First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one
setup
but during that time the caller has had two rings before the local
extension
has even begun to ring. Is there something I am doing wrong
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to ring. Is there something I am doing wrong that
On Wed, 15 Aug 2007, Matthew Harrell wrote:
The intent of this sequence is to take the incoming callerid, replace it if
known with something in the database, and branch on the state from the DB
and time of the day.
FWIW: I do something similar, but purely in dial-plan using the astdb -
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to ring. Is there something I am doing wrong that
: Check out the CDR configuration. I do my CDR via MySQL and I don't think
: that does buffering, but I know for sure the normal CSV format (and standard
: configuration file) has options for buffering before saving. I can't really
: think how that would change recieving the CDR information
Wait(2) is what I do.
Matthew Harrell wrote:
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to
Thanks. I was hoping there might be a way to detect whether the CID
routine was done or not. I've still seen occasions where it wasn't available
for callers that I know had it. Maybe my phone service is just a little
slow sometimes
Wait(2) is what I do.
Matthew Harrell wrote:
First, it
85 10 Congestion
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Anselm
Martin Hoffmeister
Enviado el: domingo, 12 de agosto de 2007 12:35
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Dialplan loop
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
On 8/10/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Aug 09, 2007 at 08:12:12PM -0500, David Bandel wrote:
[snip]
...
exten = s,n,Set(loop = $[${loop} + 1])
exten = s,n,Set(loop=$[${loop} + 1])
Thanx Tzafrir. It works now. I guess the documentation I read that
said that white
On Thu, Aug 09, 2007 at 08:12:12PM -0500, David Bandel wrote:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
Am
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
Am using 1.4.10 and have reviewed doc/
exten = s,1,Set(TIMEOUT(digit)=5)
Ok let's forget the asterisk dialplans for a second. Maybe I am
misunderstanding the 'national' option. When I set the
dialplan=national for the zaptel config what exactly does that do? I
was under the impression that re-wrote numbers.
On 7/24/07, Anselm Martin Hoffmeister [EMAIL PROTECTED]
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, July 23, 2007 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialplan
Hi,
What dialplan option do I need to send a call out like this:
NPA-NXX- local
Am Montag, den 23.07.2007, 14:33 -0400 schrieb Matt:
Hi,
What dialplan option do I need to send a call out like this:
NPA-NXX- local calls
1-NPA-NXX- - long distance
Won't 'national' send it out NPA-NXX- no matter if it's long
distance or not?
I do not understand your point
Hi,
What dialplan option do I need to send a call out like this:
NPA-NXX- local calls
1-NPA-NXX- - long distance
Won't 'national' send it out NPA-NXX- no matter if it's long
distance or not?
___
--Bandwidth and Colocation Provided by
or not for long distance.
D.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, July 23, 2007 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialplan
Hi,
What dialplan option do I need
Hi,
I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for
outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed
version ) to this version and in my opinion a lot more troubles arose
For outgoing calls I use a Digium B410P with chan_misdn
hi list,
I'm looking for a way to execute commands in my dialplan specifically when a
caller has hung up. my curretn dialplan looks like this:
exten = s,1,Answer
exten = s,n(restart),BackGround(intro)
exten = s,n,Read(Enter,4,4)
exten = s,n,Voicemail(${Enter},u)
exten =
On 4/26/07, Michael Kamleitner [EMAIL PROTECTED] wrote:
On 4/26/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Michael Kamleitner [EMAIL PROTECTED]
Date: Wed, 25 Apr 2007 17:47:34 +0200
however, I've continued to experiment again and again, and strangely it
seemed to work _some_ times,
On 4/26/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Michael Kamleitner [EMAIL PROTECTED]
Date: Wed, 25 Apr 2007 17:47:34 +0200
however, I've continued to experiment again and again, and strangely it
seemed to work _some_ times, even when passing 4digit-extensions. now I
think I got the
hi community,
I'm new to this list asterisk in general, so let me first say thx to
everybody involved in providing such great tools ressources!!
I'm currently trying to implement a simple voicebox-system.
for demonstration purposes, I've successfully connected my cellphone via
bluetooth using
On Wed, Apr 25, 2007 at 01:21:40PM +0200, Michael Kamleitner wrote:
hi community,
I'm new to this list asterisk in general, so let me first say thx to
everybody involved in providing such great tools ressources!!
I'm currently trying to implement a simple voicebox-system.
for
Try moving 2 digit extensions before single digit. I believe asterisk
matches the first found extension which is always the single digit
extensions the way you have it
Bart
Michael Kamleitner wrote:
hi community,
I'm new to this list asterisk in general, so let me first say thx to
On 4/25/07, Barton Fisher [EMAIL PROTECTED] wrote:
Michael Kamleitner wrote:
I'm currently trying to implement a simple voicebox-system.
for demonstration purposes, I've successfully connected my cellphone
via bluetooth using the current chan_cellphone-patch on the current
SVN-version of
thx for all of your suggestions... I'm learning more about asterisk every
minute :)
Barton, I tried to replace 'WaitExten' with 'Background' as you suggested,
and at first was disappointed that didn't change the behavior.
Than I tried Roberts suggestion, using 'Read' instead of 'WaitExten' -
From: Steve Davies [EMAIL PROTECTED]
Date: Wed, 25 Apr 2007 15:58:25 +0100
On 4/25/07, Barton Fisher [EMAIL PROTECTED] wrote:
Michael Kamleitner wrote:
[good stuff sniffed]
A very simple workaround to achieve what you want might be to replace
WaitExten(5)
with
Background(silence/5)
I use
From: Michael Kamleitner [EMAIL PROTECTED]
Date: Wed, 25 Apr 2007 17:47:34 +0200
thx for all of your suggestions... I'm learning more about asterisk every
minute :)
Barton, I tried to replace 'WaitExten' with 'Background' as you suggested,
and at first was disappointed that didn't change the
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak
Edoardo Serra wrote:
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's
Цитат на писмо от Doug Shubert [EMAIL PROTECTED]:
Hello,
I'm trying to use MySQL for Dialplans and have followed
the
Asterisk RealTime Extensions setup.
The MySQL table is called extensions and I have entered
two records..
ext 1000 and 2000.
I also added
switch = Realtime/[EMAIL
Hello,
I'm trying to use MySQL for Dialplans and have followed the
Asterisk RealTime Extensions setup.
The MySQL table is called extensions and I have entered two records..
ext 1000 and 2000.
I also added
switch = Realtime/[EMAIL PROTECTED]
in extensions.conf
and
extensions =
Steve, I was hoping for something native to Asterisk, ie something not
requiring a new process.
Steve Totaro wrote:
Madplay
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source?
Anyone ever tried pointing app_playback at a named pipe ?
On 2 Apr 2007, at 17:11, Doug Garstang wrote:
Steve, I was hoping for something native to Asterisk, ie something
not requiring a new process.
Steve Totaro wrote:
Madplay
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Madplay
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means it has to decode them.
Doug Garstang wrote:
Stephen Bosch wrote:
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Your mistake is replying to an existing thread and changing the subject
line instead of starting a new one.
Start a new thread, and people are more likely not only to notice your
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream, but
I assume that plays MP3's, which means it has to decode them. I'm
looking for something that could play .wav or .ulaw (g711) streams.
Doug.
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means it has to decode them. I'm
looking for something
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means it has to decode them. I'm
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Your mistake is replying to an existing thread and changing the subject
line instead of starting a new one.
Start a new thread, and people are more likely not only to notice your
message, but reply to it.
-Stephen-
Stephen Bosch wrote:
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Your mistake is replying to an existing thread and changing the subject
line instead of starting a new one.
Start a new thread, and people are more likely not only to notice your
message, but reply to it.
Eric ManxPower Wieling wrote:
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means
Doug Garstang wrote:
Eric ManxPower Wieling wrote:
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that
Thanks all for your input.
Based on the comments given I guess I could
replace the X100p for a TDM22B and then connect the Fax machine to the
TDM22B
and in the dial plan hae
exten = fax,1,Dial(SIP/40) (being the TDM port) and then it will ring
the fax machine
and pass the call in from the PSTN
On Mon, 12 Feb 2007, Barry Fawthrop wrote:
Thanks all for your input.
Based on the comments given I guess I could
replace the X100p for a TDM22B and then connect the Fax machine to the TDM22B
and in the dial plan hae
exten = fax,1,Dial(SIP/40) (being the TDM port) and then it will ring the
fax
On Fri, 9 Feb 2007, Barry Fawthrop wrote:
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Question is how does asterisk detect the call without answering?
I'm not wanting Asterisk to handle the call if it is a fax if
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan checkup
Thanks Guys
I already have the fax machine a brother all-in-one Printer,
scanner, fax.
I realize the s,3, answers the line
But How can I get s,2, to detect if it is a fax and take it
from
Hi Gordon
Following you dial plan
How does Asterisk know to move from s,2, to either incoming,1, or fax,1,
The only jump I recognize it Goto(internal,incoming,1) which should take
all calls to incoming,1, and not fax,1,
OT: is spandsp rxfax handled by astlinux ?
Thanks again
Barry
Gordon
On Sat, 10 Feb 2007, Barry Fawthrop wrote:
Hi Gordon
Following you dial plan
How does Asterisk know to move from s,2, to either incoming,1, or fax,1,
The only jump I recognize it Goto(internal,incoming,1) which should take all
calls to incoming,1, and not fax,1,
In this particular
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk Box
Incoming Line: Can I have in the dial Plan
[incoming]
exten =
On Fri, 2007-02-09 at 18:35 -0500, Barry Fawthrop wrote:
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk Box
From: Barry Fawthrop [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 18:35:43 -0500
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk
Thanks Guys
I already have the fax machine a brother all-in-one Printer, scanner, fax.
I realize the s,3, answers the line
But How can I get s,2, to detect if it is a fax and take it from there
without answering?
Or can someone explain what make an incoming goto exten = s,..
From: Barry Fawthrop [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 21:49:17 -0500
Thanks Guys
I already have the fax machine a brother all-in-one Printer, scanner, fax.
I realize the s,3, answers the line
But How can I get s,2, to detect if it is a fax and take it from there
without answering?
It
- Original Message
From: Gordon Henderson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 31 January, 2007 12:20:11 AM
Subject: [asterisk-users] Dialplan programming vs. AGI vs. ???
Just a general question
Hello!
Is there any easy way to use the caller ID display info
(CALLERID(name) in Asterisk) in dialplan just as we could use the number in:
exten = _X./67803287, 1, action
I have a SIP GSM device, and when a call comes in, it passes me the
caller ID like so:
-- Sip message Header:
From:
Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
But I'm curious as to the approach others use. Is doing dialplan
coding in an AGI more efficient, or do people just do it that way
because it's easier than learning dialplan code? Or are there some
things that people think they can't do any other
Gordon Henderson wrote:
Just a general question on dialplan programming... I've implemented a
fairly full-featured system using dialplan code only. I've not used any
AGI for it, yet it ticks all the boxes I want it to tick (diverts,
follow-me, voicemail, dnd, outdialing restrictions, simple
Lee Jenkins wrote:
Gordon Henderson wrote:
Just a general question on dialplan programming... I've implemented a
fairly full-featured system using dialplan code only. I've not used
any AGI for it, yet it ticks all the boxes I want it to tick (diverts,
follow-me, voicemail, dnd, outdialing
Just a general question on dialplan programming... I've implemented a
fairly full-featured system using dialplan code only. I've not used any
AGI for it, yet it ticks all the boxes I want it to tick (diverts,
follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant,
and
From:Gordon Henderson [EMAIL PROTECTED]Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing
From: Yuan LIU [EMAIL PROTECTED]
But I'm curious as to the approach others use. Is doing dialplan
coding in an AGI more efficient, or do people just do it that way
because it's easier than learning dialplan code? Or are there some
things that people think they can't do any other way?
So I'm just
Hi Asteriskers,
I have the following :
exten = 1,1,Playback(sample)
exten = 1,2,Read(response,,1)
exten = 1,3,GotoIf($[${response} != *]?300:100)
exten = 1,100,Playback(hello)
exten = 1,101, [[[ do stuff ]]]
exten = 1,300,Playback(reject)
exten = 1,301,Hangup
Which plays a confirmation sample,
Hi,
I'm analyzing freepbx extensions. When creating ivr with freepbx, it
writes like this:
exten = ,1,Answer
exten = ,n,GotoIf($[${CONTEXT}=from-internal]?USERCID:SETCID)
exten = ,n(USERCID),Macro(user-callerid,)
exten = ,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =
exten = ,n,Queue(|t|||300)
exten = *,1,Macro(agent-add,,)
exten = **,1,Macro(agent-del,,)
So my question is , what means these one/two asteriks (*,**
).Maybe it is like priority.?
It means that to login as an agent on the queue you have to dial
* and
Hi folks,
Moving on to a new install, I'm jumping straight to v1.4
Without using Priority jumping I'm wondering what the 'standard' way
to indicate to the calling party that the number the dialed is busy or
unavailable. So,if I have an entry in extensions.conf like this:
[outbound]
exten =
Russell Horn wrote:
Hi folks,
Moving on to a new install, I'm jumping straight to v1.4
Without using Priority jumping I'm wondering what the 'standard' way
to indicate to the calling party that the number the dialed is busy or
unavailable. So,if I have an entry in extensions.conf like this:
On 1/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Looks at macro-stdexten in extensions.conf.sample. Also see show
application dial
Ah, that's exactly what I was looking for - thanks.
Russell
___
--Bandwidth and Colocation provided by
Eric ManxPower Wieling[EMAIL PROTECTED] Wrote on: 11/16/2006 7:36 PM:
Special extensions like a, o, i, etc do not seem to be read from
include = 'ed contexts.
Is this a bug or as designed?
In this case, it is not an include. The i seems to work fine. From an off
list discussion, it
Asterisk 1.2.12.1
The * key and the 0 key do not seem to be detected in my dialplan. I am
using a and o to detect them.
It simply falls thru to my i where it says, I am sorry that is not a valid
extension.
joe a.
___
--Bandwidth and Colocation
Special extensions like a, o, i, etc do not seem to be read from
include = 'ed contexts.
joe a. wrote:
Asterisk 1.2.12.1
The * key and the 0 key do not seem to be detected in my dialplan. I am using a
and o to detect them.
It simply falls thru to my i where it says, I am sorry that is not
Did not know how to make up a subject line for this.
I have a dial plan that allows a caller can try my cell phone. And that's
fine. If the call cannot be made, it sends caller back to voice menu.
However, I'd like a way for the caller to elect to go back to the voice menu,
if they end up
Subject:[asterisk-users] dialplan issue - 1 0 should be evaluated false
Helo List,
Sorry I missed the rest of my email in my previous post. Please see below.
I'm having an issue using the AND () operator evaluation in the code of my
dialplan. The dial plan is coded to detect inbound DTMF digits from
stuff.
[]'s
MM
-Original Message-
From: Esteban Guana-Jarrin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Fri, 27 Oct 2006 15:11:33 +1000
Delivered: Fri, 27 Oct 2006 01:51:20
Subject:[asterisk-users] dialplan issue - 1 0 should be evaluated false
Helo
MM,
The $[] made it work. Thanks a lot for your assistance.
obligado :)
See the debug output below.
-- Executing NoOp(SIP/123-d14f, no) in new stack
-- Executing NoOp(SIP/123-d14f, yes) in new stack
-- Executing GotoIf(SIP/123-d14f, 0?7:4) in new stack
-- Goto (test-check,s,4)
--
Helo List,
I'm having an issue using the AND () operator in the code of my dialplan.
The dial plan is coded to detect inbound DTMF digits from callers. key 1
is equivalent to yes and key 2 is equivalent to no in my dial plan.
When a caller presses 1, yes is passed as a varialble and same when
Helo List,
Sorry I missed the rest of my email in my previous post. Please see below.
I'm having an issue using the AND () operator evaluation in the code of my
dialplan. The dial plan is coded to detect inbound DTMF digits from callers.
key 1 is equivalent to yes and key 2 is equivalent to
Just a thought I had.
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Douglas Garstang wrote:
Just a thought I had.
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
Doug.
Doug,
It would be cool, but for now you can use System() and logger. If you
need to get something done quickly...
--
Kristian Kielhofner
You could uses System() and the Logger command. Wouldn't be hard.
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialplan Syslog
Just
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
That could be usefull, but what is wrong with : System(logger Asterisk
can use syslog) ?
___
--Bandwidth and Colocation provided by Easynews.com --
: Re: [asterisk-users] Dialplan Syslog
It'd be cool if someone wrote a syslog() dialplan
application for Asterisk *hint* *hint*
That could be usefull, but what is wrong with : System(logger Asterisk
can use syslog) ?
___
--Bandwidth
On Wed, 4 Oct 2006, Kristian Kielhofner wrote:
Douglas Garstang wrote:
Just a thought I had.
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
Doug.
Doug,
It would be cool, but for now you can use System() and logger. If
you need to get
Can anyone give me dial plan for thirdparty confrencing without channel
redirect.
I think channel redirct command is not supported in asterisk now.
Thanks
Imthiyaz
mail2web - Check your email from the web at
Hi all,
I'm trying to find some info on how to create my own dialplan
applications. Like f.i. Echo (ast_echo.c in apps). The API used in there
is what I would like docs on.
TIA
/Rob
___
--Bandwidth and Colocation provided by Easynews.com --
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert it
to an int. But the math function ony takes arguments as int. Can anyone suggest
how to do that?
eg:-
exten =
On 14:23, Wed 30 Aug 06, [EMAIL PROTECTED] wrote:
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert
it to an int. But the math function ony takes arguments as int.
Hi Michael,
Thanks a lot. I am working on an agi script and it does it. Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Michiel van Baak wrote:
Glad I could help. I agree, these mailing lists are a life saver. I
personally have only been using Asterisk for about 5 months now, in fact
I have never even delt with any PBX's before (complete newbie) but
everyone here is very helpful and I am picking up a lot.
Kevin
David Cook wrote:
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2 =
66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 =
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those Doh! moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3=800, etc. rather than by 2's, (so it doesn't overlap with
8XX area codes) and select my
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort
of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
exten = _1877NXX, Dial,
exten =
On 8/18/06, David Cook [EMAIL PROTECTED] wrote:
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort
of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan or matching
On 8/18/06, David Cook [EMAIL PROTECTED] wrote:
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches
sort of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten
Can anyone help point me in the right direction? I have calls coming
into Asterisk over a PRI, all going to the same #. I need to have
asterisk route the calls to a different location based on the NPANXX
of the callerId for the inbound call. Something like
exten = 123,1,$newnumber =
You can do:
exten = 123,1,GotoIf($[${CALLERID(num):0:6}=123456]?50);if cidnum is
123456 goto 50
Or you can do this:
exten = 123/_123456.,1,Goto(50);if cidnum is 123456 goto 50
exten = 123/_234567.,1,Goto(51);if cidnum is 234567 goto 51
exten = 123/,1,Goto(70);anything else or blank goto 70
exten
How's tou're service with Sellvoip, I was not able to intergrate them into my system and they had no phone support. I'm using Gafachi now but prefer the rates Sellvoip provide.
-- Original message -- From: "AR Tarzi" [EMAIL PROTECTED]
SellVoIP appears to follow a US
Subject: Re: [Asterisk-Users] Dialplan -
strip IDD prefix and insert another
How's tou're service with Sellvoip, I was not able to intergrate them
into my system and they had no phone support. I'm using Gafachi now but prefer
the rates Sellvoip provide.
--
Original
Hello.
Is there any way to forward incoming call to voicemail in
one ring if the person on the extension is busy.
Regards
---
Navneet Shah
Systems Administrator
YL Consulting, Inc.
210-340-0098
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