<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>> 1. Re: Mailing list growth (Bob Knight)
<DIV></DIV>> 2. RE: SIP and error talking to voicemail (Leopoldo Santiago)
<DIV></DIV>> 3. IConnect audio quality (Chris Albertson)
<DIV></DIV>> 4. USA dial plan (Senad Jordanovic)
<DIV></DIV>> 5. Re[2]: [Asterisk-Users] Help with compiling (=?ISO-8859-1?B?Um9iaW4gQ2FsbWVn5XJkIFNpdXJ1YQ==?=)
<DIV></DIV>> 6. Re: * dialing before line is open? (Steven Critchfield)
<DIV></DIV>> 7. Re: latest cvs == broken tdmoe (Sean Swallow)
<DIV></DIV>> 8. zapbarge w/o the mute (john lawler)
<DIV></DIV>> 9. Re: Cisco Gear (Iain Stevenson)
<DIV></DIV>> 10. Re: USA dial plan (Steven Critchfield)
<DIV></DIV>> 11. soft fax machine (j.m.jackson@thecompany.org)
<DIV></DIV>> 12. Re: USA dial plan (Eric Wieling)
<DIV></DIV>> 13. Re: Screen Pop & Remote Agents (Philipp von Klitzing)
<DIV></DIV>> 14. RE: USA dial plan (ml@neoninternet.com)
<DIV></DIV>> 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. 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/ 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. 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. I still get permission
<DIV></DIV>>denied when I try to TFTP manually also. 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 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". I use it eclusively for outbound
<DIV></DIV>>calling. I like the rates they charge but people I call complain about
<DIV></DIV>>the audio quality. They say it sounds like I'm using a "cheap mic." or
<DIV></DIV>>they
<DIV></DIV>>complain about echo. The sound is very clean at my end. 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>> What service (other than Iconnecthere) should I consider moving
<DIV></DIV>> to? Here is what I want:
<DIV></DIV>>
<DIV></DIV>> * Acceptable audio quality
<DIV></DIV>> * Take outbound calls from Asterisk to PSTN
<DIV></DIV>> * I do NOT need a did. (no need for incomming calls)
<DIV></DIV>> * I prefer "pay as you go" with little/no fixed monthly fee
<DIV></DIV>> * They need to provide call termination to PSTN in the USA,
<DIV></DIV>> Canada and Japan
<DIV></DIV>> * Price maters but is secondary after the above
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>=====
<DIV></DIV>>Chris Albertson
<DIV></DIV>> Home: 310-376-1029 chrisalbertson90278@yahoo.com
<DIV></DIV>> Cell: 310-990-7550
<DIV></DIV>> Office: 310-336-5189 Christopher.J.Albertson@aero.org
<DIV></DIV>> 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. You really need
<DIV></DIV>>TL> to think about keeping your machine binaries consistent. 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>> RoCaS HB - development, hosting and support
<DIV></DIV>> Robin Calmegård Siurua (CEO) - robin@rocas.se
<DIV></DIV>> Flodins väg 44, 645 50 STRÄNGNÄS SWEDEN
<DIV></DIV>> work: +46 8 50555680 fax: +46 8 50555679
<DIV></DIV>> mobile: +46 73 6436805 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. 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? I theorize that perhaps Asterisk picks
<DIV></DIV>> > up the line and starts dialing too quickly for our service
<DIV></DIV>> > occasionally. Is there a way that I could stick a delay before the
<DIV></DIV>> > digits are sent down the wire to test this theory? I checked out the
<DIV></DIV>> > wiki but didn't see much pertaining to this. 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 <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>> 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>> 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? Are we talking a 7960 for the price of a SNOM?
<DIV></DIV>>
<DIV></DIV>> 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>> > 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 <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. 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>> > 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. This section has links to a wide
<DIV></DIV>>variety of 3rd party Asterisk related pages. 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>> > 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, 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. In small areas where local only is involved you usually only dial 7 digits. In metro areas with multiple area codes, you use 10 digit dialing. Some places you use 10 digit dialing or 1 + area code, depends on the phone company. 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". 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>> > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>=====
<DIV></DIV>>Chris Albertson
<DIV></DIV>> Home: 310-376-1029 chrisalbertson90278@yahoo.com
<DIV></DIV>> Cell: 310-990-7550
<DIV></DIV>> Office: 310-336-5189 Christopher.J.Albertson@aero.org
<DIV></DIV>> 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>>_______________________________________________
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<DIV></DIV>>http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>End of Asterisk-Users Digest
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