Displaying 20 results from an estimated 700 matches similar to: "Full queue issues"
2007 Apr 09
1
${QUEUESTATUS}
There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when
they would be set;
If someone could correct errors with these definitions ot would be
appreciated;
TIMEOUT - the max time specified in the queue command elapsed, only
checked between retries so may not be 100% accurate.
FULL - the number of callers
2006 Feb 09
4
Queue - check agent
I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's?
What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call.
Thank you for your time.
--
Tomislav Par?ina
Lama Computers Split
2007 Jan 31
3
Queue Status
Hello all,
I think Lee has given me a head start, but I'm still running in a circle
(at least i'm in the lead).
The problem is with my queues. The phones go to their own voicemail
after 5 rings.
That's about the same time I allow the phone to ring before trying
another phone in the queue. Is there a way to tell asterisk....?
If this call is coming from a queue, do not follow a
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
2007 Sep 28
1
Ringing Groups, SIP Forward and looping problem
I've a big problem with SIP forwarding back into 'ringing groups'
creating what can only be described as call storms :-(
I have a 'ringing groups' of SIP phones with an effective dialplan (much
simplified) like so:
; Purchase ledger
[ptsn_inbound]
exten => _846061,1,Dial(Local/6061 at groups)
....
[groups]
exten =>
2007 May 17
1
Cascading Queues
Hey Everyone,
Have a couple of questions here..
Scenario 1:
We are working with a client that currently has one support queue with
about 10 agents. They are starting to get pretty long hold times for
their customers and they have requested three queues. Queue 1 will have
a timeout of 4 minutes. After that it will move to Queue 2 which has a
default timeout of 3 minutes. After that we will
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 =>
2008 May 05
3
TDM410P driver?
Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver?
Att
Vin?cius Fontes
Desenvolvimento
Canall Tecnologia em Comunica??es Ltda.
2011 Mar 10
4
Asterisk queues : command to run when a call is being bridged
Hi,
Is there any way to run a command (AGI script, whatever else) at that moment
when the call that was in the queue is being bridged to a specific agent?
An examples of what I would want to do with this is, for example, have
Asterisk ask the caller for his 4 digit customer ID before being put in the
Queue. Once I know who the caller is being connected to (which agent) I'd
run a
2009 Feb 19
2
Not answering call when queue is full or has no members
Hello all,
I'm trying to prevent answering a channel when a queue is either full or has
no members. It seems I'm forced to answer a call before I call Queue() or
else the audio is in the early media (which is unacceptable because of the
short duration of early media on ISDN).
Is there any way to let Queue() automatically answer a channel if the call is
going to be placed in the
2007 Jul 30
1
Queues with logged in agents that are not reachable
Hello, I am using 1.4.8 and have a question about Queues.
I noticed that if I have an agent logged in using AgentCallBackLogin
and that agent is unreachable for some reason (SIP phone unplugged)
calls to him/her will completely yack.
For example:
1-Agent 500 is the only one logged into queue number 1.
2-A call comes into queue number 1
3-The call is pushed to agent 500 at extension 21 which is
2010 Feb 05
6
Still on spandsp/app_fax and T.38
This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :)
Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it
2010 Feb 05
8
Losing local SIP phones when internet goes down?
Hi,
I'm getting some strange behaviour on Asterisk 1.4 running on Debian
Stable (Lenny). I suspect it's something to do with my setup, rather than
a bug, but I'm struggling to see it, and would appreciate any input.
Setup: PC with two ethernet cards: eth0 goes to local network, including
two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes
to router and
2008 Sep 23
5
Extension registration
Hi all,
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host=192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
When trying to register from a softphone installed on a PC behind a nat with
IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what
could be the issue?
Regards
-------------- next part
2008 Apr 30
1
One way audio...
I have a big headache. I have an Asterisk server connected to an Avaya
PBX. Everything is working between those two. The problem is that I
have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the
Internet to the Asterisk server through a Fortinet firewall. When
calling from a PAP2T I get one way audio, the remote site can hear me
but I cannot hear them. If I do an "rtp
2008 Nov 11
3
OT: Polycom Firmware available (by accident?)
Not sure if Polycom is changing their policy or if this is an accident,
but you can actually download SIP 3.1.1 right from their web site.
Anyone looking for firmware should get it now before it disappears.
SIP app and release notes can be found here:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html
-Dave
2008 Dec 22
2
Manager API - standardization?
Hi all,
I know I'm probably stirring up a hornet's nest with this question/comment
but I've spent the last few days working on a PHP-based class for the
manager interface as we're preparing for a pretty big upgrade at our call
center and I'm revamping all of the management apps I've written. I can
connect to the manager interface and send query strings back and forth all
2008 May 20
7
Busy out a zap channel?
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that channel, I need to busy the line so calls
will come over the other lines.
--
?Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director
2007 May 05
2
Manager API Output
Hi,
Is there any way that I can store my manager API output that is:
My question is that is there any why using that I can get the QueueStatus
and store the result in some text file for further processing.
<?php
$strHost = "127.0.0.1";
$strUser = "cron";
$strSecret = "1234";
2005 Jul 15
1
Manager API commands QueueStatus and Queues
Hello,
I'm trying to write a php script that issues the QueueStatus and Queues
manager API commands to Asterisk and records the returned event data
into a MySQL table.
On voip-info.org, I see that QueueStatus returns such things as Queue,
Max, Calls, Holdtime, Completed, Abandoned, ServiceLevel, and
ServiceLevelPerf. However, there are no descriptions for these values.
I've tried