> Does someone know if Asterisk has been 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?! ;-)
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C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
Without the underscore, the variable won't be inherited 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 I
vmu);
> break;
>
>
>
> if (vms.repeats > 3)
> cmd = 't';
> }
> }
> if (cmd == '
hink it should.
One thing 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
id-nonspecific dialplan
entry, and simply fails when it doesn't find any.
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C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
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--
___
es:
> two to the internet doing 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.lo
e it
through... unless the callee puts the tone for a "1" in his 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/wi
lobals] section in
extensions.conf.
From doc/configuration.txt: "The content of the other file 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
ey pass, or you can just make test
calls and log the value of the ${EXTEN} variable with Verbose() calls,
something like this:
[incoming]
exten => _X.,1,Verbose(Incoming call to ${EXTEN});
exten => _X.,n,Playback(welcome);
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www
eout is
> an issue.
>
> I'd be grateful for any troubleshooting tips.
Try different values of dtmfmode (rfc2833, inband, info) in sip.conf
for the SIP peer that you call in from. Asterisk is probably monitoring
the wrong method for DTMF.
--
C. Chad Wallace, B.Sc.
The Lodging Co
At 3:09 AM on 21 Jan 2010, __ wrote:
> On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
> wrote:
> >
> > At 5:59 PM on 19 Jan 2010, __ wrote:
> >
> >> Test case:
> >> We have e1 trunk and mult
lso use 'core show application System' and such on the Asterisk
CLI.
GLHF!
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C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
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--
___
e mailing list please.
> >
> > Thanks.
> > rick
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> And a proper mail client will also parse the headers and provide
> unsubscribe information/buttons based on that...
--
C.
At 12:36 PM on 30 Dec 2009, hadi motamedi wrote:
> Dear All
> Can you please give me more hint on how Asterisk Dictate() works?
> Thank you
http://lmgtfy.com/?q=asterisk+dictate
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C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0
l(DAHDI/G1/1112,18)
exten => 0,n,Dial(DAHDI/G1/1113,18)
...where DAHDI/G1 is the PRI connected to the ericsson (group=1 in
chan_dahdi.conf), and 18 seconds is 3 rings.
You might be able to use Queue(), but I'm not sure if you can add a
hunt group and external number as a queue member--yo
E_OFFSET}]:00-$[7+${TIME_OFFSET}]:00|*|*|*?goodmorning);
Playback(hello);
Hangup();
goodmorning:
Playback(goodmorning);
};
};
Basically, just change each of the hours in your time specs to this:
$[+${TIME_OFFSET}]
--
C. Chad Wal
at you're looking for. Instead,
it'll ring both (or all) devices at once, and the first one to answer
will get the call. 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 dialp
er a call from 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
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Description: PGP signature
__
dy Landy wrote:
> Nothing. I don't know what in the world is going on with my setup.
>
[...]
> I'm already frustrated with this.
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C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
either in extensions.conf 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
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Description: PGP signature
___
> I don't know much about game 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/
Open
At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote:
> "C. Chad Wallace" 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,
mmediately, and it wouldn't have to make any more 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
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 amoun
is no record or the time has passed, put 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: 0x262208A
modules.conf (with sparse
comments), and here's a list of the files 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-chan
T(${DID})=0]?accept)
exten => _X.,n,Busy()
exten => _X.,n(accept),Set(GROUP()=${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 Wall
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
d to
improve our call quality. We've since moved to a partial PRI instead
of those analog lines, so we don't have to worry about that anymore. :-)
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP
a persistent
variable, which will be inherited by sub-channels, like after a Dial.
exten => _SIP011XX,n,Set(__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://
$ ps -ef | grep /sbin/asterisk | grep -v grep
>
> You should get something like:
>
> root 12477 12476 0 Aug03 ?00:02:09 /usr/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.
Some do, and some don't--you would have to talk to them.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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___
-- Bandwidth and Colocation Provided b
an_dahdi.conf
http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf.sample
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
___
-- Bandwidth and
, if they want that option.
> >
> > And, in your dial plan, check for the existance of that flag. If
> > it's there, then don't jump to the voice mail app, just jump to
> > your context that would play back an audio file that the user has
> >
ab several
> > > values in the astdb using say, asterisk -rx "database show" >
> > > output.txt and work with that and then set a new value such as
> > > asterisk -rx "database put $key $value". The whole process can
> > > take over 1 second f
as the OpenSER project itself morphed 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.kama
t;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.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID:
Did you read my previou msg
>
> Hookstate (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
exc
hing breaks, you get to keep both parts. ;-)
--
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
'm guessing all your Zap channels come from the PRI,
which is connected to the PSTN. If so, then you're right--you just
need one context for your zapata.conf which you would use on all your
ISDN channels. Just don't let that context dial out.
I don't know if you'd want
On our installation, the calls are allocated from the first FXO port
(Zap/25) up. So we set Asterisk to dial out 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
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 th
y.
The tricky part would be passing the dialed number through... But if
you set an inheriting channel var, 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
2/lime-3", "1") in new stack
> -- Executing [EMAIL PROTECTED]:4] GotoIf("IAX2/lime-3",
> "("8585970327")?15:5") in new stack
> -- Goto (macro-forward,s,15)
> -- Executing [EMAIL PROTECTED]:15] Hangup("IAX2/lime-3"
0P in the same box.
However, I am using Debian, and I'm not sure if modprobe and udev work
the same way in FC6.
TTYL.
- --
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
-BEGIN PGP SIGNATURE-
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