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empire underground

2004-Jan-09 11:32 UTC

[Asterisk-Users] Screen Pop & Remote Agents

<html><div style='background-color:'><DIV class=RTE>Hi
Everyone,<BR></DIV>
<DIV class=RTE>&nbsp;&nbsp; Im new to Asterisk and I would like to
get some imput from all of you.<BR>First I would like to start by telling
all of you I am starting a call center Within the next 2 weeks with 12 agents to
start. I have also been looking at alot of dialers! The prices are outragous! So
a friend of mine has turned me on to Asterisk ;). Now The info I need IS what
cards do you think I should get? I've heard alot of good things about the
digium cards, also I will be on a analog T1. So I will need to know what cards
to buy?<BR></DIV>
<DIV class=RTE>Ok now for some of the more tricky stuff. Is is possible to
have screen pops for my reps through a web site? So when the call is connected
the yget all the coustomers info on screen? For the second tricky part is can I
have remote agents? (reps working from home and log into the pbx by either Voip
or even Dialing in)? I saw somewhere when I was searching That someone developed
a windows screen pop client, but I would like to know if their is one for linux
as well? I am trying to stay away from M$ as much as possible! as for some of
the more basic questions, can I put a .csv file in the sql DB and have it dial
from there? and will I be able to set a Dial Plan to only call certin area
codes? stuff like that. The reason I ask all this is because all of these over
priced dialers do just that. Also can Asterisk be set with the FTC laws to 3%
droped call ratio?<BR>If all of the questions I have asked here have
allready be answered some point in time... can someone please point me in the
right direction to get all the answers.<BR>Thank you
<BR>&nbsp;&nbsp; Christopher</DIV></div><br
clear=all><hr> <a
href="http://g.msn.com/8HMBENUS/2737??PS=">Get reliable dial-up
Internet access now with our limited-time introductory offer. </a>
</html>

Philipp von Klitzing

2004-Jan-09 12:39 UTC

head link

[Asterisk-Users] Screen Pop & Remote Agents

Hi!
>  Im new to Asterisk and I would like to get some imput from all of you.
> First I would like to start by telling all of you I am starting a call 
> center Within the next 2 weeks with 12 agents to start.
Asterisk is probably a very good choice for you, however you'll need more 
time than those 2 weeks to be fully operational (unless you get some 
consultant to assist you).
> has turned me on to Asterisk ;). Now The info I need IS what cards do
> you think I should get? 
Before spending tons of $$$ the first step is to install Asterisk and 
test it with IP phones (soft- or hardware). Look at queues and agents and 
check out if they do what you need. Have a look at GnoPhone if you need a 
Linux based software phone that includes a web browser and can take URL 
arguments along with a call.

http://www.voip-info.org/wiki-Asterisk+call+queues
http://www.gnophone.com/
> pops for my reps through a web site? So when the call is connected the 
> yget all the coustomers info on screen?
Use GnoPhone, or write your own little app (using PHP or PERL etc) that 
regularly reloads the call agent's browser (you might be a bit too slow 
that way, so it might be necessary for the agent to initiate the reload 
manually). If you prefer fance stuff then you could use FLASH instead of 
HTML with the help of PHP, ming and Actionscript.
> For the second tricky part is can I have remote agents? (reps working
> from home and log into the pbx by either Voip or even Dialing in)? 
Yes - the Wiki link above about "call queues" has the info and links
that
you need to look at.
> as much as possible! as for some of the more basic questions, can I
> put a .csv file in the sql DB and have it dial from there? 
Take a look at sample.call in your Asterisk installation. See also:
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
> and will I be able to set a Dial Plan to only call certin area codes? 
You mean you want to dial random numbers within a certain area code?
I wouldn't consider that to be a nice way of doing things...
> reason I ask all this is because all of these over priced dia rs do
> just that. Also can Asterisk be set with the FTC laws to 3% droped
> call ratio? 
Explain?


Cheers, Philipp

CW_ASN - Gus

2004-Jan-09 15:13 UTC

head link

[Asterisk-Users] Screen Pop & Remote Agents

<snip>>
> Yes - the Wiki link above about "call queues" has the info and
links that
> you need to look at.Also, could be great is you install a new patch, to add some great
functionalities to your call center. This path is located:
http://bugs.digium.com/bug_view_page.php?bug_id=0000214

Regards,

Gus

empire underground

2004-Jan-09 16:01 UTC

head link

[Asterisk-Users] Re: Screen Pop & Remote Agents

<html><div style='background-color:'><DIV class=RTE>
<P>The FTC ratio is what the FTC allows for droped calls, say the dialer
is calling 10 phone #'s per agent connected. As for the ip phones are they
needed? or can I just use regular phones with headsets? and as to the 2 week
setup I know it will take alot longer but I have a temp dialer until I can get *
up and running. All Im really looking for * to do is Remote agents to be able to
log in anywhere in the USA, and for it to only make outbound auto dialing or
predictive dialing if possible. The dialer Im using right now allows remote
agents, predictive dialing, and time zone selecting for a dial plan (calls by
state in the DB). When connected to the dialer the rep loges in and the phone
rings the rep picks up the phone and gets dead air. Then when the dialer gets a
live person on the phone the call is routed to the next available agent in
queue. When the call is transfered to the agent they get a screen pop in a
webpage with the persons info (name, address, ect...) when they end the call
they click a button to log the call in the web page that pops up. ie... call
back, do not call, paid, ect... then the call drops and the agent still has dead
air waiting for another call. Now i dont mind if the agent logs in and the phone
has to ring and they have to pick it up but the screen pop has to be on the fly
and come up as soon as the live person is on the phone. Now even with a
predictive dialer some garbage comes through like answering machine and so on...
but its far and few between.<BR><BR></P></DIV>
<DIV></DIV>>From: asterisk-users-request@lists.digium.com 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Subject: Asterisk-Users digest, Vol 1 #2421 - 15 msgs
<DIV></DIV>>Date: Fri, 09 Jan 2004 14:20:04 -0600 
<DIV></DIV>> 
<DIV></DIV>>Send Asterisk-Users mailing list submissions to 
<DIV></DIV>> asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>To subscribe or unsubscribe via the World Wide Web,
visit
<DIV></DIV>>
http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>or, via email, send a message with subject or body
'help' to
<DIV></DIV>> asterisk-users-request@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>You can reach the person managing the list at 
<DIV></DIV>> asterisk-users-admin@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>When replying, please edit your Subject line so it is
more specific
<DIV></DIV>>than "Re: Contents of Asterisk-Users
digest..."
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>Today's Topics: 
<DIV></DIV>> 
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;1. Re:
Mailing list growth (Bob Knight)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;2. RE: SIP
and error talking to voicemail (Leopoldo Santiago)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;3. IConnect
audio quality (Chris Albertson)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;4. USA dial
plan (Senad Jordanovic)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;5. Re[2]:
[Asterisk-Users] Help with compiling
(=?ISO-8859-1?B?Um9iaW4gQ2FsbWVn5XJkIFNpdXJ1YQ==?=)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;6. Re: *
dialing before line is open? (Steven Critchfield)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;7. Re: latest
cvs == broken tdmoe (Sean Swallow)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;8. zapbarge
w/o the mute (john lawler)
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp;9. Re: Cisco
Gear (Iain Stevenson)
<DIV></DIV>>&nbsp;&nbsp; 10. Re: USA dial plan (Steven
Critchfield)
<DIV></DIV>>&nbsp;&nbsp; 11. soft fax machine
(j.m.jackson@thecompany.org)
<DIV></DIV>>&nbsp;&nbsp; 12. Re: USA dial plan (Eric
Wieling)
<DIV></DIV>>&nbsp;&nbsp; 13. Re: Screen Pop & Remote
Agents (Philipp von Klitzing)
<DIV></DIV>>&nbsp;&nbsp; 14. RE: USA dial plan
(ml@neoninternet.com)
<DIV></DIV>>&nbsp;&nbsp; 15. Re: USA dial plan (Chris
Albertson)
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 1 
<DIV></DIV>>Date: Fri, 09 Jan 2004 10:33:07 -0800 
<DIV></DIV>>From: Bob Knight <BK@MINUSW.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Subject: Re: [Asterisk-Users] Mailing list growth 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>Mark Spencer wrote: 
<DIV></DIV>> 
<DIV></DIV>> >I still think we need something more fine
grained.&nbsp;&nbsp;I think we can add the
<DIV></DIV>> >asterisk-biz list, and eventually something akin
to a newbie list, but
<DIV></DIV>> >need a more appropriate name, IMHO. 
<DIV></DIV>> > 
<DIV></DIV>>like an asterisk-virgin 
<DIV></DIV>>* for the very first time 
<DIV></DIV>> 
<DIV></DIV>>Now lets see how long it will take you to get that
tune out of your head.
<DIV></DIV>> 
<DIV></DIV>>-- 
<DIV></DIV>>Bob Knight 
<DIV></DIV>>[-w] the work option 
<DIV></DIV>>bk@minusw.com 
<DIV></DIV>>925-449-9163 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 2 
<DIV></DIV>>From: "Leopoldo Santiago"
<LSANTIAGO@PORTALAUTOMOTRIZ.COM>
<DIV></DIV>>To: <ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Cc: <ML@NEONINTERNET.COM>
<DIV></DIV>>Subject: RE: [Asterisk-Users] SIP and error talking
to voicemail
<DIV></DIV>>Date: Fri, 9 Jan 2004 12:37:29 -0600 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>> > 
<DIV></DIV>> > Late night. I've been to
http://www.grandstream.com/TEMP/FIRMWARE/&nbsp;&nbsp;I just
<DIV></DIV>>would like to find 1.0.4.17 so I know I'm not
introducing any new bugs if I
<DIV></DIV>>have to go back.&nbsp;&nbsp;I meant to say if
you know somewhere else to get 1.0.4.38.
<DIV></DIV>>I also tried just downloading it from my grandstream
but it didn't seem to
<DIV></DIV>>even want to try it -- probably the same
problem.&nbsp;&nbsp;I still get permission
<DIV></DIV>>denied when I try to TFTP manually
also.&nbsp;&nbsp;hmm...
<DIV></DIV>> > 
<DIV></DIV>> > If anyone has either of them, I'd
appreciate a copy!
<DIV></DIV>> > 
<DIV></DIV>> 
<DIV></DIV>>We have a copy of&nbsp;&nbsp;1.0.4.39. If you
want, you can get at:
<DIV></DIV>> 
<DIV></DIV>>http://www.supercomputo.com/b13p4.39.zip 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 3 
<DIV></DIV>>Date: Fri, 9 Jan 2004 10:46:56 -0800 (PST) 
<DIV></DIV>>From: Chris Albertson
<CHRISALBERTSON90278@YAHOO.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Subject: [Asterisk-Users] IConnect audio quality 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>Hello, 
<DIV></DIV>> 
<DIV></DIV>>I've subscribbed to
"IConnect".&nbsp;&nbsp;I use it eclusively for outbound
<DIV></DIV>>calling.&nbsp;&nbsp;I like the rates they
charge but people I call complain about
<DIV></DIV>>the audio quality.&nbsp;&nbsp;They say it
sounds like I'm using a "cheap mic." or
<DIV></DIV>>they 
<DIV></DIV>>complain about echo.&nbsp;&nbsp;The sound is
very clean at my end.&nbsp;&nbsp;I'm using
<DIV></DIV>>a Bundgtone phone with meadi routed through Asterisk
to IConnect.
<DIV></DIV>>It's not the BT100 phone as the Audio is OK in
cases where I don't
<DIV></DIV>>use Iconnect. 
<DIV></DIV>> 
<DIV></DIV>>Question: 
<DIV></DIV>>&nbsp;&nbsp; What service (other than
Iconnecthere) should I consider&nbsp;&nbsp;moving
<DIV></DIV>>&nbsp;&nbsp; to?&nbsp;&nbsp;Here is
what I want:
<DIV></DIV>> 
<DIV></DIV>>&nbsp;&nbsp; * Acceptable audio quality 
<DIV></DIV>>&nbsp;&nbsp; * Take outbound calls from
Asterisk to PSTN
<DIV></DIV>>&nbsp;&nbsp; * I do NOT need a did. (no need
for incomming calls)
<DIV></DIV>>&nbsp;&nbsp; * I prefer "pay as you
go" with little/no fixed monthly fee
<DIV></DIV>>&nbsp;&nbsp; * They need to provide call
termination to PSTN in the USA,
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp; Canada and
Japan
<DIV></DIV>>&nbsp;&nbsp; * Price maters but is secondary
after the above
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>===== 
<DIV></DIV>>Chris Albertson 
<DIV></DIV>>&nbsp;&nbsp; Home:&nbsp;&nbsp;
310-376-1029&nbsp;&nbsp;chrisalbertson90278@yahoo.com
<DIV></DIV>>&nbsp;&nbsp; Cell:&nbsp;&nbsp;
310-990-7550
<DIV></DIV>>&nbsp;&nbsp; Office:
310-336-5189&nbsp;&nbsp;Christopher.J.Albertson@aero.org
<DIV></DIV>>&nbsp;&nbsp; KG6OMK 
<DIV></DIV>> 
<DIV></DIV>>__________________________________ 
<DIV></DIV>>Do you Yahoo!? 
<DIV></DIV>>Yahoo! Hotjobs: Enter the "Signing Bonus"
Sweepstakes
<DIV></DIV>>http://hotjobs.sweepstakes.yahoo.com/signingbonus 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 4 
<DIV></DIV>>From: "Senad Jordanovic"
<SENAD@BOLTBLUE.COM>
<DIV></DIV>>To: <ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Date: Fri, 9 Jan 2004 18:50:24 -0000 
<DIV></DIV>>Subject: [Asterisk-Users] USA dial plan 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>Hi, 
<DIV></DIV>> 
<DIV></DIV>>Do the callers in USA dialling from USA Telco lines
always have to
<DIV></DIV>>prefix the CITY/AREA code with "1" in order
<DIV></DIV>>To successfully make a call to other USA
destinations?
<DIV></DIV>> 
<DIV></DIV>>---- 
<DIV></DIV>>I have not been to USA (yet) :) 
<DIV></DIV>> 
<DIV></DIV>>Ta 
<DIV></DIV>>SJ 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 5 
<DIV></DIV>>Date: Fri, 9 Jan 2004 19:50:15 +0100 
<DIV></DIV>>From:
=?ISO-8859-1?B?Um9iaW4gQ2FsbWVn5XJkIFNpdXJ1YQ==?= <ROBIN@ROCAS.SE>
<DIV></DIV>>Organization: RoCaS 
<DIV></DIV>>To: Tilghman Lesher
<ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Subject: Re[2]: [Asterisk-Users] Help with compiling 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>At 19:31:16, Tilghman Lesher wrote: 
<DIV></DIV>>TL> On Friday 09 January 2004 11:23, Robin
Calmegård Siurua wrote:
<DIV></DIV>> >> I have some problems when trying to install
Asterisk on Mandrake.
<DIV></DIV>> >> 
<DIV></DIV>> >> gcc -shared -Xlinker -x -o
pbx_gtkconsole.so pbx_gtkconsole.o
<DIV></DIV>> >> `gtk-config --libs gthread` 
<DIV></DIV>> >>
/usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackw
<DIV></DIV>> >>are-linux/bin/ld: cannot find -lXext
collect2: ld returned 1 exit
<DIV></DIV>> >> status 
<DIV></DIV>> >> make[1]: *** [pbx_gtkconsole.so] Error 1 
<DIV></DIV>> >> make[1]: Leaving directory
`/usr/src/asterisk/pbx'
<DIV></DIV>> >> make: *** [subdirs] Error 1 
<DIV></DIV>> >> 
<DIV></DIV>> >> 
<DIV></DIV>> >> What is this? :/ 
<DIV></DIV>> 
<DIV></DIV>>TL> Um, why do you have Slackware libraries
installed on a Mandrake
<DIV></DIV>>TL> machine? 
<DIV></DIV>> 
<DIV></DIV>>Hahaha. Sorry. It's Slackware, I'm just
extremly tired.
<DIV></DIV>> 
<DIV></DIV>>TL> The answer you're looking for is that you
don't have the X
<DIV></DIV>>TL> libraries installed, but you may have worse
problems.&nbsp;&nbsp;You really need
<DIV></DIV>>TL> to think about keeping your machine binaries
consistent.&nbsp;&nbsp;Things tend
<DIV></DIV>>TL> not to work correctly if you don't. 
<DIV></DIV>> 
<DIV></DIV>>This is not my machine. I'm just installing
Asterisk on it.
<DIV></DIV>> 
<DIV></DIV>>Thanks for your help, 
<DIV></DIV>> 
<DIV></DIV>>Robin 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--- 
<DIV></DIV>>&nbsp;&nbsp;RoCaS HB - development, hosting
and support
<DIV></DIV>>&nbsp;&nbsp;Robin Calmegård Siurua (CEO) -
robin@rocas.se
<DIV></DIV>>&nbsp;&nbsp;Flodins väg 44, 645 50 STRÄNGNÄS
SWEDEN
<DIV></DIV>>&nbsp;&nbsp;work:&nbsp;&nbsp; +46 8
50555680&nbsp;&nbsp;fax: +46 8 50555679
<DIV></DIV>>&nbsp;&nbsp;mobile: +46 73
6436805&nbsp;&nbsp;web: www.rocas.se
<DIV></DIV>>--- 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 6 
<DIV></DIV>>Subject: Re: [Asterisk-Users] * dialing before line
is open?
<DIV></DIV>>From: Steven Critchfield <CRITCH@BASESYS.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Date: Fri, 09 Jan 2004 12:57:59 -0600 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>On Fri, 2004-01-09 at 11:42, john lawler wrote: 
<DIV></DIV>> > Hi guys, 
<DIV></DIV>> > 
<DIV></DIV>> > I've had a sporadic problem recently with
one of my users on our POTS
<DIV></DIV>> > line.&nbsp;&nbsp;About 1/3 of the time
he dials a number (usually from a speeddial
<DIV></DIV>> > on his phone, I think), he'll get some
phone company message (from the
<DIV></DIV>> > outside) about how the call could not be
completed as dialed or
<DIV></DIV>> > something like that. 
<DIV></DIV>> > 
<DIV></DIV>> > However, the logs (and the console) always show
the correct dialed digits.
<DIV></DIV>> > 
<DIV></DIV>> > Anyone have a similar
symptom?&nbsp;&nbsp;I theorize that perhaps Asterisk picks
<DIV></DIV>> > up the line and starts dialing too quickly for
our service
<DIV></DIV>> > occasionally.&nbsp;&nbsp;Is there a way
that I could stick a delay before the
<DIV></DIV>> > digits are sent down the wire to test this
theory?&nbsp;&nbsp;I checked out the
<DIV></DIV>> > wiki but didn't see much pertaining to
this.&nbsp;&nbsp;I figure maybe you can put
<DIV></DIV>> > some control characters on the beginning of a
dial string to have it
<DIV></DIV>> > pause, but it doesn't seem to be
',' or 'p'.
<DIV></DIV>> 
<DIV></DIV>>p could be pulse, the character you are looking for
is w.
<DIV></DIV>>show application dial 
<DIV></DIV>> 
<DIV></DIV>>Also, asterisk will have to have received every digit
before it dials
<DIV></DIV>>the pots line. So you may wish to look at what
asterisk thought it was
<DIV></DIV>>dialing to determine if it is a DTMF detection error
at asterisk or at
<DIV></DIV>>the remote end. You should be able to watch the
console, or track back
<DIV></DIV>>the CDR logs to see what was dialed out by your user.
<DIV></DIV>>-- 
<DIV></DIV>>Steven
Critchfield&nbsp;&nbsp;<CRITCH@BASESYS.COM>
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 7 
<DIV></DIV>>Date: Fri, 9 Jan 2004 11:06:29 -0800 (PST) 
<DIV></DIV>>From: Sean Swallow <SEAN@SWALLOW.ORG>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Subject: Re: [Asterisk-Users] latest cvs == broken
tdmoe
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>Sorry for the reply to my own post. 
<DIV></DIV>> 
<DIV></DIV>>Aparently Gary has submited the fix to the digium bug
list. If you haven't
<DIV></DIV>>already, please check it out. =) 
<DIV></DIV>> 
<DIV></DIV>>Thanks, 
<DIV></DIV>> 
<DIV></DIV>>-- 
<DIV></DIV>>Sean Swallow 
<DIV></DIV>> 
<DIV></DIV>>On Fri, 9 Jan 2004, Sean Swallow wrote: 
<DIV></DIV>> 
<DIV></DIV>> > Thanks Mark... We've hacked it here to work
for now, but look forward to
<DIV></DIV>> > your CVS changes. 
<DIV></DIV>> > 
<DIV></DIV>> > 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 8 
<DIV></DIV>>Date: Fri, 09 Jan 2004 13:31:23 -0600 
<DIV></DIV>>From: john lawler <MAILLIST@TGICE.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Subject: [Asterisk-Users] zapbarge w/o the mute 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>I've got a couple of different situations where
I'd like to do something
<DIV></DIV>>like zapbarge into a specific channel but I'd
like to be able to
<DIV></DIV>>actually talk to the party or parties on the
channels, not just listen
<DIV></DIV>>like w/ zapbarge. 
<DIV></DIV>> 
<DIV></DIV>>There are two scenarios I can think of right now
where it'd be very handy.
<DIV></DIV>> 
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp; a) when the
outside line is ringing and Asterisk is waiting to
<DIV></DIV>>answer or Asterisk has just answered and is playing
some greeting or
<DIV></DIV>>w/e, I'd like to be able to dial an extension and
be connected to the
<DIV></DIV>>person [i.e., bridge the call] (and preferably have
the rest of the
<DIV></DIV>>dialplan exit as if they had dialed my extension). 
<DIV></DIV>> 
<DIV></DIV>>&nbsp;&nbsp;&nbsp;&nbsp; b)
there's a current conversation going on between two parties and
<DIV></DIV>>I'd like to be conferenced into it, but I'd
like to initiate the
<DIV></DIV>>conference, not have to wait for one of the parties
to three-way me in.
<DIV></DIV>> 
<DIV></DIV>>I'm sure there's already a way to do this, I
just haven't come across it
<DIV></DIV>>yet. 
<DIV></DIV>> 
<DIV></DIV>>Thanks guys, 
<DIV></DIV>> 
<DIV></DIV>>jl 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 9 
<DIV></DIV>>Date: Fri, 09 Jan 2004 19:40:02 +0000 
<DIV></DIV>>From: Iain Stevenson <IAIN@IAINSTEVENSON.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Subject: Re: [Asterisk-Users] Cisco Gear 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>Prices?&nbsp;&nbsp;Are we talking a 7960 for
the price of a SNOM?
<DIV></DIV>> 
<DIV></DIV>>&nbsp;&nbsp; Iain 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--On Friday, January 9, 2004 6:00 pm +0000 Adthrawn 
<DIV></DIV>><ADTHRAWN@ADTHRAWN.FREESERVE.CO.UK> wrote: 
<DIV></DIV>> 
<DIV></DIV>> > Hi, 
<DIV></DIV>> > 
<DIV></DIV>> > I know it's not really the place, but if
anybody in the UK (or US) is
<DIV></DIV>> > interested, I'm clearing out lots of new
Cisco stock...
<DIV></DIV>> > 
<DIV></DIV>> > 7970G's (colour LCD), 7960G's,
7940G's, 7920G's (wireless IP phone),
<DIV></DIV>> > 7935's (conference phone) and 3550-24-PWR
switches.
<DIV></DIV>> > 
<DIV></DIV>> > I also have boxes of 7914's, the
single-7914 foot stand and double-7914
<DIV></DIV>> > foot stand (these are required to connect a
7914 to a 7960G).
<DIV></DIV>> > 
<DIV></DIV>> > And some useful locking and non-locking
wallmount brackets for 79xx range.
<DIV></DIV>> > 
<DIV></DIV>> > We also have lots of PSU's for the whole
79xx range.
<DIV></DIV>> > 
<DIV></DIV>> > I'll now feel ashamed, and sink into my
seat :-)
<DIV></DIV>> > 
<DIV></DIV>> > Best, 
<DIV></DIV>> > Ad. 
<DIV></DIV>> > 
<DIV></DIV>> > _______________________________________________
<DIV></DIV>> > Asterisk-Users mailing list 
<DIV></DIV>> > Asterisk-Users@lists.digium.com 
<DIV></DIV>> >
http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > To UNSUBSCRIBE or update options visit: 
<DIV></DIV>>
>&nbsp;&nbsp;&nbsp;&nbsp;http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 10 
<DIV></DIV>>Subject: Re: [Asterisk-Users] USA dial plan 
<DIV></DIV>>From: Steven Critchfield <CRITCH@BASESYS.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Date: Fri, 09 Jan 2004 13:40:45 -0600 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>On Fri, 2004-01-09 at 12:50, Senad Jordanovic wrote: 
<DIV></DIV>> > Hi, 
<DIV></DIV>> > 
<DIV></DIV>> > Do the callers in USA dialling from USA Telco
lines always have to
<DIV></DIV>> > prefix the CITY/AREA code with "1" in
order
<DIV></DIV>> > To successfully make a call to other USA
destinations?
<DIV></DIV>> 
<DIV></DIV>>1 usually signifies a long distance call. It also is
prepended to all
<DIV></DIV>>calls using the full 10 digit areacode + exchange and
number.
<DIV></DIV>> 
<DIV></DIV>>So it is possible in the USA to dial a 7 digit number
without a 1
<DIV></DIV>>prepended for local calls in some areas. It is
possible for some 7 digit
<DIV></DIV>>numbers to be long distance(toll) and require a 1 to
be prepended to the
<DIV></DIV>>7 digits as a acceptance of the extra charge. All 10
digit dialing is
<DIV></DIV>>done with a 1 prepended. 
<DIV></DIV>>-- 
<DIV></DIV>>Steven
Critchfield&nbsp;&nbsp;<CRITCH@BASESYS.COM>
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 11 
<DIV></DIV>>Date: Fri, 9 Jan 2004 14:44:45 -0500 (EST) 
<DIV></DIV>>From: j.m.jackson@thecompany.org 
<DIV></DIV>>To: Asterisk-Users@lists.digium.com 
<DIV></DIV>>Subject: [Asterisk-Users] soft fax machine 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>Has anyone implemented a soft-fax within *? If not,
is there a SIP or IAX
<DIV></DIV>>or IAX2 client for * that would function as a
receive-only fax device?
<DIV></DIV>>Thanks! 
<DIV></DIV>> 
<DIV></DIV>>--Mike 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 12 
<DIV></DIV>>Subject: Re: [Asterisk-Users] USA dial plan 
<DIV></DIV>>From: Eric Wieling <ERIC@FNORDS.ORG>
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Organization: BTEL Consulting 
<DIV></DIV>>Date: Fri, 09 Jan 2004 13:52:41 -0600 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>Generally speaking, Yes. The usual dial plan in the
USA is as follows:
<DIV></DIV>> 
<DIV></DIV>>NXX-XXXX (Free Local Call to number in same Area
Code)
<DIV></DIV>>NXX-NXX-XXXX (Free Local Call to number in different
Area Code)
<DIV></DIV>>1-NXX-XXXX (Toll Call to number in same Area Code) 
<DIV></DIV>>1-NXX-NXX-XXXX (Toll Call to number in different Area
Code)
<DIV></DIV>>1-800-NXX-XXXX (Toll Free Call) 
<DIV></DIV>>1-855-NXX-XXXX (Toll Free Call) 
<DIV></DIV>>1-866-NXX-XXXX (Toll Free Call) 
<DIV></DIV>>1-877-NXX-XXXX (Toll Free Call) 
<DIV></DIV>>1-888-NXX-XXXX (Toll Free Call) 
<DIV></DIV>> 
<DIV></DIV>>Yes, in most places in the USA local calls are
totally free, no per min
<DIV></DIV>>charge. 
<DIV></DIV>> 
<DIV></DIV>>Some parts of the USA have "Local Toll
Calls", that is calls that are
<DIV></DIV>>dialed as NXX-XXXX that are not free, but have a very
small per min
<DIV></DIV>>cost.&nbsp;&nbsp;Los Angels is one of these
places I think.
<DIV></DIV>> 
<DIV></DIV>>On Fri, 2004-01-09 at 12:50, Senad Jordanovic wrote: 
<DIV></DIV>> > Hi, 
<DIV></DIV>> > 
<DIV></DIV>> > Do the callers in USA dialling from USA Telco
lines always have to
<DIV></DIV>> > prefix the CITY/AREA code with "1" in
order
<DIV></DIV>> > To successfully make a call to other USA
destinations?
<DIV></DIV>> > 
<DIV></DIV>> > ---- 
<DIV></DIV>> > I have not been to USA (yet) :) 
<DIV></DIV>> > 
<DIV></DIV>> > Ta 
<DIV></DIV>> > SJ 
<DIV></DIV>> > 
<DIV></DIV>> > 
<DIV></DIV>> > _______________________________________________
<DIV></DIV>> > Asterisk-Users mailing list 
<DIV></DIV>> > Asterisk-Users@lists.digium.com 
<DIV></DIV>> >
http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > To UNSUBSCRIBE or update options visit: 
<DIV></DIV>>
>&nbsp;&nbsp;&nbsp;&nbsp;http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>-- 
<DIV></DIV>>Go to
http://www.digium.com/index.php?menu=documentation and look at
<DIV></DIV>>the "Unofficial Links"
section.&nbsp;&nbsp;This section has links to a wide
<DIV></DIV>>variety of 3rd party Asterisk related
pages.&nbsp;&nbsp;My page is the
<DIV></DIV>>"Asterisk Resource Pages". 
<DIV></DIV>> 
<DIV></DIV>>BTEL Consulting 504-899-1387 or 850-484-4545 or
877-677-9643
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 13 
<DIV></DIV>>Date: Fri, 09 Jan 2004 20:39:46 +0100 
<DIV></DIV>>From: Philipp von Klitzing
<KLITZING@POOL.INFORMATIK.RWTH-AACHEN.DE>
<DIV></DIV>>Subject: Re: [Asterisk-Users] Screen Pop & Remote
Agents
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Organization: AEGEE 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>>Hi! 
<DIV></DIV>> 
<DIV></DIV>> >&nbsp;&nbsp;Im new to Asterisk and I
would like to get some imput from all of you.
<DIV></DIV>> > First I would like to start by telling all of
you I am starting a call
<DIV></DIV>> > center Within the next 2 weeks with 12 agents
to start.
<DIV></DIV>> 
<DIV></DIV>>Asterisk is probably a very good choice for you,
however you'll need more
<DIV></DIV>>time than those 2 weeks to be fully operational
(unless you get some
<DIV></DIV>>consultant to assist you). 
<DIV></DIV>> 
<DIV></DIV>> > has turned me on to Asterisk ;). Now The info I
need IS what cards do
<DIV></DIV>> > you think I should get? 
<DIV></DIV>> 
<DIV></DIV>>Before spending tons of $$$ the first step is to
install Asterisk and
<DIV></DIV>>test it with IP phones (soft- or hardware). Look at
queues and agents and
<DIV></DIV>>check out if they do what you need. Have a look at
GnoPhone if you need a
<DIV></DIV>>Linux based software phone that includes a web
browser and can take URL
<DIV></DIV>>arguments along with a call. 
<DIV></DIV>> 
<DIV></DIV>>http://www.voip-info.org/wiki-Asterisk+call+queues 
<DIV></DIV>>http://www.gnophone.com/ 
<DIV></DIV>> 
<DIV></DIV>> > pops for my reps through a web site? So when
the call is connected the
<DIV></DIV>> > yget all the coustomers info on screen? 
<DIV></DIV>> 
<DIV></DIV>>Use GnoPhone, or write your own little app (using PHP
or PERL etc) that
<DIV></DIV>>regularly reloads the call agent's browser (you
might be a bit too slow
<DIV></DIV>>that way, so it might be necessary for the agent to
initiate the reload
<DIV></DIV>>manually). If you prefer fance stuff then you could
use FLASH instead of
<DIV></DIV>>HTML with the help of PHP, ming and Actionscript. 
<DIV></DIV>> 
<DIV></DIV>> > For the second tricky part is can I have remote
agents? (reps working
<DIV></DIV>> > from home and log into the pbx by either Voip
or even Dialing in)?
<DIV></DIV>> 
<DIV></DIV>>Yes - the Wiki link above about "call
queues" has the info and links that
<DIV></DIV>>you need to look at. 
<DIV></DIV>> 
<DIV></DIV>> > as much as possible! as for some of the more
basic questions, can I
<DIV></DIV>> > put a .csv file in the sql DB and have it dial
from there?
<DIV></DIV>> 
<DIV></DIV>>Take a look at sample.call in your Asterisk
installation. See also:
<DIV></DIV>>http://www.voip-info.org/wiki-Asterisk+auto-dial+out 
<DIV></DIV>> 
<DIV></DIV>> > and will I be able to set a Dial Plan to only
call certin area codes?
<DIV></DIV>> 
<DIV></DIV>>You mean you want to dial random numbers within a
certain area code?
<DIV></DIV>>I wouldn't consider that to be a nice way of
doing things...
<DIV></DIV>> 
<DIV></DIV>> > reason I ask all this is because all of these
over priced dia rs do
<DIV></DIV>> > just that. Also can Asterisk be set with the
FTC laws to 3% droped
<DIV></DIV>> > call ratio? 
<DIV></DIV>> 
<DIV></DIV>>Explain? 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>Cheers, Philipp 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 14 
<DIV></DIV>>Date: Fri,&nbsp;&nbsp;9 Jan 2004 12:55:39
-0700
<DIV></DIV>>From: ml@neoninternet.com 
<DIV></DIV>>Subject: RE: [Asterisk-Users] USA dial plan 
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>> > Hi, 
<DIV></DIV>> > 
<DIV></DIV>> > Do the callers in USA dialling from USA Telco
lines always have to
<DIV></DIV>> > prefix the CITY/AREA code with "1" in
order
<DIV></DIV>> > To successfully make a call to other USA
destinations?
<DIV></DIV>> > 
<DIV></DIV>> > ---- 
<DIV></DIV>> > I have not been to USA (yet) :) 
<DIV></DIV>> > 
<DIV></DIV>> > Ta 
<DIV></DIV>> > SJ 
<DIV></DIV>> 
<DIV></DIV>>In all cases of long distance, 1 plus the area code
is used.&nbsp;&nbsp;In small areas where local only is involved you
usually only dial 7 digits.&nbsp;&nbsp;In metro areas with multiple area
codes, you use 10 digit dialing.&nbsp;&nbsp;Some places you use 10 digit
dialing or 1 + area code, depends on the phone
company.&nbsp;&nbsp;&nbsp;&nbsp;I've seen this happen on the
east coast.
<DIV></DIV>> 
<DIV></DIV>>Kevin 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>Message: 15 
<DIV></DIV>>Date: Fri, 9 Jan 2004 12:01:05 -0800 (PST) 
<DIV></DIV>>From: Chris Albertson
<CHRISALBERTSON90278@YAHOO.COM>
<DIV></DIV>>Subject: Re: [Asterisk-Users] USA dial plan 
<DIV></DIV>>To: asterisk-users@lists.digium.com 
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--- Senad Jordanovic <SENAD@BOLTBLUE.COM>wrote:
<DIV></DIV>> > Hi, 
<DIV></DIV>> > 
<DIV></DIV>> > Do the callers in USA dialling from USA Telco
lines always have to
<DIV></DIV>> > prefix the CITY/AREA code with "1" in
order
<DIV></DIV>> > To successfully make a call to other USA
destinations?
<DIV></DIV>> 
<DIV></DIV>>Not "always".&nbsp;&nbsp;My local
phone company (Verizon in So. California)
<DIV></DIV>>said we'd have 
<DIV></DIV>>to dail 1-310-xxx-xxxx for local numbers (I'm in
310 area) but
<DIV></DIV>>as of current time I can still dail just 7 digits and
it works
<DIV></DIV>>fine. 
<DIV></DIV>> 
<DIV></DIV>>I think over time, yes, we are headed to the
requirement to
<DIV></DIV>>alwauys dial 1-<10 digit number> 
<DIV></DIV>> > 
<DIV></DIV>> > ---- 
<DIV></DIV>> > I have not been to USA (yet) :) 
<DIV></DIV>> > 
<DIV></DIV>> > Ta 
<DIV></DIV>> > SJ 
<DIV></DIV>> > 
<DIV></DIV>> > 
<DIV></DIV>> > _______________________________________________
<DIV></DIV>> > Asterisk-Users mailing list 
<DIV></DIV>> > Asterisk-Users@lists.digium.com 
<DIV></DIV>> >
http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > To UNSUBSCRIBE or update options visit: 
<DIV></DIV>>
>&nbsp;&nbsp;&nbsp;&nbsp;http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>===== 
<DIV></DIV>>Chris Albertson 
<DIV></DIV>>&nbsp;&nbsp; Home:&nbsp;&nbsp;
310-376-1029&nbsp;&nbsp;chrisalbertson90278@yahoo.com
<DIV></DIV>>&nbsp;&nbsp; Cell:&nbsp;&nbsp;
310-990-7550
<DIV></DIV>>&nbsp;&nbsp; Office:
310-336-5189&nbsp;&nbsp;Christopher.J.Albertson@aero.org
<DIV></DIV>>&nbsp;&nbsp; KG6OMK 
<DIV></DIV>> 
<DIV></DIV>>__________________________________ 
<DIV></DIV>>Do you Yahoo!? 
<DIV></DIV>>Yahoo! Hotjobs: Enter the "Signing Bonus"
Sweepstakes
<DIV></DIV>>http://hotjobs.sweepstakes.yahoo.com/signingbonus 
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>--__--__-- 
<DIV></DIV>> 
<DIV></DIV>>_______________________________________________ 
<DIV></DIV>>Asterisk-Users mailing list 
<DIV></DIV>>Asterisk-Users@lists.digium.com 
<DIV></DIV>>http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> 
<DIV></DIV>> 
<DIV></DIV>>End of Asterisk-Users Digest 
<DIV></DIV></div><br clear=all><hr> <a
href="http://g.msn.com/8HMAENUS/2749??PS=">Make your home warm and
cozy this winter with tips from MSN House & Home. </a> </html>

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