From there on you can do
anything you want with the variable ${numb}
If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.
-Bruce
On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman wrote:
Thanks, but I'm missing something her
le, for examples of Asterisk dialplan.
-Bruce
On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman wrote:
Thanks, but I'm missing something here, the dial command is where?
I need to do something like:
Dial(1234)
Read(1 digit)
DoSomthing(based on digit from 1234)
And as far as I u
at, Jul 10, 2010 at 2:16 PM, eyal goltzman wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Hi,
I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
--
_
7;
Subject: Re: [asterisk-users] How to change the IP in the SIP contact header
Have you tried setting
externip=
In the [general] of your sip.conf?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman
Sent: Monday, Jul
Hello,
I'm trying to use a SIP trunk service and the provider ask me to have the IP
address of the contact header as my public IP and not as my private one, how
can I do it?
See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w
is my public address:
sipINVITE sip:14
Hello,
I'm trying to register to my provider sip trunk, I got from him an host IP
(a.b.c.d) to connect to and my provider recognize me based on the fixed IP
(x.y.z.w) he gave me (no need for username and password)
In the sip.conf I add:
[mytrunk]
type=friend
insecure=no
host=a.b.c.d
fromdomain=x
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using AMI Originate to call 2 outside numbers
and connect them
On Sat, Jul 03, 2010 at 01:33:25AM +0300, eyal goltzman wrote:
> Hello,
>
> Can I use AMI Originate to call 2 outside numbers (SIP) and connect them?
> How
Hello,
Can I use AMI Originate to call 2 outside numbers (SIP) and connect them?
How?
Thanks
Eyal
--
_
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Hello,
After installing and learning Asterisk I found myself with a need for a
minimal set of empty configuration files with only the "must have" stuff in
order to setup a SIP only machine, is there a place to find it?
Thanks,
Eyal
--
__
Hello,
When I call "dialplan reload" I can see the following lines:
== Parsing '/etc/asterisk/extensions.conf': == Found
-- Registered extension context 'default' (0x8a72410) in local table
0x8a679d0; registrar: pbx_config
-- Added extension '_1XX' priority 1 to default (0x8a72410)
.
nt: Friday, June 25, 2010 4:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a default dial plan that is not in
extention.conf?
On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote:
> Hi,
>
>
>
> I have a trivial peace of dialplan for exten 100
Hi,
I have a trivial peace of dialplan for exten 100. I try to change it to _1XX
and the asterisk act according to a different (Default??) dial plan and not
the one I want? Is that possible? Where is the other dialplan sits? In my
extention.conf I can't see something that look like what asteris
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