It´s prety easy to do that (i've made exactly the same for a directory):
asuming a 3 digit extensions on zap channel
[directory]
exten => _XXX,1,flash;no need to put a channel, this flashes
the active channel
exten => _XXX,2,sendDTMF(w${EXTEN})
exten => _XXX,3,hangup
The pbx system is
I'll try :)
Lets say we are dealing with device a in sip.conf (sip/a), and sip/a
uses context a-outbound for dialing outbound:
[a-outbound]
exten => _1NXXNXX,1,Set(CDR(accountcode=1234)
exten => _1NXXNXX,2,Dial(whatever)
the above will make sure no matter what happens if sip/a transfer t
Well spanish is a language I don't speak.
On 8/11/06, Juan Pablo Abuyeres <[EMAIL PROTECTED]> wrote:
Ok, let's give it a try in Spanish please :)
On Fri, 2006-08-11 at 16:08 -0400, C F wrote:
> OK, just let me know what other language you want it in, I'm fluent in
> more than english, just let
Thanks Tony.
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Sat 8/12/2006 2:22 AM
To: asterisk-users@lists.digium.com
Cc:
Subject: [asterisk-users] Re: AgentcallbackLogin()
In article
On Fri, 11 Aug 2006, Senad Jordanovic wrote:
> [EMAIL PROTECTED] wrote:
> > Hi
> >
> > After a month or so using Asterisk we've had or first downtime period
> > due to a faulty RAM chip on the server, so we're starting to think
> > about the possible high-availability solutions.
>
> Hi
>
> If y
Hi everybody
My name is Jose Manuel Cortes, i'm from
Colombia and im working in a asterisk implementation for my thesis. The initial
system was a pbx and a LAN separated, now with the asterisk server the system
is:
before
Telco1 ---PBXTelco2
now
TelcoPBXasterisk_server--
Hi Rich,
I'm using a wholesale voip origination provider - they don't deal with
end users. As such they have statically defined my Asterisk box on
their end - there's no registration or authentication by my system
with theirs - other than them hardcoding the destination IP of my
server in their s
Hello,
The call for papers for SCALE 5x, the 2007 Southern California Linux
Expo, is now open. This event will be our fifth annual show. It will
be held on Feb 10-11, 2007 at the Los Angeles Airport Westin. We are
expecting around 1,300 in attendance this year. We are non-profit,
community run
That is Camp on
I believe what the OP wanted to do was to dial a number EXTERNAL to
Asterisk until the call went through
Auto Redial and Camp On are NOT the same
Telephony 101
John Novack
Philipp von Klitzing wrote:
Look here:
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
On 8/9/06, Stephen G <[EMAIL PROTECTED]> wrote:
Hi there,
I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA
motherboard as an always on, low powered solution.
Hi Stephen,
I have an A200 with 1 x FXO module board in it.
I'm in the Uk using a BT line and am having a lot of b
I have zero problem compiling the addons 1.23 on FC4 and RH4, but for some
reason, when I try to compile them on FC3, I get this:
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
app_addon_sql_mysql.o app_addon_sql_mysql.c
app_addon_sql_mysql.c:164:36: macro "AST_LIST_REMOVE"
Alexander Lopez wrote:
>>> This might be what you're seeking;
>>>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
>> If the phone rings, then the channel IS available. The solution is to
>> disable call waiting on the SIP device.
>
> The s option needs to be used:
> s - Co
I have the following configuration in using of 2 asterisks.
ARA(DB) / AST1 (10.0.1.0)---UA1 (ext:3634, IP:192.168.1.10)
\ AST2 (10.0.1.5)---UA2 (ext:4634, IP:192.168.1.20)
I have a dial plan in order to allow UA1 make calls to UA2.
exten => 4634,1, Dial(SIP/[EMAIL
Mr. Jones wrote:
This is essentially a follow-up to my previous email on the 404 I was
seeing with my DIDs.
I think it maybe more involved with the SIP headers I'm receiving from
the company providing my origination.
Here's what's interesting.
I have inbound 800 service and outbound terminatio
Thank you for your response. As you said, I changed the context
"default" to "general". Now,
1) When I am making call to our DID, its ringing. But, call is not
transferring to 105 extension.
No one can guess at the above without you providing something from the
CLI to indicate what is going
At 11:54 AM 8/11/2006, you wrote:
Thanks for the suggestion. I can't seem to get it to work.
This is what I put in my extensions.conf
We only have one number that we want to keep trying right now, so I
tried to set it so by calling extension 777, it would start the
system retrying. (The actua
Hi
better way Look up is
ngrep or debug mode at asterisk console
when the call come in, see what happends
and paste that here so , some experts can answer the problem
i feel first you register, and see what happend
Still you feel the problem not solved, paste the Debug here for refernce
Look here:
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
> Also many so-called "legacy" hybrid PBX switches have had this for many
> a year. Hard to compete when well used features that have been around for
> 20 years are lacking
> >> I want something that will keep trying a busy numb
Hi all,
Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys.
[EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2).
Can anybody help me how to write this code for a dialtone of frequency 425 which is continous.
Thanks
Dan
__
Hi all,
Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys.
[EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2).
Can anybody help me how to write this code for a dialtone of frequency 425 which is continous.
Thanks
Dan
___
Sounds to me like you don't have a proper connection with Stanaphone.
The only time you'll get these problems is when they cannot contact you
to forward the call to your system.
Double check you firewall settings. They need to be able to reach your
system on port 5060UDP (assuming SIP) as well as
Hi All
I have a Cisco 7960 which is connected remotely to an Asterisk server.
Both are unfortunately behind NAT.
The Phone registers and is show in sip show peers, with the correct
public ip for the phone and a 100ms qualify time
(1) I can dial the phone from another phone, it will ring but n
>From memory, it wasn't that complex.
What bit are you stuck on?
PaulH
On Fri, 2006-08-11 at 16:51 +0200, Tijl Van den Broeck wrote:
> If you ever see the chance to get the coding again I'm also
> interested. I'm setting up something very similar here.
>
> greetings
>
> Tijl Van den Broeck
>
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Can someone tell me why this is not valid...
>
> [start]
> exten => 1000,1,Answer
> exten => 1000,2,Wait,1
> exten => 1000,3,AgentcallbackLogin(1000||[EMAIL PROTECTED])
> exten => 2000,1,Macro(DialProxy,115551212)
> exte
Hi,
you have in your sip.conf:
register => xyz.abc:[EMAIL PROTECTED]
This register command doesn't tell asterisk what to do with it.
Take for example this register command and other definitions in sip.conf:
register => sipgate-id:[EMAIL PROTECTED]/sipgate-id
and this peer definition
[sipgate]
Hi Ram,I have given the onfiguration files in the last of this mail. Please read that. I registered with teliax and making calls to US using Teliax. As you said, I executed the command "sip show registry". But, Its not showing any registered users. But, how i am doing outgoing calls to US?Looking f
Hi Chandra
You check in the console asterisk -r
sip show regis
will show you the account is Registered with your Voip Provider or not
If not try add in the conf file
register=account:[EMAIL PROTECTED]/account
Ram
On 8/12/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
Hi,Thank you for y
Hi, Thank you for your response. As you said, I changed the context "default" to "general". Now, 1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension. 2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls
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