Hi,
I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's
running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and
copied all config files from original to the new server. But when a caller lands inside
the queue no queue message
:[EMAIL PROTECTED] On Behalf Of Rajkumar S
Sent: Monday, 20 February 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Queue Messages now playing when caller is
insidequeue
Hi,
I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h
Peter Fern wrote:
In queues.conf:
[queuename]
announce-frequency = XX ; where XX = number of seconds
I had already given it. From my orig mail:
[callcenter]
music=default
leavewhenempty = yes
monitor-format = wav
strategy=rrmemory
timeout=15
retry=5
servicelevel = 60
wrapuptime=5
On Tue, Mar 4, 2008 at 7:18 PM, marek cervenka [EMAIL PROTECTED] wrote:
can you recommend cleansimplestable solution for incoming call popup
(in browser)?
ADM http://adm.hamnett.org/ can invoke browsers when a call arrives.
raj
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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
On Mon, Mar 17, 2008 at 6:30 PM, Grygoriy Dobrovolskyy
[EMAIL PROTECTED] wrote:
Forgot to add:
Multiple queues fo sip phone, it is normal that sometimes it is ringed, as
reported busy for 1 queue and free for another. you limitited incoming call
to max 1 ' incominglimit=1' so ;)
My
Thanks Atis,
On Tue, Mar 18, 2008 at 3:50 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
As for current problem - i suspect that device state don't get updated
correctly for Queue application, so Queue tries to dial device, and
call-limit blocks it from doing so. There's a patch, currently in
Hi,
I am using asterisk-1.4.15, and using AddQueueMember to add SIP
interface to the queue. Each sip interface is member of multiple
queues
The queue does not recognize that an agent is busy and keeps trying to
call the busy agent. I have identified two patches that can fix the
problem, one at
On Thu, Apr 3, 2008 at 12:16 PM, Rajkumar S [EMAIL PROTECTED] wrote:
If some one has a combined patch that addresses both this issues for 1.4.x
series
that would be great!
Just caughtup with Atis in #asterisk and got the url of
state_interface patch against 1.4.19, its at
http://ftp.iq
Hi,
I am using asterisk 1.4.19, my requirement is to find out which agents
were ringed by the queue when a call is abandoned (or connected) in a
call center. While this information is available in parts in
queue_logs and cdr, there is no way to correlate this information. For
example this is the
Hi,
Any one with any experience with VoIP hard phones or adapters
supporting speex? I looked around google but could not find any phones
supporting speex.
raj
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Hi,
I have a typical call center with queues and agents added via
AddQueueMember. One of my requirement is to implement a forgot
password function. If a caller does not remember the password, he
calls up an unauthenticated line and the agent manually authenticates
him. Then the caller should have
Hi,
I have a typical call center with queues and agents added via
AddQueueMember. One of my requirement is to implement a forgot
password function. If a caller does not remember the password, he
calls up an unauthenticated line and the agent manually authenticates
him. Then the caller should have
Hi,
While looking for the cause of disturbance in call I found this error
coming in console
RTCP SR transmission error, rtcp halted
Google search only shows some bug reports relating to MOH and Hold.
What could cause this message? Could this be a symptom causing call
disturbance? Where should
Hi,
How can I stress test an asterisk IVR? I am looking for some kind of
sip phone which can be programmed to send out digits after specified
time to simulate users pressing menu items. If it can originate large
number of calls simultaneously then it's great!
Does any one have any
Hi,
There are 3 new settings (setinterfacevar, setqueueentryvar,
setqueuevar) and membermacro settings in 1.6 queues.conf. What is
the potential use of these settings? The variables set are useful, but
there is no indication of the purpose they could be used? Any one with
some light on
On Wed, Feb 18, 2009 at 3:51 AM, David Backeberg dbackeb...@gmail.com wrote:
As for actually putting delays and pressing the right buttons, you're
on your own. You would need to write a custom AGI script specific to
your IVR, and call it from your call file, which you then put in a
bash loop.
On Tue, 17 Feb 2009, Mark Michelson wrote:
The purpose of exposing these values is to allow for an administrator to
use these for any purpose he may desire.
An example would be really great :)
I am confused because these values are exported just before the call is
connected and I am
Hi,
Russell's blog[1] is down and there are not much information about
this any where else. Any one with more information about res_ais and
how it is used?
raj
[1]
http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/
Hi,
I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via
two identities. Each asterisk server runs a queue and snom is a
member of queue in both servers. Currently when snom is receiving call
from one asterisk server, it can still receive a call from the other
asterisk, because
On Thu, Feb 19, 2009 at 8:28 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
Easy solution: Disable call waiting on the phone.
But asterisk will attempt a call since it's status is idle, and will
generate events which will confuse ADM I am using to display a url
for call.
Advanced
On Thu, 19 Feb 2009, Philipp Kempgen wrote:
Rajkumar S schrieb:
and will generate events which will confuse ADM I am using to display a
url for call.
ADM?
Asterisk Desktop Manager. http://adm.hamnett.org/
core show function DEVICE_STATE (on 1.6) is a good start.
Thanks.
raj
Resurrecting an old thread.
On Fri, 9 May 2008, Russell Bryant wrote:
Benoit Plessis wrote:
So i'm wondering if someone already as made a dialplan function that
could toggle the 'Use' flag of an agent ? or if this kind of function
would be integrated into the core if i build it ?
snip
Hi,
I am trying to get * log to mssql server. I have odbc and freetds
configured, but my insert query is missing calldate which is a NOT
NULL field in database schema.
cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed:
INSERT INTO cdr
Hi,
I am trying to log queue_log to odbc (MS SQL) I have res_odbc.conf
configured and modules.conf have
preload = res_odbc.so
preload = res_config_odbc.so
extconfig.conf has queue_log = odbc,asterisk.
When I start asterisk I get the following messages. The important one being:
Realtime
Hi,
I am using Asterisk 1.6.0.9. I have calls coming from another asterisk
server via IAX and lands in a queue. I have noticed that if there are
no agents logged in the queue no CDR is generated. If there is one
agent logged in then the phone rings and a CDR is generated even if
the call was
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote:
I'd try adding
transfer=no
in the B iax.conf
I'm guessing the box in the middle (B) is somehow transferring itself out of
the call
but retaining a ghost call entry.
It would be interesting to know what state those
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote:
iax2 show netstats
The show netstats gives:
a16-in1*CLI iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote:
I'd try adding
transfer=no
in the B iax.conf
This does not help, I still have some ghost calls in B
a16-in1*CLI core show channels
Channel Location State Application(Data)
IAX2/a16-in1-sangoma
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digium card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients
Hi,
The servers B C are running in a virtual machine (linux kvm) and
uses ztdummy for timing. Server A has a digium card. I am not sure if
this is the cause of the problems I am facing.
raj
On Fri, Jul 3, 2009 at 7:16 PM, Rajkumar Srajkum...@gmail.com wrote:
Hello,
I have a 3 server
Hi,
The servers B C are running in a virtual machine (linux kvm) and
uses ztdummy for timing. Server A has a digium card. I am not sure if
this is the cause of the problems I am facing.
raj
On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote:
On Fri, Jul 3, 2009 at 12:36 PM,
Hi all,
Did some more digging in. I changed the trunk from IAX to SIP and
still there are not much difference. So I guess it's not an IAX
problem. I have enabled DTMF logging and captured the DTMF logs for
two servers. (A: where E1 card is connected asterisk-1.4.25,
dahdi-linux-2.1.0.4) and B
Hi,
Is there a way to do call notification to a desktop when a call is
connected from a queue to an agent ? I have seen the call notification
page in wiki, but they do not deal with queues.
raj
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Hi,
I am looking for some docs to help configure a AudioCodes Mediant 1000
with asterisk, any tips or examples are appreciated.
raj
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Hi,
Is it possible to pick up a call that's in queue and pass it to an
agent directly. The use case is that some times some important calls
land up in queue which I need to pickup immediatly and pass it on to
an agent.
raj
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On 10/3/06, Greg Delgado [EMAIL PROTECTED] wrote:
I want to pop up a web page when a queue member phone
rings but, instead of displaying the clid, I want to
display the DID number the call came in. Any ideas how
to best implement this?
Checkout Asterisk Desktop Manager at
Hi,
I bought an asterisk TE110P to connect to our Reliance Infocomm E1
line to asterisk, I have loaded the driver, but looking for an
appropriate zaptel.conf and zapata.conf. I googled a lot but there
does not seems to be any india specific configuration. If any one has
successfully configured
Hi,
I am using 1.2.12.1 (actually was using 1.2.11, and upgraded) it's
connected to a Cisco ATA 188. The phones connected to ATA can register
to * and two phones connected to ATA can call each other. I can hear
Music On Hold, when called using the following fragment
exten = 6000,1,Answer
exten
On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
See if adding an answer line helps:
Rajkumar S wrote:
exten = 200,1,Playback(tt-allbusy)
exten = 200,n,Playback(moo2)
change to:
exten = 200,1,Answer
exten = 200,n,Playback(tt-allbusy)
exten = 200,n,Playback(moo2)
Nope
Hi,
I am configuring an audiocodes Medant1000 to talk to my asterisk box.
So far I have successfull in landing a single call from mediant to my
*box. my sip conf is as follows:
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[3911700]
type=friend
host=dynamic
dtmfmode=info
On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Has anyone used the AudioCodes MP-20x?
http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem to find anyone actually using
them...
I am using an AudioCodes Mediant1000 and now
On 10/23/06, Andrew Nowrot [EMAIL PROTECTED] wrote:
I've been testing this for 3 weeks now. No problems so far. This gateway has
many features including IPSec and is not that expensive.
Appreciate if you can post the sample configs to wiki or to the list.
There is no information about
Joseph wrote:
[EMAIL PROTECTED] wrote:
I am using firefly as my iax client, and it does not seems to work when
I use 1001,1,Dial(IAX2/1001) instead of 1001,1,Dial(IAX/1001)
Change the lines below from IAX to IAX2
Thanks a lot Joseph for your reply.
As you can see from my mail, I had
Bohuslav Coufal wrote:
Hi all,
is it possible to monitor RTP protocol (latency, errors, ...) by
Asterisk or other software.
Try http://tstat.tlc.polito.it/
quote
Tstat, a passive sniffer able to provide several insight on the traffic
patterns at both the the network and transport levels.
Paradise Dove wrote:
i have the same problem.
it seems to be a bug.
Is this related to the problem i posted yesterday (in a mail with
subject Zap channel not hangingup
raj
On 6/5/05, Master Abi [EMAIL PROTECTED] wrote:
Hi
I am trying to develop a night divert. Caller dials in after
Rich Adamson wrote:
The *proper* way to see it is with a voltmeter. your off-hook voltage should
be between roughly -5 and -15 Volts DC. CPD should either disconnect the
battery (0V) or reverse the battery (-5-15VDC) briefly upon remote party
hangup.
Just to add to Andrew's comment above,
Hi,
We have a small callcenter with about 5 agents, logging in via SIP
(SJPhone) using AgentCallbackLogin and incoming calls via Zap. I am
running Asterisk 1.0.9-BRIstuffed-0.2.0-RC8h (Rapid Distribution)
Some times some agents forget to logout when they go and when the next
person comes
Hi,
Can some one tell me what is the meaning of all the fields of show queue
callcenter? for example in my system it gives:
callcenter has 0 calls (max unlimited) in 'roundrobin' strategy (33s
holdtime), C:429, A:12, SL:0.0% within 0s
How is the holdtime calculated? what is A and SL?
Hi,
I have a small callcenter with 3 agents who login using
AgentCallbackLogin. They normally receive calls, but needs to call
outside also. When they call outside, though they are busy the show
agents shows them as available, and calls gets routed to them. How can
I make them busy when they
BJ Weschke wrote:
For your outbound calling problem, if you're operating with CVS-HEAD
you can PauseQueueMember and then UnpauseQueueMember as part of the
dial-plan for your outbound calls for those agents.
Thanks, I think this will do the trick. For short breaks, I can wrap
this around an
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the thank you message.
I started
Bump!
raj
Rajkumar S wrote:
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing
Capt MS wrote:
thanks for the reply
Is Digium card compatible with EPABX standards
available in india , further how much does a card with
three FXS and one FXO interface cost,
Do u have any experience of implenting the same ,
I am in army what we lookin at is voice gateway to
interface our
Hi,
I run a small inbound callcenter with 3 agents doing techsupport. The
agents are logged in via softphone, using agentcallback login. Some
times the agents PC running softphone hangs, and they reboot the PC. But
* is not aware of this and tries to send calls to the PC, which gets
Hi,
What must be done to enable callerid and call progress monitoring (disconnect
notification) for Zap lines connected to BSNL phones in India. I am willing to get
documentation, test or write the necessary code to get it working. I have gone through the
indications.conf, will that be
Gurminder Arora wrote:
Hi raj,
Perhaps both of us are going through same tunnel...
Along with all the Zap users in India :) Occasionally there will be a
post in the list about Zap support for India, but even now there is no
CallerID or Call Progress monitoring for India. I know a bit of
Hi,
Is it possible to monitor conversation of logged in Agents? Currently I
am using ZapScan to monitor incoming calls, but I would like to monitor
individual agents.
raj
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Hi,
Is it possible to define maximum talk time in a queue? ie any one who
joins a queue should not be able to talk more than say 5 minutes to
the agent.
raj
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To
an old call center system with asterisk, which had this
facility. So the clients are pretty used to getting cut at 5 minutes.
So every one tries to make the calls short and sweet :)
thanks a lot,
raj
On Wed, 25 Oct 2006 15:06:35 +0200, Rajkumar S
[EMAIL PROTECTED] wrote:
Hi,
Is it possible
Hi Lenz,
On 10/26/06, Lenz [EMAIL PROTECTED] wrote:
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
In this dial command you're free to add whatever option you may like,
including the ones to limit call length.
I hope this helps
That did help. Thanks a lot!!
raj
On 10/26/06, Lenz [EMAIL PROTECTED] wrote:
When you log in a callback agent, you enter first the agent code, and then
the extension he's sitting at. The context is usually specified in the
dialplan command, but the result is that asterisk knows that agent 103 is
sitting at [EMAIL PROTECTED]
Hi,
I have a requirement to limit the calls to our agents via a queue to 5
minutes. I had posted this to a previous thread by name Maximum
talktime in a queue? One work around that was suggested was to use
the S(x) in the dial command to the agents, so that all calls to that
extension would be
On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Rajkumar S wrote:
-- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack
-- Started music on hold, class 'default', on channel 'SIP/1002-74e9'
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1
On 11/1/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Rajkumar S wrote:
On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Someone correct me if I'm wrong: The Dial string is missing a '/n'
parameter for the Local channel. Without /n, Asterisk will do a native
transfer to SIP/1001 and lose
Hi,
Any one using Rubi asterisk manager interface
http://rubyforge.org/projects/rami/ ?
How stable/usable it is?
raj
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Hi,
If I use L(x[:y][:z]) in Dial app the call is limited to x
milliseconds, Is it possible for the callee to extend the call past x
milliseconds?
raj
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On 11/13/06, nik600 [EMAIL PROTECTED] wrote:
i have an application developed with bayonne.
I would like to know if i can do these things whit asterisk:
- IVR integration with database (mysql, insert,delete,update,select)
Yes, you have to write AGI scripts to do this.
- TTS
No idea.
-
Hi,
When a call is made to a queue and picked up by agents at least 2 CDR
entries are made, one from local to the agent's (sip) phone, and from
incoming line to Agent. There are other entries generated when other
conditions happen, like agent do not pickup phones and so on.
Going through the
Hi,
I am using ChanSpy with Asterisk 1.2.12.1. My extensions.conf has the
following lines for ChanSpy
exten = 1234,1,ChanSpy(Agent)
exten = 1234,2,Hangup
When I dial 1234 I can listen to one agent talking, but nothing
happens if I press * or another agent number followed by #.
Also archives
On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote:
anyone know the status of disconnect supervision on POTS lines in India?
Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have
disconnect supervision..
It does not work afaik, you may not get caller id also. I tested upto
On 1/1/07, ram [EMAIL PROTECTED] wrote:
On 12/30/06, Rajkumar S [EMAIL PROTECTED] wrote:
On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote:
anyone know the status of disconnect supervision on POTS lines in India?
Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have
Hi,
This is slightly off topic, but here I go any way...
VoIP traffic has lot's of smaller packets, and since each packet can
generate an interrupt, is there any way to determine the irq rates in
a machine, and more importantly to know if I am hitting any of the
limits in Linux or to determine
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt. From what I understand the basic
idea in the example is to
1. Authenticate a caller with VMAuthenticate
2. Get his SIP Channel number
3. Use
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt. From what I understand the basic
idea in the example is to
1. Authenticate a caller with VMAuthenticate
2. Get his SIP Channel number
3. Use
On Jan 4, 2008 4:21 PM, BJ Weschke [EMAIL PROTECTED] wrote:
AddQueueMember(queuename[|interface[|penalty[|options[|membername):
Thanks BJ Weschke and Alexandre Snarskii. Your mails together gives
complete solution to my problem!
raj
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Hi,
I have a standard E1 line, but want to receive only 10 calls
simultaneously. I want to give engaged tone to the 11th caller
onwards. Can I configure E1 to do this?
raj
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Hi,
I am moving my call center to 1.4. Previously I was recording calls in
agents.conf with the following config
recordagentcalls=yes
recordformat=wav
createlink=yes
So I had the filename in all calls which was *connected to agents*. I
am looking for a similar functionality for 1.4.
I am now
Hi,
I have a queue with one agent added using AddQueueMember
(FAO|Local/[EMAIL PROTECTED]|0||Agent/602). My extensions.conf is
[general]
static=yes
writeprotect=yes
autofallthrough=no
clearglobalvars=no
priorityjumping=no
[from-sip]
exten = 11000,1,Dial(SIP/11000,,t)
exten =
Hi,
I am using asterisk-1.4.15
I have a queue with one agent added using AddQueueMember
(FAO|Local/[EMAIL PROTECTED]|0||Agent/602).
Once this command executes queue show FAO shows:
FAO has 0 calls (max unlimited) in 'roundrobin' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within
On Thu, Feb 14, 2008 at 9:52 PM, Naveen Palani [EMAIL PROTECTED] wrote:
How can i pass the arguments from my dialplan to the ruby file. Is there a
way i can do it with the agi script?
Set them as variables in your extensions.conf and use them inside your
agi scripts.
raj
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients
Hi,
I am using RT engine to log queue_log to a mysql database. My extconfig is
[settings]
queue_log = mysql,asterisk16_production
Logging to mysql is working fine.
But I find that the queue_log file now only has QUEUESTART lines for eg:
1250519094|NONE|NONE|NONE|QUEUESTART|
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over SIP trunk from which
calls get routed to third server (C) (1.6.0.9) via IAX trunk.
SIP clients are
is more even.
Thanks and regards,
raj
On Fri, Aug 14, 2009 at 12:31 PM, Rajkumar S rajkum...@gmail.com wrote:
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B
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