Displaying 20 results from an estimated 2000 matches similar to: "Queue still tries to ring agent when busy"
2007 Apr 26
1
How does Realtime read config files?
Hi...
I just had a real quick and simple question... I have a asterisk
implementation setup w/ real time off of a mySQL database for SIP peers and
queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there
are some new configuration features i would like to use. I was wondering if
i could just add to the database table a column for the new config option?
if this will work or
2013 May 01
1
Call "stuck" in queue
Asterisk 11.1.0
One queue with strategy=leastrecent. (Full queues.conf below.)
Occasionally (several times today), a caller will get "stuck" in the
queue - there are operators available to take the call, but the caller
stays in the queue for a long time. Any idea what might cause this, or
where I can start looking to debug it? I'm going to start digging
through the queue log
2009 Sep 24
1
Asterisk 1.6 Transfer issue[Edited]
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123
2010 Jan 04
1
Some minor configuration issues with queues
Hello list !
I have some configuration issues with queues, but I'm sure they are
minor and for someone who has already configured queues it could be
trivial.
This is my queue configuration :
[VC_support_queue]
musicclass = default
strategy = ringall
timeout = 20
retry = 5
wrapuptime=15
autofill=yes
autopause=no
maxlen = 0
setinterfacevar=yes
announce-frequency = 0
2013 Apr 10
4
ACD problem
? Hi,
I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my
2013 Apr 18
5
Dynamic realtime + queues
Hi,
?
I am trying to store queues.conf to a MySQL database using dynamic realtime. I have a working ODBC connection and the queueing system already works but I want to store the queues.conf file to a database. I am following the guide from Asterisk the definitive guide, the ebook can be found at: http://ofps.oreilly.com/titles/9781449332426/asterisk-DB.html
?
I have a database called asterisk
2010 Jul 20
0
Got SIP response 603 decline, then the call hang up
Hi to all, I have a strange behavior in my asterisk server.
I have a queue for 5 agents, the calls enter the queue an go to the agents
normally, but if I need to transfer or dial directly to an agent extension
that is already in a call, the pbx hung up the actual call (not the
transferred call).
This is what I see in the log.
Called 103
-- Agent/103 is ringing
--
2008 Mar 17
1
update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
Hi,
I am using asterisk-1.4.15, My sip configs is like
[2501]
type=friend
username=2501
secret=2501
canreinvite=no
host=dynamic
dtmfmode=rfc2833
context = sip
disallow=all
allow=ulaw
incominglimit=1
nat=1
queue.conf is like
[gen-enq]
joinempty = yes
musiconhold = default
strategy = rrmemory
servicelevel = 60
timeout = 60
retry = 5
wrapuptime=5
announce-frequency = 90
announce-holdtime = yes
2015 Jun 19
2
Calling multiple phones at once
Hello All!
I asked week a so ago about how to call multiple phones alltogether (home, office, cell)
Dial app looks simple, this is kind of what I have now:
---------------------
[globals]
IVAN_HOME_OFFICE=SIP/BF8
IVAN_OFFICE=SIP/CFC
IVAN_CELL=SIP/83 at callcentric
[internal]
exten => 101,1,Dial(${IVAN_HOME_OFFICE}&${IVAN_OFFICE}&${IVAN_CELL},60)
same => n,VoiceMail(101 at
2014 Feb 12
1
Realtime Call Queues : call members in certain order
Hello,
I'm using MySQL realtime Call Queues (table /queues/ and table
/queue_members/).
I would like to ring the members of the call queue in a certain order.
Therefore I use ring strategy /lineair /and I put the members into the
table /queue_members/ in the order in which they have to be rang.
So I have the queue :
| name | musicclass | announce | context | timeout |
2011 Feb 20
1
MEMBERINTERFACE and MEMBERNAME questions
Hi!
Did play around with queues and need some help. I thought that MEMBERINTERFACE and MEMBERNAME should be set to the ?device? the call was queued to not the device that called the queue, or do i miss something?
Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on 2011-01-31 13:38:23 UTC
0317998985 calls Kinna (0320209030)
Tomas Ekman (SIP/0317998972) receives the call but
2007 Aug 21
1
Call queue problem
Hi all,
We have an 8 agent support desk setup with 2 call queues running
Asterisk 1.4.5. Every so often agents will receive a call from the
queue that only rings once not allowing them time to answer. The call
doesn't seem to be dropped, just seems to go to voicemail. The agents
are also mentioning they do not receive the 30 second wrapuptime I have
specified in queues.conf. We're
2010 Dec 26
1
Asterisk 1.8 Realtime Queue not working
I have configured my mysql database by following this link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
The only difference is that I am using ODBC instead of MySQL with Realtime.
Within extensions.conf I have the following for my queue
exten => 9**2**1611,1,Answer
exten => 9**2**1611,2,Queue(irock.com,tT,,,300)
exten => *50,1,Answer
exten =>
2010 May 04
1
problem with ringinuse=no, queue members receive randomly two calls
Dear all
on a debian amd64 i've installed (from source) asterisk 1.4.30
On the system we have in average 50 concurrent calls in queue and 40
sip members.
I'm experiencing an apparently random problem:
sometimes some users receive 2 calls from asterisk, apparently
ignoring the ringinuse=no settings.
It appears on users that are members of many queues
As you can see from the log, the
2007 Jul 16
1
Cisco 7940 log on/off
Hi All,
Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?
My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could have different users sitting there. Is
there any concept of "logging on" in these environments?
Cheers,
Adrian
2008 Apr 24
1
Full queue issues
Hello everyone.
I got a little problem in here: I want to set up a queue so that if anything of these happens:
a) No agents logged in
b) All agents busy
Then the user gets diverted somewhere. I used this (for testing purposes only, of course):
exten => 7080,1,Answer()
exten => 7080,n,Queue(teste)
exten => 7080,n,Goto(${QUEUESTATUS})
exten => 7080,n(ERROR),NoOp(${QUEUESTATUS})
exten
2007 Jan 21
2
Backports to 1.2.14 of 1.4.0 app_queue features.
Nothing much to be said.. I backported ringinuse, autofill and the QueueLog
application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't
matter.
They have received minimal testing but appear to function correctly. As always
with these things, don't blame me if they connect your callers to a phonesex
line, etc.
http://bum.net/patches/
Cheers,
Gavin.
2008 Mar 19
0
Deadair in queues.
Hello,
Asterisk Server A makes an outbound call, and upon connect:
exten
=>1,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${connectto},,tT
)
(${connectto} most of the time happens to be 12345 at 66.xx.xx.66 or 54321 {IP
masqueraded ofcourse})
..transfers it to * Server B (i.e 66.xx.xx.66) via SIP.
(Background info, Server B registers on Server A as 1000, and Server A
2015 Jun 19
0
Calling multiple phones at once
Hi again!
Also, given my setup below, how do I send caller id to my cell?
SIP/83 at callcentric is my cell, when I get incoming call when someone dials into Asterisk - I just see public calcentric?s DID number.
I want to send a number of who CALLED IN into the Asterisk and possibly add couple numbers upfront or something like this to signal me that this call comes through the PBX and not
2016 Nov 15
2
iaxmodem errors.