5)Regards,-- Anthony RodgersBusiness Systems AnalystDistrict of North VancouverWeb: http://www.dnv.orgRSS Feed: http://www.dnv.org/rss.aspOn Sep 14, 2005,
at 9:15
AM, PJ Santos wrote:> I need create one configuration to provide one Interactive Voice > Response.> > I read any doc
rovide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to.
On 9/13/05, PJ Santos <[EMAIL PROTECTED]> wrote:
Hi All,
I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press
Hi all,
Somebody already carried through the integration enters the DAC Nortel S1 with ITG and Asterisk?
Regards.
PJSantos
Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe!___
--Bandwidth and Colocation sponso
Hi All,
I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator.
What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP.
I will try with Oracle, Have somebody work with it?
Regards
Jsalas.
-Mensaje original-----De: PJ Santos [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, September 07, 2005 6:31 PMPara: Asterisk-Users@lists.digium.comAsunto: [Asterisk-Users] IVR Documentation an Sample.
Hi everybody.
Hi everybody.
I need documentation and sample, about IVR.
Sample about database access with IVR+Asterisk, if its possible.
Thanks.
Paulo Santos
Brasil/RJ.__Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.
A while back, I found a site that had the entire asterisk-users
mailing list archive in mbox format. Does anybody know if and where
such a thing is availible?
PJ
--
All men know the utility of useful things;
but they do not know the utility of futility.
-- Chuang-tzu
On Mon, 19 Jan 2004, David Gomillion wrote:
> Andrew wrote:
>
> First, what's wrong with PoE? Is it any worse than installing tons of
> channel banks?
Can anybody recommend a good PoE product? I am interested in getting
that implemented.
PJ
--
Wisdom is not a product of
On Sat, Dec 06, 2003 at 11:14:40PM +, Tristan 'Minty' Colgate wrote:
> Hi,
>
> My apologies for those on the channel who may take offense to this, atleast
> the ones to whom it was not aimed, but the fact is that after making a simple
> enquiry on the IRC channel I was in absolute shock...
.
On Wed, Dec 03, 2003 at 10:42:40PM -0500, TeleSIP wrote:
> A good rootkit will also modify the date and time of the replaced binaries
> so they will look the same as the original.
>
> Try to replace your "ps" command with that from a trusted RH9 machine. If
> it works ok then you must do a clean
It is a shame that within a couple of hours they can tell you to remove helpfull
documentation, but not (seemingly) help answer questions regarding there Cisco stuff
on this list. I think Cisco must have their priorities mixed up!
Just my opinion... which also means I won't support a company lik
On Wed, Nov 26, 2003 at 01:55:36PM -0500, Clif Jones wrote:
> Thanks for the truly useful feedback. I'm having a real hard time with
> the FAQ pages listing
> RH 8 & 9 FIRST in the list of Linux distros that Asterisk compiles and
> runs on and having
> any bugs (oh I mean RH problems) discarded.
On Fri, Nov 14, 2003 at 01:08:50PM -0800, Steve Sobol wrote:
> Senad Jordanovic wrote:
>
> > Funny, I am doing the same at the moment... :)
> >
> > We are allowing * to dump call records onto a remote database server.
> > Once there we can do all sort of things with it.
> >
> > My only concern
On Wed, Nov 12, 2003 at 09:22:56AM -0800, Chris Albertson wrote:
...
> There are several other very good free SQL DBMSes. One of them
> is actually supported by one of the world's largest software
> companies, SAP.
>
> SAP and MySQL signed an agreement where MySQL will co-market SAPDB
> and the
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote:
>
> I second that, and I think I remember hearing Mark talking about it too. But.
>
> What type of encryption can you do that does not introduce latency?
>
> That said, I would like it to support hardware encryption cards.
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Dan
> Sent: 03 November 2003 19:18
> To: Asterisk Users
> Subject: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for
> downlaod...
>
> as promise, at:
>
> http://www.laser.com/dante
> or
> http://www.geocities.com/tdanro
On Tue, Nov 04, 2003 at 02:49:39AM +0100, Christopher Arnold wrote:
> On Tue, 4 Nov 2003, Shoval Tom wrote:
>
> > Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many
> > other places (like our branch office, my home dial-up account, my parents
> > dial-up account)
> >
> Do y
I have not found the type of license you are using for this demo. Can you please
confirm which one you plan to use for this.
On Sun, Oct 19, 2003 at 05:16:29AM -0700, Azher Amin wrote:
>
> Thnx for the interest in the ivr sample [btw: I am not an expert in PERL/AGI :) ]
> comments are welocome
still accept US $'s and know the implecations, I think we
can work something out.
I can provide additional info to interested people
email to: pj at cassens*dot*com I will reply as soon as possible.
Again, I need a quick turnaround! Skills in * + database + AGI are likely mand
On Tue, Oct 07, 2003 at 04:40:36AM -0500, Steven Critchfield wrote:
>
> What point do you feel that a user is too advanced to us your wizard, or
> at what point do you think a user of your wizard will be more pissed at
> being hindered by the product than helped?
>
> I'm not trying to insult you,
on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote:
> This probably has an easy solution, but I found it yet. How can I get out
> of a remote console after using ssh to get into the box, making changes,
> reload etc. without stopping *?
>
> Thanks in advance.
Looks like "exit" will rele
On Thu, Oct 02, 2003 at 09:58:37AM +1000, Jamie Carl wrote:
...
> As for the rest of this discussion, I have already started
> work on this Asterisk Web Interface. (visit
> http://astweb.sourceforge.net). The current release is
> still only the CDR section, but things are starting to
> evolve
Comments inline:
On Tue, Sep 30, 2003 at 07:38:07AM +0100, WipeOut wrote:
> Mark Evans wrote:
>
> >>I think we're getting away from the original purpose of this program.
> >>Are people really that desparate for a full, web-based admin/user
> >>interface?
> >>
> >>
> >
> >I sure am, I want to
On Mon, Sep 29, 2003 at 11:09:06AM +0100, WipeOut wrote:
>
> >I was thinking of using
> >
> >http://developer.berlios.de/
> >
> >As SF has had many problems recently :(
> >
> >Regards
> >
> >Mark
> >
> >
> >
> Yea, I have noticed Sourceforge has been a little flaky lately.. Thought
> they would
On Fri, Sep 26, 2003 at 06:40:12AM -0700, TC wrote:
> Welcome
> I have been updating this doc with links to user documenation
> as i come across it
> http://bugs.digium.com/bug_view_page.php?bug_id=070
ERROR: Access Denied.
as user anonymous... guess I need to create an account.
On Tue, Sep 23, 2003 at 08:54:50AM -0400, costas wrote:
> Hi,
>
> I am an experienced developer with Windows and familiar with Linux. I am looking for
> a SIP solution.
>
> 1) How does Asterisk compare to VOCAL in terms of support.
Sorry, don't know
>
> 2) Is Asterisk free?
yes
>
> 3) Wh
On Mon, Sep 22, 2003 at 04:33:44PM -0400, Ariel Batista wrote:
>
> I am very interested in why your no longer using these phones. We have one for
> testing and so far it's not working. Are they not working? Is the main problem
> configuration? Any information will be very helpful.
>
There we
What am I if I use mutt...besides virus free? ;)
On Mon, Sep 22, 2003 at 04:30:49PM -0400, Sean Heiney wrote:
> Actually, MS Outlook by default blocks all executables. I'm not sure why
> there is so much negativity around the Outlook client. Perhaps we
> should all go back to the cave and use Pin
Don't know yet if it helps, but if you read the link at:
http://www.voip-info.org/tiki-index.php?page=NAT+and+VOIP
it will point you to:
http://www.sipcenter.com/files/SIPNATtraversal.pdf
However has the voip-info.org site; your stuff ROCKS!!
On Fri, Sep 19, 2003 at 03:11:31PM -0500, C. Johnso
I am about to try our TDM400P E model from the developer kit (not the "Lite") we just
got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so
the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as
"Found a Wildcard FXS: Wildcard S400P
Does this thread help?
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote:
> I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> overseas IP connection, and somehow SIP seemed to work better.
>
On Thu, Sep 18, 2003 at 09:22:55PM -0600, John Brown wrote:
> HI folks, nice conversation, but it has *nothing* to do with
> the subject line.
Sorry, I tried twice (forgot once and did a second time). Someone else also tried and
now you. End result... I think this may be beaten to death... for no
On Thu, Sep 18, 2003 at 06:09:27PM -0400, Steve Creel wrote:
> I am NOT a VoIP guru. I am NOT an Asterisk guru. I am NOT a telephony
> guru. Take that as a disclaimer for the information below, as well as to
> say that the best learning comes from reading anything you can get your
> hands on. T
I expect a user list to be for users' questions. I expect a user list to support that
what it's a list for. In return *I* should help someone when/if I can! There is no
"for Nothing". You help me, then I help some newbie 10 years from now when I
understand this stuff. So, in the meantime, my onl
Sorry about changing the original incorrect subject of "Re: [Asterisk-Users]
Grandstream Source?" . Many have already written that thread off and this may be a
good place to start on a positive note.
Yes, I forgot to mention some of the sites that I have found usefull. I do have to say
that htt
I have to defend us newbies on this.
This environment does not facilitate sequential knowledge building! Based on my entry
to Asterisk, I should have already known
T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea
(still trying to figure out "skinny"...cisco som
You guys are a tough crowd. I do have to admit I did "get" this one, however.
I don't know about Senad, but this is not an easy list to pick up on. In order to
search the list, you have to know the terms/acronyms. In order to know the terms, you
have to learn/ask. Many of you know this stuff ba
Yes, Please share.
On Tue, Sep 16, 2003 at 03:05:33PM -0400, Yifang Dai wrote:
> On Tue, Sep 16, 2003 at 03:27:44PM -0300, Paulo Mannheimer wrote:
> > Hi Rich,
> >
> > We have done this before. We basically developed a small client that
> > sits on every machine and communicates with * to get inf
Looks *exactly* like the X100P (X101P) card I just got from digium last week...
On Mon, Sep 15, 2003 at 11:03:14AM -0400, Daryl G. Jurbala wrote:
> >
> > "These cards are replicas of the X100P sold for use in an Asterisk (
> > www.asterisk.org) phone system. They are fully functional and
> > wo
Hello, trying to get gnophone to fully function under RH9 and finally have the browser
working from the rpm from 2001. Turns out I had to get an old Mozilla 1.2 version and
alter the startup gnophone script to point to it (MOZILLA_FIVE_HOME and
LD_LIBRARY_PATH).
How much better is the CVS stuff
Yes, this is RH9. Thank you for the info.
On Fri, Sep 12, 2003 at 02:59:46PM -0700, Scott Stingel wrote:
> If you're running RedHat 9, there is a known problem.
>
> Try executing the following line in the shell before starting asterisk:
>
> export LD_ASSUME_KERNEL=2.4.1
>
> Hope this works!
>
Trying to figure out why I'm having all of my test (and demo) perl script in a defunct
status. Each run creates a problem:
ps output
root 26253 1356 0 16:39 pts/100:00:00 asterisk -vvvc
root 26270 26253 0 16:40 pts/100:00:00 [pj.pl ]
root 26271 26253 0 16:40 pts/100:00
Oops should have looked a little harder to find the "say digits".
Sorry.
On Fri, Sep 12, 2003 at 10:21:32AM -0500, PJ Welsh wrote:
> I searched for "say number" in the * google archives and have not found reference to
> options for "say number". I would like
I searched for "say number" in the * google archives and have not found reference to
options for "say number". I would like to have * say digits instead of the hundreds
and thousands. EG, "1234" would say one two three four.
___
Asterisk-Users mailing l
u would want to do a project pricing.
Again, for someone with the experience, I do not believe this to be hard. We need help
to prove we can internalize this project or loose it to the outsource/blackbox/money
people. I do not want to give in to the dark side...
Thank you
Please email: pj at
Nice goin'!! I will use this for a reference point to establish baseline numbers of
phone lines. The good news is that the equation is not linear (eg 45 people need 5
lines, 100 need 10 lines). So I can double my potential users and ONLY need 2 more
lines (qty 7). The bad news is that I don't 10
Top posting only:
This is great info. A couple of you have already replied with very helpfull and
usefull information. Thank you very much!!
I am very excited to hear that I can test without purchasing the hardware. I googled
and found a IAXClient at http://iaxclient.sourceforge.net/. Is that t
Hello all:
Thank you for taking the time to read this post.
Background:
I am a new user to IVR systems and asterisk. I have been tasked with helping to set up
a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so
people that will call in 3 or 4 time each
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