design your dial-plan for routing a specific number to different context ,
you can try func_odbc for query to DB if you have a large number of setup.
ideally its called click to call but you are made it as, miss call and you
will get a call.
regards
dhaval
On Mon, Mar 28, 2011 at 5:21 PM, Roger
On Tue, Mar 29, 2011 at 12:23 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
design your dial-plan for routing a specific number to different context ,
you can try func_odbc for query to DB if you have a large number of setup.
ideally its called click to call but you are made it as, miss
Hi,
I'm trying to setup Asterisk so that:
1. I call a specific number that goes to a defined extension from my
phone (an external line).
2. Asterisk notes my phone number (the CLID) and hangs up without
picking up the call.
3. Asterisk initiates a call to my phone and prompts me for a passkey.
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote:
Is there a better way of handling the post-hangup
processing?
Callfiles?
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New to Asterisk?
On Thu, Mar 20, 2008 at 8:37 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten =
Jim Duda wrote:
== Spawn extension (incoming-dial, fax, 0) exited non-zero on 'Zap/4-1'
Yes, I DO think that's a little odd. It should be priority 1, shouldn't it.
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I'm struggling to get my dialplan to work with a simple analog fax
machine.
I have TDM400B zaptel card with an FXO and FXS port. I have the FXO
port connected to the POTS machine and the FAX machine connected to the
FXS port.
The FAX machine itself works fine, I can FAX outgoing messages
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
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:
hi,
can anyone please guide me as to how i can implement this in extensions.conf:
my PSTN line normally has its longdistance capability locked which can be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long distance using the
zap channel, how can i
hi,
can anyone please guide me as to how i can implement this in
extensions.conf:
my PSTN line normally has its longdistance capability locked which
can
be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long distance using the
zap
channel, how
thanks steve,
the reason i cannot remove the restriction on the telco line is that an
analog fone is connected to the phone jack of the x101p and some
visitors occasionally use the fone and they're supposed to only call
local toll free numbers.
Your suggestion of doing the restrictions within
How about this?
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
-or-
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
I have not seen restrictions set before dialing since local numbers
would not fall under the restriction but that is what you said. Usually
you dial the number
Thanks Steve,
But this will not work for me because after yourcodehere the line
will give a confirmation tone (similar to a congestion tone only
faster) then after flashing or certain period will turn into a busytone
and to get the dialtone again i need to Flash again before i can dial
${EXTEN}.
Wow, what a pain. I would just pickup an FXS and be done with it.
Thanks Steve,
But this will not work for me because after yourcodehere the line
will
give a confirmation tone (similar to a congestion tone only faster)
then
after flashing or certain period will turn into a busytone
This might work if you switch it around a little.
http://www.voip-info.org/wiki-Asterisk+cmd+Flash
-Original Message-
From: MZ [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 27, 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Yeah, and unlocked ATAs are not available in the market here.
I even had to pay almost twice the cost of my X101p clone for shippingOn 11/28/05, Steve Totaro [EMAIL PROTECTED]
wrote:Wow, what a pain.I would just pickup an FXS and be done with it.
Thanks Steve, But this will not work for me
Sometimes for me unknown reasons a wakeup call cannot delivered to a
phone and ends up in the voice mail box (and consequently sent via email
to the phone user).
It would be nice to find the reason why the phone was not reachable, but
for sure it is useless to send a wakeup call to the
What I want is for an incoming call to ring for say 20 seconds, then
hangup, then call an external script. A simple callback setup.
If I do this, at priority 3 the caller doesnt' get hungup, but
instead the line just keep ringing after callbback.agi is run. Why
is that?
exten =
-Original Message-
From: Chad Brown
Sent: Tuesday, December 21, 2004 8:02 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not
thePSTN
Flynn,
Yes, that makes sense. However, in my case I have incoming calls
arriving on an IAX channel from
Yes...Crystal.
Thanks Flynn
-Original Message-
From: el Flynn [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 21, 2004 10:31 PM
To: Chad Brown
Subject: Re: [Asterisk-Users] Dialplan help - Can dial any user but not
thePSTN
Chad Brown wrote:
Flynn,
You are being patient with me
What is the most efficient way to allow inbound callers to
dial internal users yet restrict them from outbound PSTN calls? Today I have a
basic greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I transfer the
Chad Brown wrote:
What is the most efficient way to allow inbound callers to dial internal
users yet restrict them from outbound PSTN calls? Today I have a basic
greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I
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
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