n Behalf Of
*Nile Kaledon
*Sent:* Monday, October 25, 2010 12:06 PM
*To:* asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
*Subject:* Re: [asterisk-users] Dial plan help
Hi Jigar,
I use visual dialplan too. Nice tool.
Here you can find
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, October 25, 2010 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan help
Un
Un-top-posting...
> On 2010-10-25 2:01 PM, "Jigar Joshi" wrote:
>
> Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat
> are the chapters I should cover for my requirement.
> Because Its too long book :P
On Mon, 25 Oct 2010, Zeeshan Zakaria wrote:
> Chapte
igium.com] On Behalf Of Zeeshan
Zakaria
>Sent: Monday, October 25, 2010 1:07 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] Dial plan help
>Chapters 4, 5 and 6 is a good start.
>Zeeshan A Zakaria
Specifically, read PP 100-
Chapters 4, 5 and 6 is a good start.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-25 2:01 PM, "Jigar Joshi" wrote:
Ok Thanks Guys.
Can you guyz suggest me upto which chapters orwhat are the chapters I should
cover for my requirement.
Because Its too long book :P
igium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nile Kaledon
> *Sent:* Monday, October 25, 2010 12:06 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] Dial plan help
>
>
>
> Hi Jigar,
>
>
>
> I use visual d
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon
Sent: Monday, October 25, 2010 12:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dial plan help
Hi Jigar,
I use visual dialplan too
Hi Jigar,
I use visual dialplan too. Nice tool.
Here you can find some dial plan examples and tutorials that may help you:
codezone.apstel.com
Nile
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New t
I totally agree with Steve's wise advice. One should at least give himself a
week learning asterisk fundamentals and related Linux basics before jumping
into creating dialplans or setting up Telecom systems. Asterisk's official
book's first few chapters cover all the basics which every asterisk use
Hi Jigar
> I am facing issue while generating a dial plan for the following case:
> all caller should be asked a code to enter than All the callers should be
connected one extension.
Try DISA component, and then use MeetMe component if you want callers to go
to conference or Dial component if you
> On Mon, 18 Oct 2010, Jigar Joshi wrote:
>
>> @Gilles here are my requirement.can you please help me .
On Mon, 18 Oct 2010, Steve Edwards wrote:
> Are you putting this "out to bid" or are you just too lazy to read ATFOT
> (http://downloads.oreilly.com/books/9780596510480.pdf)?
On Sat, 23 Oct 2
Jigar Joshi wrote:
>
> Currently I have created a dial plan using vdp I tried submitting it
> here but I don't know how to extract text version for the same .
>
After Googling a bit, I found that VDP is Visual Dial Plan for
Asterisk. Neat little application, but I doubt you'll find many if any
Steve & Alex thanks for your help. I've got it working perfectly now.
-Jon
- Original Message -
From: "Alex Balashov" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, August 24, 2008 9:22 AM
Subject:
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
-
On Sun, Aug 24, 2008 at 8:11 AM, Jon Weisman <[EMAIL PROTECTED]> wrote:
> I'd like to do the following can someone guide me on how to accomplish this?
>
>
> Call comes in via PRI and tries to go out via SIP if for some reason the ISP
> is down and the call can not go out i want it to fail over and
This is some pretty basic stuff... (someone will probably send you a RTFM)
Start with the sample dialplan (make samples I think)...trace the dialplan
along to understand how it works
Check the wiki and then post anything that you need help with
From: [EMAIL PROTECTED]
[mailto:[EMAIL PR
>So how do we set it up if I'm out of the office, or on the mobile phone and
>can't answer the call.
>How does it know to go to voice mail?
You set it to ring for a certain duration then go to voicemail after n seconds.
You'll want an incoming call to go to a context at which point you can start
A DigitTimeout(3) will do wonders to (and fix the non existing priorities).
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens [EMAIL PROTECTED]
Verzonden: vrijdag 3 december 2004 21:52
Aan: Asterisk Users Mailing List - Non-Commer
On Fri, 2004-12-03 at 16:12 -0500, [EMAIL PROTECTED] wrote:
> Thanks Luki,
>
> I can't believe I overlooked that. It's working now.
>
> Steven,
> It wasn't working because I overlooked the priority. However is it still a
> bad idea to have ivr and extensions beginning with the same number or wit
Jon,
* scans through the valid extension in the context every time the user enter
a digit. If you only have single digit extensions, dialing 2 is definitive
and * can jump to that extension without waiting for further digits. But if
you have 2 and 200 defined, a single 2 is ambiguous and asterisk
, you get my extension.
Watch that digittimeout..it helps.
-Matthew
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, December 03, 2004 2:52 PM
s and they have been
printed, so it won't be possible to change now.
Thanks,
Jon
- Original Message -
From: "Luki" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, December 03,
> exten=>200,Goto(office,102,1);forward to 102 in office context
> exten=>201,Goto(office,110,1);forward to 110 in office context
These are invalid -- no priority -- and hence dropped. Didn't you see the
errors while loading (it's easy to miss, there's plenty of stuff output).
Change to:
exten=>
On Fri, 2004-12-03 at 15:52 -0500, [EMAIL PROTECTED] wrote:
> All,
>
> I've got a problem here. We are using a Digium 4 T-1 board in our * server.
> The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
> that when someone dials our DID it goes to the s extension and starts
I would use:
exten => _NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _NXXNXX,3,Congestion
exten => _NXXNXX,102,1,Busy
exten => _NXXNXX,103,1,Busy
That way if number you dial is busy it will not immediately try dialing
the same
exten => _NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _NXXNXX,3,Congestion
The above will attempt to dial out your Zap interface first. If that
fails, it will dial out using "username" for the username and the
password, IP address
Checkout http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
and http://www.voip-info.org/wiki-Asterisk+t+extension
You could use extention t, which is reached after dial times out.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon Brow
27 matches
Mail list logo
