Hi,
I specified within the extnum.conf package to read the configuration for the extensions.conf file from the database.
When i start asterisk i see that it binds the extensions.conf file to the db!.
Now i made an entry within the ast_config table of the database - when i make an reload -
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible?
Any help would be
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
extensions continue to ring. Variations of answer() and ringing() don't
seem to
On Tue, 2004-08-17 at 15:44, Patrick Lidstone (Personal e-mail) wrote:
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
main menu
Dial 1 for support
|
I'm not sure if this is your issue or not, but it looks like ext= 1,
starts over at the bottom of the 1's. You have 1,1-10 and then 1,1 and
2 after it. I can see how asterisk might get confused if you sent your
call back to ext 1 at starting point 1 or 2.
On Tue, 2004-08-17 at 15:03, defiance
On Tue, 2004-08-17 at 17:03, defiance wrote:
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
On Tuesday 17 August 2004 18:03, defiance wrote:
exten = 1,1,SetCallerID(Toll Free No Cpub)
...
exten = 1,1,Playback(cpub-support)
Do you see a problem? 'cos I sure do...
You can use the numbers over again if you use Goto and jump to a different
context.
-A.
That makes sense, but how do I send it to each context?
chris
On Tue, 2004-08-17 at 17:07, Steven Critchfield wrote:
On Tue, 2004-08-17 at 17:03, defiance wrote:
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a
CONTEXTS
[main context]
Dial 1 for support
| Dial 2 for special
| Dial 3 sales
| Dial 5 For sales
[support context]
; don't include main context
On Tue, 2004-08-17 at 17:33, defiance wrote:
That makes sense, but how do I send it to each context?
use goto(newcontext,s,1)
On Tue, 2004-08-17 at 17:07, Steven Critchfield wrote:
On Tue, 2004-08-17 at 17:03, defiance wrote:
I am making some changes to the dial plan at the request of the
Thanks a million man that works beautifully, and thanks for giving me an
example, I am still pretty new at this so that helped alot.
chris
On Tue, 2004-08-17 at 17:31, William Glynn wrote:
CONTEXTS
[main context]
Dial 1 for support
| Dial 2 for special
minutes...
- Original Message -
From: defiance [EMAIL PROTECTED]
To: asterisk [EMAIL PROTECTED]
Sent: Tuesday, August 17, 2004 3:49 PM
Subject: RE: [Asterisk-Users] dialplan woes
Thanks a million man that works beautifully, and thanks for giving me an
example, I am still pretty new
Can anyone shed some light on this ??? Or is this not the right sort
of question to ask?
It simply doesn't work with the current software. You need to code this at
the C source level.
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Can anyone shed some light on this ??? Or is this not the right sort
of question to ask?
It simply doesn't work with the current software. You need to code this at
the C source level.
/me daydreams about being able to fork in the dialplan.
Actually you sort of can - wonder what
Hi,
-Original Message-
/me daydreams about being able to fork in the dialplan.
Actually you sort of can - wonder what something like:
Dial(Zap/1LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
would do?
That works. I use it on a daily basis. Does not make your CDR any more
Can anyone shed some light on this ??? Or is this not the right sort of
question to ask?
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Wednesday, 4 August 2004 11:44
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
Does anyone know how to do the following:
1. Caller calls in
2. Asterisk answers.
3. Asterisk rings nominated extensions
4. Caller keys in certain digits while extensions are ringing
5. Caller is directed to another extension based on the digits keyed in
I can achieve this if I have Asterisk
All,
I was a bit too focused on where I thought the problem was - turns out
I wasn't crazy and the dialplan does work as expected. The problem was
with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for
the premature post for help.
Begin forwarded message:
From: Ben Witso [EMAIL
Matthew,
Dial works on a fall thru principle. Thus:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,Dial(SIP/2000,30)
should suit your purpose (not taking into account vm), to add another exten just add
it on the dial 'list':
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,Dial(SIP/2000,30)
exten
*** BEGIN FORWARDED MESSAGE ***
On 07/06/2004 at 23:34 Andy Powell [EMAIL PROTECTED] wrote:
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Tue, 08 Jun 2004 14:54:33 +0200
Subject: Fwd: Re: [Asterisk-Users] dialplan experts needed
Sorry misread your message, you want
exten = 555,1,Dial(SIP/1000,30)
exten = 555,102,Dial(SIP/2000,30)
exten = 555,103,Dial(SIP/3000,30)
exten = 555,104,Voicemail2(u3278)
exten = 555,105,Hangup
exten = 555,2,VoiceMail2(u3278)
exten = 555,3,Hangup
...should be
exten = 555,1,Dial(SIP/1000,30) ; Unanswered = 2, Busy = 102
On 08/06/2004 at 11:15 John Fraizer wrote:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,102,Dial(SIP/2000,30)
exten = 555,103,Dial(SIP/3000,30)
exten = 555,104,Voicemail2(u3278)
exten = 555,105,Hangup
exten = 555,2,VoiceMail2(u3278)
exten = 555,3,Hangup
...should be
That's why
I have the same situation - i.e. three different extensions scattered
about. But I don't try them each individually. When a call comes in my
asterisk attempts to ring up to four different devices at the same time.
To do this using your dial plan is easy - i.e.
exten =
In this dialplan, the SIP user agent is a Sipura two line adapter with line
1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set
up so that 1000 and 2000 are lines in hunting on incoming extension 555.
I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring
Good day All.
Is there a way to pass DTMF signals to AGI script during conversation ?
Actually here what I want to make:
Users are usually dial using dialplan and when someone press *4 (during
conversation) I want to have agi script to deal with that, but those
users should keep talking and even
On Thu, 27 May 2004, Vladyslav wrote:
Good day All.
Is there a way to pass DTMF signals to AGI script during conversation ?
Actually here what I want to make:
Users are usually dial using dialplan and when someone press *4 (during
conversation) I want to have agi script to deal with
I need to set up a dialplan that is time sensitive.
That means it will change after a specific time of day.
How can that be done ?
Examples ?
Rgds
EEA
_
Last ned MSN Messenger gratis http://www.msn.no/computing/messenger - Den
I need to set up a dialplan that is time sensitive.
That means it will change after a specific time of day.
How can that be done ?
This is *right* out of the handbook. Please, please *please* do some basic
research before asking questions to the list... I mean come on now.
Regards,
Andrew
On 2004 Mar 09, at 12:36, M H wrote:
I need to set up a dialplan that is time sensitive.
That means it will change after a specific time of day.
How can that be done ?
CLI show application GotoIfTime
-Tilghman
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Asterisk-Users mailing list
[EMAIL
Hi, I need to set up a dialplan for a voice menu and two extensions.
One extension is to a GSM phone and one to a VOIP phone e.g. a Grandstream
phone that is on the local network here. Most important is the GSM phone,
the VOIP phone can come later.
This system is to be used for a very small
Hi guys,
I've posted my dialplan design in flowchart format and would welcome
your comments for improvements. I'd like to know my logic looks ok
before I go much farther or start rewriting my extensions.conf.
http://www.triadinternetsystems.com/asterisk/
For the Standard Extension macro, I
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of fredrik
chabotSent: zaterdag 6 september 2003 18:25To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Dialplan
questionHi,Dialplan QuestionI'm in
holland and I have:[naarbuiten]ignorepat
Hi,
Dialplan Question
I'm in holland and I have:
[naarbuiten]
ignorepat = 0
; interlocaal
exten = _00[1-9],1,Dial(Modem/g1:${EXTEN})
exten = _00[1-9],2,Congestion
; locaal
exten = _0[1-9]XX,1,Dial(Modem/g1:${EXTEN})
exten = _0[1-9]XX,2,Congestion
And sometimes I
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