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an FYI, Cisco phones running SIP do *not* do shared line
appearances, on *any* system.
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updated the Communications Rules
located at http://www.asterisk.org/community/rules, so check those out before
posting.
Thanks,
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That is incorrect. AsteriskNOW (actually, the AsteriskGUI) edits files in
place, leaving any old information in them. This allows you to fully customize
your users and dialplan without interfering with the GUI's operation.
Aaron Daniel
Community Relations Specialist
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(256
phones at
the same time:
exten = 100,1,Dial(SIP/101SIP/102,30,t)
exten = 100,2,VoiceMail([EMAIL PROTECTED])
There are all kinds of fancy variations on this theme, but the idea is
the same: one user with many phones, one extension, one voicemail box.
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ask Digium
hth
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... last command you ran... i.e. stop now).
Douglas, why're you even running stop now on a live production
server. If you're not, quit complaining and watch what you type before
hitting enter.
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send me the scripts as well.
Thanks!
David
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On Tue, 2006-11-14 at 12:00 -0700, David Thomas wrote:
On 11/14/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Incorrect :) IAX2 most definitely does support regcontext.
Also, I think what he means is the phone specific information must be
exactly the same from system to system or the failover
separation of services, and if one of those
boxes goes down, you'll have enough saturation to not have to worry
about it. Having the main boxes do the connections to the providers
adds a level of complexity when you need to bring a new system online.
Just my 2 cents :)
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the registrations itself.
According to what Mark Spencer wrote, it should be possible for this DUNDi
Lookup server to hold, or store (ie cache) -all- phone registration info so
that it doesn't have to query the other registration servers.
Doug.
-Original Message-
From: Aaron Daniel
. If pbx1 where to
crash, and the phone failed over to pbx2, pbx2 would then send updated
registration information to the DUNDi precache server, and thus calls would
not fail for an hour.
Douglas.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday
: unrecognized %option: reentrant
argdesc.l, line 20: unrecognized %option: bison-bridge
argdesc.l, line 21: unrecognized %option: bison-locations
make[1]: *** [argdesc_lex.c] Error 1
make[1]: Leaving directory `/usr/local/src/asterisk-svn/asterisk/pbx'
make: *** [subdirs] Error 1
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Nevermind, seems scp'ing the source directory over didn't do me any
good.
On Fri, 2006-11-03 at 09:25 -0600, Aaron Daniel wrote:
I know I compiled AEL2 into 1.2 before, considering I just copied my
source from one server to another, yet I can't seem to figure out why
I'm getting this error
/asterisk-users
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10.20.136.130:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone
CSeq: 1 INVITE
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL
I agree with Lacy.
This is nothing more than a troll. The asterisk-users list is a forum
for asterisk users to advance their knowledge of asterisk, not
constantly rag on asterisk and it's developers.
Some of us are actually interested in learning more and helping others
understand new concepts.
]
e-mail: tparcina#lama.hr
http://www.lama.hr
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://asterisk.mdaniel.net
On Thu, 2006-10-12 at 10:16 -0400, Jay R. Ashworth wrote:
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been
in telephony
interface.
On 10/11/06, Aaron Daniel [EMAIL PROTECTED] wrote:
That doesn't always work :)
There's two options... either port the volgain patch from 1.4
to 1.2 (If
anyone wants a copy, we've been using it for months... however
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of the older code somewhere.
The 1.4 patch is here:
http://bugs.digium.com/view.php?id=6237
Aaron Daniel
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Sam Houston State University
[EMAIL PROTECTED]
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On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote:
I had the same problem.
Checking voicemail
options visit:
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This happens if you have a logo_url configured for your phone and the
phone can't access it. I'm guessing you don't allow 80 through the
firewall to the server that's serving the image.
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On Fri, October 6, 2006 20:13, Robert Goodyear wrote:
Anyone know why I get HTTP Connection
: Wednesday, September 27, 2006 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RPID
DANIEL, AARON MATTHEW wrote:
Has anyone successfully gotten rpid working between two phones through
asterisk?
Aaron Daniel
Computer Systems Technician
Sam
great, it's just not feasible for us just yet, so
we're looking into other options as well. :)
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On Wed, 2006-09-20 at 22:25 +0200, Olivier wrote:
2006/9/20, Aaron Daniel [EMAIL PROTECTED]:
The biggest problem we have with the hinting functions is that
you have
to have the phones registered to the same server, and with two
identical
servers
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Maybe I should just hold a conference call about all this stuff.
On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
-BEGIN PGP SIGNED MESSAGE-
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Roy Sigurd Karlsbakk wrote:
Linked from /. today,
http://www.networkworld.com/news/2006/091206-von-sam-houston.html
no one will know when I send a real email asking how to
push the power button :)
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asterisk
pretty sure that you'll still need to include the @context for the
MWI to work correctly.
Or switch the phone to SCCP. It will give you a lot of extra
power :)
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registered, but I was able to make calls with it. I didn't try
calling it, since I never saw it register. It appeared it was
authenticating for outgoing calls.
On 8/31/06, Aaron Daniel [EMAIL PROTECTED] wrote:
I tried that image for about 5 minutes. Kept getting errors
in asterisk
On Wed, 2006-08-30 at 08:34 -0600, Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 29, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SER Dispatcher Load Balance
on a 7960 with SIP.
Do you know wether it can be done with a 7940(SIP)? Can it display
status of (for example) 4205,hint,SIP/phone1 ?
Conrad
No, not running SIP.
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[EMAIL PROTECTED]
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On Wed, 2006-08-30 at 20:10 -0700, George Pajari wrote:
I'd rather pay the fine than the liability settlement when found
negligent in a lawsuit because someone panicked, repeatedly dialled 911,
and could not reach Emergency when their coworker had a major myocardial
infarction right beside
Well, it really depends on what he's using the asterisk servers for. If
it's for voicemail or apps, he'll just have to make sure that certain
apps land on certain servers, and voicemail can be distributed for
various things. If ser can do what I've heard/read it can do, it can
handle all the
in
extensions.conf, then what use is the database?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 24, 2006 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Realtime and hints
That's what he
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in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
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($[${CALLBACK} =
YES]?s-SUCCESS,3:s-SUCCESS,4)
exten = s-SUCCESS,3,System(/usr/local/bin/vm-callout.sh ${direct_vm})
exten = s-SUCCESS,4,Voicemail([EMAIL PROTECTED])
exten = s-SUCCESS,5,Hangup()
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the extension. Asterisk
doesn't care about extra fields in the db, but it won't use them to it's
benefit unless it's programmed in.
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look like it ever receives the call.
Doug.
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I'm talking about the rotating DUNDi secret that is stored in dbsecret
in iax.conf. It doesn't exist in the SIP channel.
On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
Secret? Do you mean sbsecret in sip.conf?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL
=no
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what IT can handle. There's a matching e164 and internal
context on each server to tell the others what it can take.
On Wed, 2006-08-02 at 16:35 -0600, Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 4:02 PM
Yeah, they're both in the sip.conf files on each server in the system.
That's how I got the SIP/${server}/${exten} to work.
On Wed, 2006-08-02 at 17:01 -0600, Douglas Garstang wrote:
Ok, but don't you need to have [scm1] AND [sgw1] in sip.conf?
-Original Message-
From: Aaron Daniel
public and one
private interface, so that we can have external phones and internal
phones. Works fine.
I'm thinking this is a misconception. We even have heartbeat set up to
switch ip's around. The server actually listens on the fly to the new
ip address that comes up under it.
--
Aaron Daniel
, not on a subset of available addresses.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 11, 2006 2:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Server
The problem is in the space. You've got it as ? scid)... In order
for the label to work, you need to get rid of the space. Make it
?scid) and it should work fine.
The error's in the details:
pbx_extension_helper: No such label ' scid' in extension 's' in
context 'macro-dosomething'
On Tue,
Temporary Safety, deserve neither Liberty nor Safety.
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with the 7.4 firmware. It does exactly what we need, doesn't
decide it doesn't want to forget about registration if the server falls
out from under it, and doesn't have the server name in the caller id.
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-
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Whoops, sorry about that :) all I got in my client was Sean... ignore
that message I sent.
On Wed, 2006-07-05 at 15:10 -0500, Aaron Daniel wrote:
LOL, that's descriptive :)
On Wed, 2006-07-05 at 16:03 -0400, Sean Cook wrote:
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We
in there.
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of a coder, so I'm sure I screwed
something up).
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On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote:
How about fixing realtime SIP so that multiple Asterisk boxes can reference
the same database?
Doug.
That's kinda what I'm hoping to work towards :)
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[EMAIL
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maxlogins =3D 3;
if ((s =3D ast_variable_retrieve(cfg, general, maxlogins))) {
if (sscanf(s, %d, x) =3D=3D 1) {
maxlogins =3D x;
} else {
ast_log(LOG_WARNING, Invalid max failed login attempts\n);
}
}
And yes, that should be documented
?
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://bugs.digium.com/view.php?id=7443
The one on the bugs site is for SVN, but I do have a version that works
on 1.2.? (only tried on 1.2.9.1, so it may work on older versions). So
far it seems to work well.
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between them, with the MWI script and you'll
have fixed the redundancy problem.
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gas on the fire. df -h doesn't continuously poll the
drive for data, asterisk is (for mwi). So each timeout turns into another
timeout. Didn't you already test changing the time on checkmwi? And did
it not change the behavior (not necessarily for the better)?
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Computer
Dunno if anyone else has seen this yet:
http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/
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.
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On Thu, 15 Jun 2006, Douglas Garstang wrote:
DUNDi does not handle the situation of phone failover as well as static numbers
(ie queues), which is what we are trying to acheive.
I'm confused, explain the phone failover not working to me.
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Sam
the server to dundi.conf. So, so pbx1 has to be in pbx1's
file, just like the other servers do.
Make sense?
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the same except it has an outkey of scm2. This
should fix your issue with having dundi lookup on the local machine.
I'm not gonna try to understand your ACD stuff right now, so I'll just
figure you need DUNDi for that and give up on it :) Too busy fixing the
voicemail app.
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?
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Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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with DUNDi for over 6 months and still can't
figure it out past the most basic application.
What are you trying to do?
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
.
If that doesn't work for you, program the routes directly into the
dialplan instead of using DUNDi, it seems like you'll get better results
that way. We did that for a while until we decided to move to DUNDi.
Some people will find it more suited to their needs, some won't.
--
Aaron Daniel
-users
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Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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(SIPdefault.cnf and SIPMAC.cnf)
good luck!
-- Original message --
From: Aaron Daniel [EMAIL PROTECTED]
You have to press settings, then **#, and wait a moment to make sure it
unlocks. Then you can configure a tftp server to use.
The alternative is to configure your
hang up even though the call has ended, they just do the
congestion symbol. It's caught me off guard a couple times.
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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/asterisk
svn update
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Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose it
to anyone else. If you received it in error please notify us immediately and
then destroy it.
--
Aaron Daniel
Computer Systems Technician
: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, June 02, 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Config Revision Control
No, if you do an svn co
http://svn.server.com/svn/configs/trunk asterisk
in /etc, it'll make a folder called
Has anyone successfully gotten two separate servers to handle voicemail on
an NFS shared mount? The main thing I'm concerned about at this point is
keeping both systems from writing the voicemail file to the same
filename... any thoughts?
--
Aaron Daniel
Computer Systems Technician
Sam
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