calls
> > asterisk from the command line to verify some uptime stats. I would
> > like to not have the console log the connections.. Any ideas are
> > appreciated.
>
> Use AMI instead?
+1
Check out the attached perl script for a starting point. The
Asterisk::AMI::Common modul
it already uses
a digest authentication mechanism, which allows the server to verify the
password without having it sent in plain text. In other words, it's
already encrypted.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
a variable and hangup, and the
second there's no priority 1 for that extension... I've never tried
that... I'm assuming it just won't work.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
}/${EXTEN:1})
The ${EXTEN:1} strips off the 9 at the front of EXTEN, because babytel
doesn't want to get the 9 that you dial on the phone. If you go with
dialing without the 9, you'll use ${EXTEN}, not ${EXTEN:1}.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP
day.
The hamsters should be running in their wheels again now.
Cheers Matthew. Give them some food from me.
But remember, the key to an HA hamster cluster is staggered feeding
times!
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID
On Fri, 13 Feb 2015 18:47:02 + (UTC)
thufir hawat.thu...@gmail.com wrote:
when running asterisk -r, is there a way to turn off the messages? I
didn't find the answer in the man page.
logger mute
It toggles the messages on and off.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
killall -9
asterisk; done running in a shell, and forgot about it.
You can list all the processes with the command ps -ef
And to see if anyone else (or yourself) is logged in, run w. That
will show every individual session and where they're connected from.
--
C. Chad Wallace, B.Sc
a space between third and party.
That would make two directories, and then cd into 'third' (probably).
Then the path you passed to configure didn't exist.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
On Wed, 26 Mar 2014 16:20:58 +
Michelle Dupuis mdup...@ocg.ca wrote:
If this is to 972 area code then the next digits should be 0X or 0XX
but they are not.
You never dial the local trunk prefix when you're calling
internationally.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
without running ./configure. That's what ./configure does. So
the only sensible thing to do is to run it every time.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
this means is, you can have a register-free environment or you
can have host=dynamic, but you can't have both.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
only want to listen, you
wouldn't need a microphone.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api
this:
exten = _91NXXNXX,1,Set(CALLERID(all)=mycompanyinc123-456-7890)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation
On Mon, 9 Dec 2013 16:15:14 -0800
Bryan Anderson shadow...@gmail.com wrote:
On Mon, Dec 9, 2013 at 4:11 PM, Chad Wallace
cwall...@lodgingcompany.comwrote:
On Mon, 9 Dec 2013 15:47:57 -0800
Bryan Anderson shadow...@gmail.com wrote:
I have a call queue that rings about 15 users
absolutely forbid an agent from taking the next call, but it would make
sure every other agent had priority. You could also add a large wrap up
time, to ensure they never get a second call within a certain time
period.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com
.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
--
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
C. Chad Wallace, B.Sc.
The Lodging Company
of the previous configuration.
Have you considered switching the strategy to roundrobin or leastrecent?
You could give it a very low agent timeout (like 5 seconds), so the
caller doesn't have to wait long if it has to ring a few people.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
Polycom about it, and they said we'd
have to get our vendor to order it as a feature request, or something
like that.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
/2022067
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
after the Dial in the
Local channel.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
, typically you only have contact with one end of a call (your
users) so it's very hard to say that something didn't happen on the
other end (somewhere out in the wild, where people drive through
tunnels).
PS, Sorry for the late reply... I haven't checked the list in a week.
--
C. Chad Wallace, B.Sc
hints table can be made. Neither ways seem to work. Have you
any working example?
Did you try priority 'hint'?
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
.
Thanks!
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
down, possibly dodging signatures, disclaimers,
pasted logs, and dragons, just to get to the question being answered.
Original Message
Subject: Re: [asterisk-users] Trigger Asterisk after data inserted in
mysql
From: Chad Wallace cwall...@lodgingcompany.com
Date: Wed
place a call on the line,
it's answered immediately because there's no signalling. Call
progress information is given solely by tones in the audio. If you
want signalling, you have to use digital lines (like ISDN or SIP).
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
we had analog
lines. You'd have to look up how--maybe just in the fxotune man page.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth
this on a
Linksys but I can't find out how to do it on a Polycom.
I would be greatly appreciate is some is able to tell me how this is
accomplished.
call.autoOffHook call.autoOffHook.1.contact=
call.autoOffHook.1.enabled=1 call.autoOffHook.1.protocol=
/call.autoOffHook
--
C. Chad Wallace, B.Sc
, A would get C's voicemail, as expected.
With an attended transfer, I would expect B to get C's voicemail,
then hang up, resume the call with A and tell them that C is not
available.
A wouldn't get anyone's voicemail unless it was a blind transfer.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
reload? There may be a
syntax error before your not_logged_in line...
Also, along the same vein, you might try moving the
[internal-privledged] context (with the switch/DUNDi line) to below the
[internal] one.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP
.
For the bosses, I would suggest sending them to an actual dog and pony
show instead. But that's just me. ;-)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
could pipe it through less or copy-and-paste it into a text editor
to search it for anything about dahdi. It should tell you what's wrong.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
probably need to run make menuselect after ./configure and before
make to select dahdi for building installation.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
,Dial(${FD_L1},25,trw)
exten = 1,n,GotoIf($[${DIALSTATUS}=BUSY]?line2:voicemail)
exten = 1,n(line2),Dial(${FD_L2},20,trw)
exten = 1,n(voicemail),Voicemail(4)
exten = 1,n,Hangup()
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
+extensions.conf
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
mutually exclusive. It
may be a good idea for Asterisk to either document that they don't work
well together, or to make ringall disable autofill. Any comments?
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
on the web that sums up what the major
changes were within the four active branches?
Maybe the release announcements are what you're looking for. e.g.,
for 1.8:
http://www.asterisk.org/node/51444
And you can probably find the same for 1.4, 1.6.x, and 10 without too
much trouble.
--
C. Chad
asterisk.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
On Thu, 21 Jul 2011 13:29:09 +0800
Malvin Rito mr...@mail.altcladding.com.ph wrote:
My asterisk box was hacked! Can anyone help on how do I secure my
asterisk box, currently my box is installed with 2 NIC. 1st NIC is
for LAN access and 2nd NIC has a public IP which is registered to our
VoIP
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing as
app_macro has been installed okay?
Macro
On Tue, 07 Jun 2011 14:17:41 -0400
sean darcy seandar...@gmail.com wrote:
Call from 'sip' to extension '+1xxxyyy' rejected because
extension not found in context 'out'.
But
[out]
exten = s,1,NoOp( this is the extension: ${EXTEN})
exten = s,n,Answer()
exten =
On Tue, 3 May 2011 18:45:32 +
satish patel satish...@hotmail.com wrote:
I found following dialplan on net but somehow its not going to set
CFIM in asterisk database (asterisk 1.8.3.3). Any idea ?
exten = *72,1,Answer
exten = *72,2,Wait(1)
exten =
to act on the key. There is a
sample out there somewhere (I've seen it) that uses the same CFIM
database keys that you're setting. Wherever you got the code to set
those keys, you should be able to find the code for reading and acting
on them...
On May 3, 2011, at 5:41 PM, Chad Wallace
cwall
On Mon, 11 Apr 2011 12:58:39 +0200
magnu...@inputinterior.se wrote:
U were right, breaking it into two lines work.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)})
exten = 0424449631,n,NoOp(${name:0:-1})
-- Executing
On Thu, 27 Jan 2011 14:52:06 -0800
Jian Gao jian@sjgeophysics.com wrote:
Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk
stop working after the upgrade. Here is the sip debug:
---
--- SIP read from
On Mon, 17 Jan 2011 18:01:14 -0500
Michelle Dupuis mdup...@ocg.ca wrote:
We have an application that plays a variety of sound files on one leg
of a call (generated by a call file). We've been told that the party
listening to the audio files intermittantly hears robotic sounding
audio (on/off
On Tue, 18 Jan 2011 18:17:31 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
It is interesting to note that your mailer (MS-Outlook) has very bad
support for threading. In fact, it (combined with the MS-Exchange
server) does not really bother reproducing the mail headers that are
required
On Sun, 02 Jan 2011 17:44:19 +
duane.lar...@gmail.com wrote:
I have asterisk 1.8.0 installed and I am not able to forward a
voicemail from one users mailbox to another user.
I had the same issue. It was a regression caused by a fix for ODBC
storage, and it seems to have affected every
On Wed, 29 Dec 2010 16:41:17 -0700
Joseph syscon...@gmail.com wrote:
No, it is not a space issue, I tried:
exten = s,3,GotoIf($[${CALLERID(num)}=4715665]?4:6)
but it still goes to priority 6
Have you verified the value of CALLERID(num) by passing it to Verbose?
Could it be that there are
On Wed, 29 Dec 2010 21:55:58 -0700
Joseph syscon...@gmail.com wrote:
I've tried to simplified the dial plan and use n instead of numbers
but I've noticed it is not executing my voicemail if I substitute
number with n
In the example below when the call is not answered, it does not go to
On Tue, 23 Nov 2010 18:57:16 -0500
bakko asannu...@gmail.com wrote:
Hello,
I'm trying to use SIP_HEADER function on my dialplan but I receive
this message (on the console):
pbx.c:3367 ast_func_read: Function SIP_Header not registered
Why?
I believe function names are case sensitive,
On Tue, 16 Nov 2010 14:08:45 +
Vilius Adamkavicius vilius.adamkavic...@invade.net wrote:
For some reason we are seeing Avoiding deadlock for channel in our
Asterisk logs, the logs are getting filled up with an amazing speed
around 12000 lines a second, and all of them are Avoiding
On Sat, 13 Nov 2010 20:38:30 -0500
Thomas Perron thomas.per...@gmail.com wrote:
Here is a very very basic config. But, not working (:
I simply want to dial the DID that is registered with the SIP
provider. then, as you can see the call should dial the 703111 number
Hints please?
[...]
exten
On Thu, 4 Nov 2010 20:12:54 -0400
Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
We have three different Queues set to leastrecent strategy and from
time to time I hear someone complain that they receive short rings
(partial ring cycle) and since it's not their turn even if they
pickup
ported to that platform?
The real question is, does it have PCI slots for Digium cards?
And where do I get one of those HUGE coke cans?! ;-)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
by the Local
channel. Also, look up the /n option to the Local channel. That may
affect it, but I can't say how off the top of my head.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
';
}
}
if (cmd == 't') {
cmd = 0;
vms.repeats = 0;
}
break;
commented out till here */
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
you could do is make one pattern for each possible length.
e.g.: _XXX*X and _*X
If you need it to be variable length, I think you would need to use the
Read application instead of standard dialplan matching.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
Works ok
I suspect it's because when the call first comes in, asterisk doesn't
have the callerid info yet (it comes after the first ring). So
asterisk tries to route the call to a callerid-nonspecific dialplan
entry, and simply fails when it doesn't find any.
--
C. Chad Wallace, B.Sc
voicemail
greeting. :-)
You might also consider AMD [2] (answering machine detection), but I
don't know much about it.
[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
[2] http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
load balancing and the other to our LAN.
I would like asterisk to only accept connections coming from our LAN
but, can't find where to configure this.
Set bindaddr in sip.conf.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
will be
included at the row that the #include statement occurred.
So putting your include before your main [globals] puts the
[globals](+) in first.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP
} variable with Verbose() calls,
something like this:
[incoming]
exten = _X.,1,Verbose(Incoming call to ${EXTEN});
exten = _X.,n,Playback(welcome);
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
troubleshooting tips.
Try different values of dtmfmode (rfc2833, inband, info) in sip.conf
for the SIP peer that you call in from. Asterisk is probably monitoring
the wrong method for DTMF.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
.
GLHF!
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
--
_
-- Bandwidth and Colocation Provided by http://www.api
At 3:09 AM on 21 Jan 2010, __ wrote:
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
cwall...@lodgingcompany.com wrote:
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting
://lists.digium.com/mailman/listinfo/asterisk-users
And a proper mail client will also parse the headers and provide
unsubscribe information/buttons based on that...
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
At 12:36 PM on 30 Dec 2009, hadi motamedi wrote:
Dear All
Can you please give me more hint on how Asterisk Dictate() works?
Thank you
http://lmgtfy.com/?q=asterisk+dictate
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
is the PRI connected to the ericsson (group=1 in
chan_dahdi.conf), and 18 seconds is 3 rings.
You might be able to use Queue(), but I'm not sure if you can add a
hunt group and external number as a queue member--you might have to use
the Local channel for that.
--
C. Chad Wallace, B.Sc
);
Hangup();
goodmorning:
Playback(goodmorning);
};
};
Basically, just change each of the hours in your time specs to this:
$[hour+${TIME_OFFSET}]
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
. The others will just be disconnected. If you want
it to ring the second number only after the first one didn't work,
you'll have to do that in your dialplan by checking ${DIALSTATUS} after
Dial.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID
a SIP phone
is the first step of an attended transfer or an original call?
Thanks!
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
___
-- Bandwidth
wrote:
Nothing. I don't know what in the world is going on with my setup.
[...]
I'm already frustrated with this.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
or extensions.ael.
You probably just have to comment out the Playback line.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
___
-- Bandwidth
attempts. With the
Wait() solution, that caller would be waiting for 30 seconds regardless
of whether there's anyone else available.
Of course, I don't know your business case, so you'll have to decide
which of the two problems is worse.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote:
C. Chad Wallace cwall...@lodgingcompany.com writes:
Also, if there is another agent available, the caller would be
connected immediately, and it wouldn't have to make any more
attempts. With the Wait() solution, that caller would
consoles, and I was wondering if
someone had successfully ported Linux and Asterisk to the current
hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360?
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description
the
call through; otherwise, skip that agent.
Sorry, no example code yet... I just wanted to get the idea out there.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote:
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending
in my /etc/asterisk directory:
asterisk.conf
cdr.conf
cdr_custom.conf
extensions.ael
extensions.conf
features.conf
indications.conf
logger.conf
modules.conf
musiconhold.conf
queues.conf
sip.conf
voicemail.conf
zapata-channels.conf
zapata.conf
YMMV, HTH, HAND. :-)
--
C. Chad Wallace, B.Sc
,Busy()
exten = _X.,n(accept),Set(GROUP()=${DID})
; Now let the call through as usual...
exten = _X.,n,Goto(mainmenu,s,1)
That puts each call into a group named by the DID, and returns Busy
if there is another call on the same DID.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
that anymore. :-)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009
Oops! I missed the part where you said you use a SIP trunk. My
experiences and comments are entirely irrelevant to your case. Sorry!
At 12:02 PM on 10 Sep 2009, C. Chad Wallace wrote:
At 10:22 PM on 09 Sep 2009, John A. Sullivan III wrote:
Hello, all. I've come across a nasty
(__PARKINGEXTEN=${PARKINGSLOT})
You might only need one underscore.
For more info, see 'core show application set'.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
/sbin/asterisk -f
-g -n -p -q
$ ps -fC asterisk
Or for the uid:
$ ps --no-headers -o uid -C asterisk
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
/Asterisk+config+zapata.conf.sample
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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AstriCon 2009 - October 13 - 15
that would play back an audio file that the user has
pre-recorded
Doug
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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keys.
Either do that through the manager interface, or (if you want to
batch commands) send them directly over the unix-domain socket
asterisk.ctl .
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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Description
= *00,n,AGI(festival-script.pl|I will now attempt the call)
exten = *00,n,Set(CALLERID(all)=Notify 9000)
exten = *00,n,NoOp()
exten = *00,n,Dial(SIP/302,15})
exten = *00,n,Wait(2)
exten = *00,n,Playback(demo-congrats)
exten = *00,n,Answer()
exten = *00,n,Hangup()
TTYL...
--
C. Chad Wallace, B.Sc
into OpenSIPS?
Regards,
Chris
via a quick google:OpenSER is now OpenSIPS
www.opensips.org OpenSER continues via OpenSIPS A new name, same
project
Uhhh, I thought that was Kamailio:
www.kamailio.net
...I'm confused.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
(FXS only): Offhook --Cable plugged
Hookstate (FXS only): Onhook --Cable unplugged
^^^
Foxtrot X-ray *Sierra*
When it says FXS only, I think it's reasonable to assume that FXO is
excluded.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com
. ;-)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
Debian Hint #22: Wondering which Debian mirror is best for you? Check
out the apt-spy and netselect-apt packages, which can give you
information about how various mirror sites perform
you would use on all your
ISDN channels. Just don't let that context dial out.
I don't know if you'd want to call that context default... because
that one seems to be special in Asterisk. But maybe I'm just being
superstitious. :-)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
starting from the last FXO
port in the group by calling Dial(Zap/G2) (capital G means dial down
from last, lowercase g means dial up from first). That minimizes glare.
But, as I said before, if you only have one line, you can't do that...
--
C. Chad Wallace, B.Sc.
The Lodging Company
http
We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed
a change in the behaviour of the queues--a change that we cannot live with.
We've used AddQueueMember/RemoveQueueMember to manage logging into and
out of our queues for over a year now with Asterisk 1.2, and in that
version
, it should go through the queue and
into the Local channel to your outbound extension.
Sorry I don't have any code for you... I haven't done it yet; I'm just
putting the idea out there.
Hope this helps!
Good luck.
- --
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public
, 15) exited non-zero on
'IAX2/lime-3' in macro 'forward'
== Spawn extension (macro-forward, s, 15) exited non-zero on 'IAX2/lime-3'
-- Hungup 'IAX2/lime-3'
- --
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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in FC6.
TTYL.
- --
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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iD8DBQFGN7bcKeSNHCYiCKARAoefAKDAh/V2W3cwd
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