The idea is that the Queue() application uses different strategies to ring
agents, so it decouples you from having to worry about that. You could have
that by setting the queue to rinagll strategy.
l.
2010/1/25 bhrugu mehta mehtabhr...@gmail.com
Hi, all
Is ther any way to pass channel queue
I know this is not what you need, but you might postprocess recordings to
raise the volume level. I know this is not optimal but it's a start.
l.
2010/1/21 Scott Gifford sgiff...@suspectclass.com
Hello,
We are recording our calls to queues by putting the appropriate options in
our
Maybe I'm saying something stupid, but I thought this was what
shared_lastcall would do with a leastrecent strategy.
; shared_lastcall will make the lastcall and calls received be the same in
; members logged in more than one queue.
; This is useful to make the queue respect the wrapuptime of
Yes it's actually quite simple to do. If you want, the free version of
QueueMetrics is able to do that from the Agent's page.
l.
2010/1/13 Zhang Shukun bit...@gmail.com
2010/1/12 Lenz Emilitri lenz.lo...@gmail.com:
You can list phones directly as static members of the queue.
i know i can
Yes it is - we have thousands of happy clients worldwide. :)
My suggestion is to go for somebody who has relevant experience and is going
to do the install for you. Unless your CC is very small, you don't want to
be looking up the manuals when you went live and start having quality
issues
If
You can list phones directly as static members of the queue. this is
generally sub.optimal because if. e.g. an agent of yours is home sick, her
phone will be ringing and you'll be wasting caller time. Also by tracking
logins and logoffs you can measure agent productivity, and this is pretty
useful
If I can give a suggestion, do save the files to a different folder per
day/queue or it will get unmanageable at warp speed :)
l.
2009/12/31 Yuval Yogev yuva...@yahoo.com
I installed an Elastix based system and changed it to work in Device-Mode
since there is a call center and users has to
I think that ssmtp has something like that, IIRC. I think it can spool
failed deliveries and you can then push them though a cronjob.
l.
2009/12/20 Darrick Hartman dhart...@djhsolutions.com
On 12/20/2009 11:38 AM, meetmecall wrote:
I used msmtp for delivering mail and this is the procedure
As it is done today, the wrap-up time is not terribly useful in Asterisk, as
it is fixed-length. If you need to implement it in a real-life scenario, it
would be better to pause the agent when the call is through and have him
unpause manually when he's done the wrap-up; this way you get a
Hello list,
we have been working on a simple dialplan replacement for
AgentCallBackLogin. It is not nearly as polished as we hope for, but it
should be a working start.
It offers the following features:
- Single log-on and log-off, managing queue/agent associations centrally
- Agent pause (with
03:20:11 pm Stephen Davies wrote:
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
But more dynamical, so I would try and look up the actual channel in
the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is there a way to work
Thanks that's exactly what I was looking for! I had seen a patch for it but
did not notice this was in the main trunk.
l.
2009/12/14 Stephen Davies stephen.l.dav...@gmail.com
What you are missing is the new state-interface parameter to
AddQueueMember.
You can't use functions in a hint
See if you find this tutorial on IAX peering useful:
http://astrecipes.net/index.php?n=204
Thanks
l.
2009/12/15 Landy Landy landysacco...@yahoo.com
Hello List.
I have a question regarding connecting two asterisk servers. I'm trying to
learn how asterisk comunicates from server to server. I
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
I would like to do something like:
[myagents]
exten = XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
exten =
An alternative would be using a screen-pop application that links to a small
CRM app that will do the display for you.
l.
2009/12/7 Giedrius Augys voi...@gmail.com
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call
This is a very broad question. why don't you tell us something more
about tyour setup?
l.
2009/11/26 Daniel Stefanus shinichikud...@gmail.com
Hi guys,
Having a little problem.How can I know where queue is my agent login
from my CDR table?
Sorry my English's terrible.
Best regards,
Yes why not? when the agent is connected it can read the variables on the
calling channel what would you like to build with that? :)
l.
2009/11/24 Shaun Clark shaun_cl...@hotmail.com
Hello,
I was wondering if their is a way to use the Asterisk ACD to initiate a
call that will route
philosofy we could implement some easy marco that only ask
for the password and:
1.- sets the astdb
2.- sets the globals AGENTBYCALLERID_X=
3.- adds the agent to the queues.
Let me work deeper on this idea and see what comes up.
ML
2009/11/2 Lenz Emilitri lenz.lo...@gmail.com
We were
We were thinking about doing something similar as well. A lot of people are
asking for this. If there is anybody else interested, we could share the
load
I was thinking about creating a context like @agents, so that when you do
the log-on you basically add Local/1...@agents as a member of the
To avoid boring everybody else to death with the discussion, I created a
mailing list for that on Google Groups - see http://tinyurl.com/yjtf62s
Thanks
l.
2009/11/2 Lenz Emilitri lenz.lo...@gmail.com
We were thinking about doing something similar as well. A lot of people are
asking
You may want to start from the basics:
http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
I hope this helps
l.
2009/10/17 Nazir Ahmed Vaid nazir.v...@gmail.com
Ladies and Gentlemen,
We already have an Asterisk Call center suite installed at our contact
center. Now we wish to commence
Use an existing dialer like ViciDialer?
l.
2009/10/14 kaustuva...@bbsr.syscomes.com
Hello,
I am using Asterisk 1.4 version.
How to dial multiple numbers per second through asterisk manager
Thanks and regards
--
Loway - home of QueueMetrics - http://queuemetrics.com
You could configure them as agents and have them log off automatically after
a while they're not responding.
l.
2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP
A number of our clients has such issues. What we suggest for escalation is
to do a blind transfer to a second-level queue, so that the logging is
correct and even if second-line support cannot handle the call immediately,
you get the functionality and the logging.
Just my two euro cents,
l.
If you need this from the moment that the agent connected you can measure
the length of the leg that goes with the agent.
If you need to measure this from the moment the call was answered, you can
measure the length of the 'main' cal of the leg (the one that called the
command queue())
If you
It's a bit off topic here (I would ask this on a QM or TB forum), but
basically you redirect each IVR selection to a context where logging happens
and then redirect to the queue.
Just my two eurocents,
l.
2009/9/16 Maria Cristina Bayno falls_m...@yahoo.com
Hello Team,
IVR selection of
As an ultra cheap way of doing it, you could simply output the caller-id to
a log file and display a tail 20 of it on a web page.
Something like this:
exten = s,1,System( echo${EPOCH}|${CALLERID(num)}
/var/log/asterisk/incoming )
It should be trivial to display the last n lines of it on a web
It depends on what you want to do to people who are queued; if you want them
to be queued, you create a queue with only one member, and have agents log
on and log off as necessary; if you don't want callers to be queued, likely
I would not use a queue but woul dial the agent straight.
l.
PS. this
Aht i would do is prepare a music on hold that has embedded the
advertisements ( like one every 20 or 30 seconds) so that the caller hears
more advertisements as the call progresses; and they are queued immediately,
so no time is wasted.
l.
2009/8/27 Andy Kuo aku...@gmail.com
Hi Barry,
Thank
It's here: http://queuemetrics.com/download/qloaderd-1.17.tar.gz
It's technically a part of QueueMetrics, but it does not require a licence
to run.
Feel free to use it. :)
l.
2009/8/18 Miguel Molina mmol...@millenium.com.co
Lenz Emilitri escribió:
You should log to a file and use a piece
Have you tried unloading and reloading the zaptel driver?
l.
2009/8/18 Raimund Sacherer r...@runsolutions.com
Hello List,
our setup:
Callcenter
IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular
providers on the xircom analog port, ~60 agents
Debian 5.0.1 (Lenny)
You should log to a file and use a piece of code like our qloaderd to do the
DB update.
l.
2009/8/17 Rajkumar S rajkum...@gmail.com
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
How do you do the log-on?
l.
2009/8/6 Joao Gomes Pereira gomespere...@startel.pt
Hello to all
I have a queue where often my agents get stuck and cannot logoff.
This is very bad, because agents cannot login again, and in Queuemetrics
reports the agents appear to be online.
How can I create a
If you are completely new to Asterisk and want to run a professional
call-center, my suggestion is to stick to a hand-made, lean, minimal
configuration.
l.
2009/7/13 ashish chauhan ashishchauhan07...@gmail.com
Dear all,
I am new to asterisk.i like to configure call center using
I believe this is more a human resources problem than a technical one. You
will first need some sort of CDR analysis tool to spot calls to expensive
destinations, and then you wil track back who was the agent in change of the
call.
I am sure that if there is word out that you are tracking these
Well, at least this did not add to the wait time of your callers :)
It should be possible to do silence detection/removal automagically using
sox as well - see e.g.
http://www.justlinux.com/forum/showthread.php?t=136678
2009/6/18 Louis-David Mitterrand
You could try this one:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
If I can add a warning, be wary of having both ACD (Queue) and non-ACD
traffic on the same operator - you risk having awful performance.
Just my two eurocents,
l.
2009/6/12 Lee, John (Sydney) john@compuware.com
You should look on the log for when the sox command is called, if the
invocation makes sense or not.
l.
2009/6/7 Joao Gomes Pereira gomespere...@startel.pt
Hello
I did as you told me, but the problem remains.
Im using Asterisk 1.2.x
and this is my config:
queues.conf
Maybe this can help you? http://astrecipes.net/index.php?n=286
Thanks
l.
2009/5/31 Tamer Higazi th9...@googlemail.com
Hi people!
I am looking for a h.323 implementation guide for asterisk. I looked in
the doc folder of the latest asterisk source distribution and I didn't
fund anything
You have to consider the POW of the ACD system - you can transfer calls as
much as you like, but you have to make sure that:
- transfers are logged, and
- the ACD queue app has the correct state for all the agents involved, so it
does not try ringing agents that are busy and (worse) does not ring
Thank you! I updated the tutorial as well.
l.
2009/5/25 Atis Lezdins a...@iq-labs.net
On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri lenz.lo...@gmail.com
wrote:
Hi everyone,
after doing the same thing multiple times and struggling to remember how
it
was done, I have prepared a small
Hi everyone,
after doing the same thing multiple times and struggling to remember how it
was done, I have prepared a small tutorial that explains how to save
monitored files in different folders per day. This is quite useful
becausethe resultingfile system is way more manageable than having maybe
What exactly are tyou trying to achieve?
l.
2009/5/20 Kurian Thayil kurianmtha...@gmail.com
Hi All,
I am trying to implement ACD using Asterisk 1.2.18 and I've chosen
AgentCallbackLogin for login purpose. One AGI is written which will actually
get executed when agent dials '1001' (say) from
The main problem i see with thgis is that with old-school agents, you
could easily have association with multiple queues. And that was quite
useful.
l.
2009/5/16 David Anthony O Reilly oreil...@tcd.ie
Hi Jim
Thanks for your code!! I see you use the Voicemail system to authenticate,
have you
We use something like that in QueueMetrics to track outgoing calls for
call-centers:
http://forum.queuemetrics.com/index.php?topic=261.0
thanks
l.
2009/4/25 Sebastian s...@adinet.com.uy
Anyone thought about something like outgoing queues?
I mean, having same info that has for inbound queues
I think that all existing GUIs will in time migrate to Asterisk 1.6, if they
are not already supporting it, so using your favourite one should not be
much of an issue.
l.
2009/4/20 Gary Li garyli0...@gmail.com
Hi,
I had some experience on Asterisk 1.0.7 and 1.2.0.
Now, I want to do
My suggestion is to use a tool made specifically for this - we happen to
sell one, but there are many options with different prices and licencing
model. Don't reinvent the wheel and concentrate on added value.
l.
2009/4/14 Scott Gifford sgiff...@suspectclass.com
Hello,
I'm working on an
You tried setting the call limit for the Agent's phone?
l.
2009/4/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Apr 2009, Haim Dimer wrote:
The issue is the that the agent needs to wait on the phone for a call to
come in. I read
Are these functions what you are looking for?
QUEUE_MEMBER_COUNT: Count number of members answering a queue
QUEUE_MEMBER_LIST: Returns a list of interfaces on a queue
QUEUE_WAITING_COUNT: Returns the number of callers currently waiting in a
queue
Just my two eurocents,
l.
2009/3/31 Steve
You could store the who is who information in Asterisk, so you know
thatSIP/123 is Agent/301 before logging the agent - see e.g.
http://queuemetrics.com/faq.jsp#faq-038-agent_tracking
Thanks
l.
2009/3/27 Miguel Molina mmol...@millenium.com.co
Hi all,
For those of you people that use
The problem with this seems to be that when you make a distro, you want it
to be many things to many people (easy to use, lots of features, support
lots of hardware, you name it). When you build a medium/large call-center,
you usually want to keep it lean and mean, as you need a high uptime and do
This should be engraved in stone. IMHO, doing so even with a traditional
telco solution would be extremely risky, if one does not have an adequate
skill set and experience.
Thanks
l.
2009/3/26 Matt Riddell li...@venturevoip.com
If you are doing an install for a call centre with 100-200
As you're using a SIP channel, likely you are not limiting the number of
calls. Try setting limitonpeers and call-limit.
Hope this helps,
l.
2009/3/20 Cary Fitch ca...@usawide.net
[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device
state of this queue member,
Hello Giorgio,
you simply pass that parameter along so that from the QueueMetrics agent
page you get that URL opened automagically when you get a call. It's for
interfacing to external CRM apps, usually passing the agent code that
handles the call, the Asterisk unique-id and the caller-id for
You should look at the queue() command invocation.
Thanks
l.
2009/3/12 Darrin Henshaw dhens...@ignition.bm
Hello,
We had an incident recently where a call was in queue for an extended
period of time. We use queuemetrics for reporting, and it reports that the
call was waiting for 20
I'm only half joking: what about parsing the full log looking for command
inviocations and channel IDs? this would be completely transparent, albeit
insane :)
l.
2009/3/12 nik600 nik...@gmail.com
Hi to all.
What can i do if a customer needs to log in the CDR all the dialpan
actions related
You could try our qloaderd - it was made for MySQL, but it should be simple
enough that by changing the engine should be a no-brainer (it's a perl
script). You get the added advantage that in case anything goes wrong with
the DB system, you lose no data.
It is here:
IIRC, some early dialler of the pre-AMI era used this technique to control
the number of calls placed simoultaneously - they just counted the number of
call files in the spool dir. As they are deleted when the call is over, this
was a simple way to do the throttling.
You could use a similar
channels, thus
covering all bases.
Thanks a lot for your help!
l.
2009/2/20 Klaus Darilion klaus.mailingli...@pernau.at
Lenz Emilitri schrieb:
I think this is by design - each time the Dial() is performed, SIP
headers are reset.
No.
SIPAddHeader adds global channel variables to the incoming
Speaking of hatred, I know of a call-center (no names will be made) that
does pre-qualifying by calling and seeing if anybody answers; and passing to
the main dialler for a separate call if they actually do. :)
l.
2009/2/19 Jon Pounder j...@inline.net
ts stuff like this which makes
You can know if the queue is full before issuing the answer() or the queue()
command, so you can avoid answering at all.
l.
2009/2/19 Alex Hermann a...@speakup.nl
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
One simple thing that comes to my mind is to have the SNOM connected to only
one server, and send calls to from the queue on the second server to the
first server, so that you can enforce a acall limit.
l.
2009/2/19 Rajkumar S rajkum...@gmail.com
Hi,
I have a snom 360 connected to two
Hello list,
I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level.
If I use the following code:
exten = s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten = s,n,Queue(myQueue)
this works fine for the FIRST call
I think this is by design - each time the Dial() is performed, SIP headers
are reset.
l.
2009/2/18 Benny Amorsen benny+use...@amorsen.dk benny%2buse...@amorsen.dk
Lenz Emilitri lenz.lo...@gmail.com writes:
If I use the following code:
exten = s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID
.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
*Sent:* Wednesday, February 18, 2009 3:05 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Setting SIP header on agent calls
on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
If you use the free version of QueueMetrics, you can have the queue URL
parameter passed along and each agent can open up an external app using the
web interface. As this part is not linked to the main stats module, it works
just fine for all of your agents with no limitations.
Thanks
l.
I have a feeling we're overdoing it :)
l.
2009/2/12 Lukas Rypl r...@marconi.ttc.cz
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000
Hi,
I used this way of processing output from asterisk 1.2 and found out
that it is not 100% safe because there can appear
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get the
channel name and
We use the default web interface, it comes with a file called vmail.cgi in
the defaiult Asterisk tree.
Thanks
l.
2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com
Hi,
I'm very new to Asterisk. I tried VoiceMail() application. I'm able to
modify extensions.conf and voicemail.conf
[mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
*Sent:* Friday, January 16, 2009 10:09 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How to transfer a call from one
AsteriskServerto another
I guess you already tried
Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?
l.
2009/1/16 Paul bulkm...@monafamily.com
Can anyone tell me how I can completely move an established call off of
one Asterisk server to another?
In our case we have a server with our IVR.
.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
*Sent:* Friday, January 16, 2009 12:17 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk
You could simply have it Dial() to wherever it needs to go at the end of
the script.
2009/1/6 Rajkumar S rajkum...@gmail.com
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
I have a small script that I use to control queue access, it's an AGI script
that lets you define on-off periods on a weekly basis plus holidays. I never
get around to publishing it, though I find it quite useful - if anybody is
interested, I'll clean it up and share it :-)
l.
2008/12/23 Dan
I think that mosty operators selling DIDs will have an Asterisk tutorial on
how to use it. If they don't, don't buy it from them :-)
l.
2008/12/28 Abel Monzon abelcub...@gmail.com
Hello there!!, Am looking for a manual or documentation that explain
how to buy a DID number and how to
You could use a very simple approach and just disconnect them all without
caring if they are connected or not.
Something like:
asterisk -rx agent logoff agent/101
asterisk -rx agent logoff agent/102
asterisk -rx agent logoff agent/103
..and so on
to be executed at a given time.
Of course I
So what is the middle name that causes problems? atre you sure you don't
have strange characters in it, like spaces, nonprintables, weird
encodings, etc?
l.
In data Wed, 17 Dec 2008 00:35:04 +0100, Eve Ellen Cole
ec...@mail.plymouth.edu ha scritto:
I’ve got an interesting problem and
My suggestion is to have an agent log-off from the ACD system (or at least
pause) before attempting outbound. You can achieve that with some handy
macro that might do this transparently before the call is placed and when it
terminates.
Just my two cents,
l.
2008/11/5 Ricardo Melendez [EMAIL
I think you could minimize the incidence of the problem by having a PRI
with like 100 numbers associated, with the CO doing the routing stripping
off the last two forwarded digits. You also have a premium service
provider that forwards premium calls to one of those numbers (I think from
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Thanks
Look for the DeadAGi command.
Thanks
l.
On Thu, 12 Jun 2008 13:41:14 +0200, voip crazy [EMAIL PROTECTED] wrote:
Which is the way to run an AGI after hangup a call?
The problem I have is when the call dies the AGI dies too
I try the Dial command g option, but it does not work for me
Any
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos [EMAIL PROTECTED]
wrote:
Hi,
How can i get the deadAGI to work at this scenario
Basically when someonc calls
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the folllowing commands...
exten =
exten =
You could use a real agent with autologoff, for instance. Of course
there may be drawbacks in using the agent channel, though it's usually a
reasonable choice for most setups.
Thanks
l.
On Fri, 16 May 2008 05:39:52 +0200, Rilawich Ango [EMAIL PROTECTED]
wrote:
Hi all,
There is a
Hi Steve,
we tested Druid for a while - our setup was pretty much call-center-ish,
so we did not do much testing of hardware and advanced PBX
functionalities, while we used it to build queues and trunks and
extensions.
Generally speaking, it looked quite polished and very easy to use
I don't want to add to the list of supported players - my suggestion is to
do the recordings as WAV, so you get the same quality and compression
results but it's playable natively on any Windows box.
Another choice to get this issue solved, if you have a system that is not
24/7 in use, why
I usually use lame to do the decoding and pipe it back into sox.
l.
On Fri, 16 May 2008 12:58:41 +0200, Julian Lyndon-Smith
[EMAIL PROTECTED] wrote:
Does anyone know where I can a copy of sox for windows with mp3 built in
?
Julian
David Backeberg wrote:
No, no, no.
Don't try to
It should already work, unles you configured your queue differently? :)
l.
On Tue, 13 May 2008 14:44:44 +0200, bilal ghayyad [EMAIL PROTECTED]
wrote:
Hi list;
Any one can advise how to put the caller in the queue
in case no one available to take his call? All are
busy (having calls)?
Rule of thumb: you first try without the /n; if the new behaviour is
different from expected, add the /n
:)
Just my $0.02
l.
On Tue, 01 Apr 2008 17:33:05 +0200, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote:
Does anyone know the purpose of /n
Lord [EMAIL PROTECTED] wrote:
Lenz wrote:
Hello list,
after spending the best part of an afternoon trying to build Asterisk on
an old EPIA VIA C3, I thought that writing a tutorial would make life
easier for future compilers:
http://astrecipes.net/index.php?n=356
I had never compiled
Hello list,
after spending the best part of an afternoon trying to build Asterisk on
an old EPIA VIA C3, I thought that writing a tutorial would make life
easier for future compilers:
http://astrecipes.net/index.php?n=356
I had never compiled Asterisk for a different architecture, and I'm
Don't worry - I paste this leink becaus eyou should have e good
understanding about what the queue() cmd does to be safe in implementation
phase: http://www.voip-info.org/wiki-Asterisk+cmd+Queue
See also: http://astrecipes.net/index.php?n=118
Thanks
l.
On Tue, 05 Feb 2008 06:31:16 +0100,
You create three queues: queue A has only agent A, queue B only agent B,
and queue C only agent C.
You call the firts queue witha timeout of 120 seconds; if call timed out,
you call queue B with a timeout of 120 seconds and so on.
One note: this does not sound great from a service-level point
Really, what I would do is to set up a daily restart point when there is
no or very little activity, something like running nightly:
asterisk -rx stop when convenient
and then having the monitoring script restart it immediately. Do you need
to unload the zaptel modules as well or is
Hello there,
I just wasted some time setting up a Grandstream HT-488 to be used with
Asterisk, so I thought I'd share the experience by writing a small
tutorial at: http://astrecipes.net/index.php?n=338
Most tutorials I came across were for old versions of the firmware, and I
spent too
I believe you should define the agents in agents.conf as well! :-)
l.
On Wed, 21 Nov 2007 19:39:46 +0100, Gregory Malsack [EMAIL PROTECTED]
wrote:
Hello All,
I am hoping someone out there can enlighten me on this issue. I am using
asterisk 1.4.11. We have a call queue setup, and our
I have never tried doing this myself, but we use Bugzilla as a
well-working bug tracking tool, and it has an import script called
importxml.pl that can be used to import bugs using its own XML format. So
you would likely need some kind of AGI to create the XML and then run
importxml.pl
Try this:
sox -r 44100 -w -s -c 1 myfile.raw -r 8000 -c 1 myfile.wav
I hope this helps
l.
On Tue, 13 Nov 2007 15:26:07 +0100, Gary [EMAIL PROTECTED] wrote:
I used ChanSpy( ) recorded some test conversations. It has .raw
extension.
What kind of audio file is this? How can I play it?
The http://voipUsersConference.com/ning seems not to be working.
l.
On Thu, 15 Nov 2007 12:03:56 +0100, randulo [EMAIL PROTECTED]
wrote:
Hi,
Tomorrow, Friday Nov 16th, 2007 at 12:30 PM, we'll be exploring a
simple, well-commented example of an AGI script for asterisk. I have
absolutely
101 - 200 of 480 matches
Mail list logo
