On Tue, Mar 29, 2011 at 12:23 PM, DHAVAL INDRODIYA
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 call and you
> will ge
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 B
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? J
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.
4.
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 sen
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 => _4xx,1,Dial(SIP/${EXTEN}&IAX2/
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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ast
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 fine
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
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 chan
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 (
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:
>
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
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 => _X.,n,S
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
: Re: [Asterisk-Users] Dialplan help
>
>
> 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 busy
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 bu
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
${EXTEN}
How about this?
exten => 1234567890,1,Dial(Zap/g1/yourcodehere${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,
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 ast
>
> 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
> c
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 ini
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 mailbox
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 => xxx,1,Ringi
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 patien
-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 ch
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 transfer
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 inbou
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 P
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