice please.
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
The problem I have is the customer want a simple call distribution like this
Extn 1001, 1002, 1003 to be called on an incoming call - if they
2020 May 01
1
Length of dial string
Hi Dovid
Yes was one of the options but as the required list is dynamic becomes very
messy - and all combinations problem - where as "call all workers job xxx"
is what is needed so the ability to call 20+ numbers is what is needed - agi
does a database search for all jobx workers and constructs a dialstring with
SIP, DAHDI and Local devices.
Can someone tell me where to find maximum
2020 May 01
0
Length of dial string
Paddy,
Why not use local extensions? You can do something like this.
Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all
&Local/set3 at call_all)
[call_all]
Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105
Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111
Exten =>
2020 May 01
4
Length of dial string
Hi all
as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see
* [ASTERISK-27946
<BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] -
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't
I have been fighting with this issue for months trying to find a solution I
2020 May 04
0
Length of dial string
Hi Paddy!
This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled.
From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate
2020 Apr 02
1
PJSIP Lockup
Paddy, It's pretty easy to spot from the CLI.
A voicemail gets called. And the screen basically stops scrolling from
there. Eventually you'll get the "Task processors exceeded 500 queued
tasks" or something like that. And maybe channels attempting to hangup due
to lack of RTP (If you have no-rtp timers configured).
Once you find the problem mailbox, You can call it via any
2010 Aug 19
3
Calling Line Identity - any ideas
Hi list
I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.
I have a user that wishes to have a "multi phone" divert. By that I mean
"calls made to his extension say Ext200 can be redirected to a different
extension say Ext400 and also to his home landline.
Doing the dial is fine using
2014 Dec 16
1
Realtime not storing voicemail password changes
Hi All
I am trying to get voicemail switched over to ARA on version 13 and notice
that the password is not stored in the db when it is changed.
A little hair pulling and playing around and I think the problem is in the
function ast_update2_realtime in main/config.c.
Issued source is ==>
int ast_update2_realtime(const char *family, ...)
{
RAII_VAR(struct ast_variable *,
2015 Mar 13
0
ringing in queues
On 13 March 2015 at 14:04, Matt Hamilton <efes9999 at hotmail.com> wrote:
> We use the ringall strategy for a small queue with 4 members. When a call
> comes in, if one of the members is busy, all the phones except the busy
> phone rings (as intended). While the other phones are ringing, if this busy
> phone becomes available again, we would like to have it start ringing.
>
2006 Mar 01
1
Agents, queues and Pentalties
List,
I've got 2 queues with 10 agents in both queues. One of the agents is mainly responsible for queue_1, and the others mainly for queue_2 so i've
defined the following in my queues.conf
[queue_1]
strategy=ringall
member=>Agent/1,2
member=>Agent/2,1
member=>Agent/3,1
member=>Agent/4,1
[queue_2]
strategy=ringall
member=>Agent/1,1
member=>Agent/2,2
2015 Mar 13
2
ringing in queues
We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle.
Is this possible? I played with ringinuse (queues.conf) and callcounter
2011 Mar 14
2
Asterisk -rx command not returning data - Version 1.4.33.1
Hi List
I am having trouble running the command
siptest:~# asterisk -rx 'dialplan reload'
most times it does what I expect and I get a response as below
siptest:~# asterisk -rx 'dialplan reload'
Dialplan reloaded.
every now and then I get no response i.e.
siptest:~# asterisk -rx 'dialplan reload'
siptest:~#
and a "verbose 10" setting shows
[Mar
2005 Jul 20
1
Agent Penalty
Can anyone shed any light on an issue with agent penalties?
I have 2 queues set up with agents working both queues, but where agent
1 should have a penalty for queue 2 and agent 2 should have a penalty
for queue 1. When a call is sent to either queue, it rings agents with
and without penalties at the same time.
I set up a second system and cannot replicate the issue on the test
system. I
2004 Dec 09
0
Balanced call distribution to agents logged into multiple queues.
Here's the scenario, 5clients, each client has their own queue. There
are 3 agents, and they're all logged into all of the queues. Using
round robin or rrmemory an agent can get a call in queue 1, then hang up
and immediatly get a call for queue 2,3,4,or 5 while the other 2 agents
have not gotten a single call. How does one make it so that the round
robin feature round robins to
2007 May 23
0
Realtime Queues and Agents
I am trying to configure a new server for use in a small Call Center.
I want to use realtime queues and agents and after following the
instructions I can get the queue to show up on the system but no agents.
I am using Asterisk 1.4.4 on a CentOS 5 machine.
I have this in extconfig.conf:
queues => mysql,asteriskcdrdb,queue_table
queue_members => mysql,asteriskcdrdb,queue_member_table
I put
2010 Jan 12
2
is roundrobin and rrmemory the same meaning?
Dear all,
I can't understand the diff between roundrobin and rrmemory strategy.
Could you explain for me ?
and is roundrobin means each available interface ring once or several
times and ring another?
; A strategy may be specified. Valid strategies include:
;
; ringall - ring all available channels until one answers (default)
; roundrobin - take turns ringing each available interface
;
2005 Mar 16
0
Agent groups broken in queues? (do not follow strategy)
I attempted setting up a queue with agents that log in, and
get called with incoming calls:
Agents log in using:
exten => *88,1,AgentCallbackLogin(${CALLERIDNUM}|${CALLERIDNUM}@test-sip)
Calls get into the queue with:
exten => 6029995654,1,Queue(test-noc|t|||60)
queues.conf:
[test-noc]
strategy = rrmemory
context = test-sip
timeout = 10
retry = 4
member => Agent/@2
2016 Nov 30
2
app_queue ringall - 2 agents answer same time problem
hi,
our customer reports problem when 2 agents answer the call in the same time
faster operator (device) answer the call, but the second is showed up
(on device) and call is without sound
asterisk 13.9/app_queue with strategy ringall/operators via Local
channel with sip device (chan_sip)
do you have any tips/info before i will dig deep into logs/debug?
checked google&issues.asterisk.org
2015 Jan 03
2
Asterisk removes a charachter from sip peer name
Hello all,
Just wondering on a behavior I noticed while testing with realtime sip
peers with names like 111.222 at mydomain.com. Using Kamailio as outbound
proxy, it sends Asterisk a sip message where To header value is <
sip:111.222 at mydomain.com> and From header has value "username" <
sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends
out the