I'm trying to configure DUNDi in a tiered arrangement where 3 servers take care
of registrations and then push those registrations up to a pair of location
servers. Subsequent queries for the location of a phone will be direcected to
the location servers.
I think it's possible, but you do
It
won't work, unless you make sure that transfers go through the same asterisk
server as the orignal call went through. Using the SER dispatcher won't fix
that.
-Original Message-From: sip
[mailto:[EMAIL PROTECTED]Sent: Wednesday, September 27, 2006 2:25
PMTo: Asterisk
Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] SER with multiple asterisk deployment
Douglas Garstang wrote:
It won't work, unless you make sure that transfers go through the same
asterisk server as the orignal
I'm trying to get moh working correctly in Asterisk 1.4. A complete lack of
documentation isn't helping much.
I have this in sip.conf:
[3254101]
type=friend
...
mohsuggest=class1
[3254102]
type=friend
...
mohsuggest=class2
A call is bridged between the two extensions. When 3254102 puts
Ok, so does anyone know who the contributor of the new moh code is into
Asterisk 1.4? I'll email them directly.
Doug.
-Original Message-
From: Douglas Garstang
Sent: Tuesday, September 26, 2006 8:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk
I'd like to know if anyone has sucessfully managed to run multiple instances of
Asterisk on the same system.
- Did you run each instance as a separate user?
- Did you have any install or config problems?
- It looks like the G729 codec registration utility doesn't work when files
aren't
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local
lookup to see if a number is available locally, in order to find out if the
number is available on one of the other 9 servers, this peer has to query all 9
remaining peers.
Is that true?
Is there a way to have
I just downloaded asterisk 1.4beta2, and did a:
./configure --prefix=/home/pbx/1.4
[11:[EMAIL PROTECTED](pbx1):asterisk-1.4.0-beta2]# ls /home/pbx/1.4
bin include lib sbin share
What happened to etc? If I do a 'make samples', the default conf files get
thrown in /etc/asterisk.
Doug.
this, as it already has features for
multi-client configuration within a single Asterisk
installation/process.
Douglas Garstang wrote:
I'd like to know if anyone has sucessfully managed to run
multiple instances of Asterisk on the same system.
- Did you run each instance as a separate user
-Original Message-
From: Brian Rogan [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
Doug,
Why do you want to do this to begin
running multiple
instances, or mail me off-list if no one else is interested.
Thanks,
On 9/25/06 1:52 PM, Douglas Garstang
[EMAIL PROTECTED] wrote:
-Original Message-
From: Brian Rogan [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 12:40 PM
To: Asterisk
I've noticed sla.conf in Asterisk 1.4. I'd love to test it, but how does it
work? There's bupkiss docs, and until I have a clue how to use it, I can't test
it.
Doug.
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asterisk-users mailing
I made a post to the list on 8/7 and said that music on hold in Asterisk 1.2
seemed broken. It seemed like the moh class that should be used when party A
puts party B on hold, should be the class defined for party A. Asterisk 1.2
does it the other way around and uses party B's moh class.
Olle
Oops I got that a bit mixed up. I meant to say that when party A put party B on
hold, the moh class used should be party A's moh. Essentially the moh class
used should be what's defined for the person putting the OTHER party on hold.
Doug.
-Original Message-
From: Douglas Garstang
:
Subject: Re: [asterisk-users] MOH in 1.4 - Still Broken?
- Douglas Garstang [EMAIL PROTECTED] wrote:
Oops I got that a bit mixed up. I meant to say that when party A put
party B on hold, the moh class used should be party A's moh.
Essentially
So, is this GUI you speak of so often able to cater to CARRIERs rather than
ENTERPRISEs?
-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED]
Sent: Tue 9/19/2006 10:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Thanks for the plug Aaron. :)
We stuck OpenSER in between the phones and Asterisk, and pointed our phones
towards the OpenSER boxes for SIP registrations and subscriptions. When OpenSER
received a REGISTER or SUBSCRIBE message, it would use the send() command to
forward the messages onto each
Is anyone seeing any weird stuff with the latest Polycom 2.0.1 SIP application
software?
A few of our phones, after upgrading would come up with a 0x4000 Configuration
Error. Rebooting again a couple of times, or doing a 'Format Local Filesystem'
seemed to fix it, with no change to the config
attention, so I just thought I had done something.
On 9/20/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Is anyone seeing any weird stuff with the latest Polycom 2.0.1
SIP application software?
A few of our phones, after
:
Subject:Re: [asterisk-users] Digium GUI?
Someone already said that they saw it at VON. It was super simple to
change the look and branding but the UI itself was nothing too special.
Thanks,
Steve
Douglas Garstang wrote:
I wonder if the look and feel of this GUI will be completely
I don't think you can set a default volume, but you can configure the handset
(and headset) volume to persist between calls. Look for 'persist' in sip.cfg or
phone1.cfg.
Doug.
-Original Message-
From: Damon Estep [mailto:[EMAIL PROTECTED]
Sent: Tue 9/19/2006 9:23 AM
To:
Versions 1.6.x supported NAT as well.
-Original Message-
From: Matt Florell [mailto:[EMAIL PROTECTED]
Sent: Mon 9/18/2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Polycom
Poly 2.0.1 says it can do 48
On Sep 17, 2006, at 8:06 PM, Douglas Garstang wrote:
As far as I know, it's 12.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED
Yes, it initiates a completely new call. No sending DTMF in the current call
I'm afraid.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Sun 9/17/2006 2:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
I wonder if the look and feel of this GUI will be completely configurable. If
it's not, then I really don't think that's very useful. Service providers
wouldn't be able to use it to let their customers manage their own settings,
and customers wouldn't want to use it if it wasn't branded with
IAX has some pretty severe limitations when it comes to trunking calls between
Asterisk boxes. It can't pass variables for example, and any calls to SIP
phones at the far end will be treated as IAX calls, which is just nuts. This
means you lose a lot of SIP features, like transferring and
As far as I know, it's 12.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Sun 9/17/2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Polycom Expansion
: [asterisk-users] Sphinx2
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
The docs at that URL say that the dictionary has 'yes' in
it... although I don't understand how I can get replies like
'YOU HALF' if it doesn't exist in the dictionary.
Did you read
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
What sphinx documentation? All I could find was docs on the
code, not on how to USE the software.
:) One and the same!
Sphinx is not a commercial application.
I think you might have been mistaken, and are actually
-Original Message-
From: Norris, Sam [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 14, 2006 8:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BLF across asterisk trunks
2 asterisk boxes connected via SIP trunks.
Is there any way to subscribe to BLF on
Is there any documentation, maybe at voip-info.org, available for Asterisk 1.4?
Either in the form of _new_ docs, or docs that outline the differences and new
features that will be available in 1.4?
I'd like to avoid the months of trial-and-error that I went throo with 1.0, 1.2
if I can...
Anyone played with Sphinx2 and Asterisk much?
I followed the docs at:
http://turnkey-solution.com/asterisk-sphinx.html
and after getting the server up and running, streamed it some wav files with
the client.
It's interpretation of the word 'yes' from two different people was 'YOU HALF'
and
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sphinx2
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
Anyone played with Sphinx2 and Asterisk much?
I followed the docs at:
http://turnkey-solution.com/asterisk
Has anyone ever gotten the Polycom MyStat soft-key to do anything?
Setting the status to something like 'Away', does not generate any outgoing SIP
traffic from the phone. Calling into the phone either from a watched buddy, or
other number, acts as if the status was never changed. A call to
-Original Message-
From: Adam Goryachev [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 12, 2006 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Soundpoint Key Remap
Shawn Kelley wrote:
Hi,
Does anyone know how
Has anyone ever gotten the Polycom MyStat soft-key to do anything?
Setting the status to something like 'Away', does not generate any outgoing SIP
traffic from the phone. Calling into the phone either from a watched buddy, or
other number, acts as if the status was never changed. A call to
The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it yesterday
on 2.0.1
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 12, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
I'm pretty sure you can't do that. You can map a key to perform a single
function, such as perform the operation of another key, or dial a SINGLE digit,
but you can't make it dial a series of digits.
Doug.
-Original Message-
From: Matt Birmingham [mailto:[EMAIL
Polycom phones send a SIP SUBSCRIBE message for buddy watching.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 07, 2006 4:15 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] New polycom firmware /
I'm wondering if this is a bug in voicemail...
User A has elected to receive email notifications of voicemail and also have
the original voicemail deleted from the server, such that the WMI light is
never lit. If user B forwards a voicemail to user A (via the option in
voicemail), then user A
Polycom are analy retentive about giving out software updates.
-Original Message-
From: Nathan Alberti [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 07, 2006 10:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom new firmware
to get you the files with out upsetting Polycom.
On 9/7/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Polycom
are analy retentive about giving out software updates.
-Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED]] Sent:
Thursday
[EMAIL PROTECTED]
Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/
On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote:
That process is worse than pulling teeth!
-Original Message-From: Jessee J Holmes [mailto:[EMAIL
If you want to use MWI, and I imagine most people would, you have to cache your
realtime data, which means that changes to the tables do not become effective
immediately. They become effective after you prune the entry in memory.
Doug.
-Original Message-
From: RR [mailto:[EMAIL
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: 30 August 2006 04:05
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?
That might not be a good idea
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 30, 2006 1:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SER Dispatcher Load Balance How-To?
Andy Chung (Power-All) wrote:
Hi all,
I
-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 How-To?
Well, it really depends on what he's using the
That might not be a good idea. If you transfer or forward calls on your phones,
you MUST make sure the transferred or forwarded call goes back to the same
Asterisk box it was handled on. If you use the dispatcher, and load balance,
there is no guarantee of that, and transfers and forwarding
is there any alternatives?
Thanks!
Andy
Douglas Garstang wrote:
That might not be a good idea. If you transfer or forward calls on
your phones, you MUST make sure the transferred or forwarded call goes back to
the same Asterisk box it was handled
why it is working but it is. My first line I have in
extensions.conf and the second I have in MySql.
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 24
that each time u create a new one you can just
add it to the
DB.
Dovid
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 25, 2006 11:09 AM
Subject: RE
, 2006 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime and hints
Doug I have
Exten = 10,hint,SIP/11010
and in mysql I have
exten = 10,1,Dial(SIP/11010)
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED
Subject: RE: [asterisk-users] Realtime and hints
That's what he was gettin at. Take the second line out, and put the
first priority in the database.
On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote:
But... you need _both_ in your dialplan.
My extensions.conf has
Title: Message
We had
a similar problem. Eventuallywe gave up and just used apache. We found
that _exactly_ the same content would not work with IIS, but WOULD work with
Apache.
-Original Message-From: Phil Menico
[mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06
-Original Message-
From: Matthew Crocker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions -- Comments?
Add a boolean field to the table then create a
: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 22, 2006 12:14 PM
Subject: RE: [asterisk-users] Realtime Extensions -- Comments?
-Original Message-
From: Jason Parker [mailto:[EMAIL
The unofficial docs on the voip wiki for the realtime extensions table
structure is:
CREATE TABLE `extensions_table` (
`id` int(11) NOT NULL auto_increment,
`context` varchar(20) NOT NULL default '',
`exten` varchar(20) NOT NULL default '',
`priority` tinyint(4) NOT NULL default '0',
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 2:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Apache for FastAGI
Douglas Garstang a écrit :
Here's
I'm not sure how one would build a HTTP header on the client side, given that
all you have to work with is a single line entry in extensions.conf.
-Original Message-
From: Tielin Xu [mailto:[EMAIL PROTECTED]
Sent: Friday, August 18, 2006 12:41 PM
To: asterisk-users@lists.digium.com
Yes,
the config files are the same across a single version of the SIP application
software for 301/501/601. If you upgrade however, you'll need to use the new xml
files supplied with that version. Polycom needs to get a clue and either make
the files backwards compatible or provide a
be included in the dial-plan. Later, all
you have
to do is change the context back with a simple SQL UPDATE statement.
Douglas Garstang wrote:
The unofficial docs on the voip wiki for the realtime
extensions table structure is:
CREATE TABLE `extensions_table` (
`id` int(11) NOT NULL
for default sip settings) which are then joined
together
in various ways to produce views for the extensions and a sipdevices.
Simon
On 22 Aug 2006, at 15:20, Douglas Garstang wrote:
The unofficial docs on the voip wiki for the realtime extensions
table structure is:
CREATE
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions -- Comments?
Douglas Garstang wrote:
Uhm... what abouts
-Original Message-
From: Jason Parker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions -- Comments?
- Douglas Garstang [EMAIL PROTECTED] wrote
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Monday, August 21, 2006 3:39 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisk 'Hosting'
JM == Jeremy McNamara [EMAIL PROTECTED] writes:
JM Why do you need multiple instances?
How where you able to interact with the callee after they had answered the
call? You lose control of the dial plan after someone answers, until they hang
up.
-Original Message-
From: Roy Kidder [mailto:[EMAIL PROTECTED]
Sent: Monday, August 21, 2006 5:05 AM
To: Asterisk Users
Does anyone know if realtime extensions support the use of labels?
ie:
exten = acdpause,1,Answer
exten = acdpause,n,Wait,1
exten = acdpause,n,PauseQueueMember(|Agent/${CALLERIDNUM})
exten = acdpause,n,GotoIf($[${PQMSTATUS} =
Can realtime be used with hints? How would you get the following into the
database given that the priority column is numeric, and that there is no
application for the first entry?
exten = 2944006,hint,SIP/2944006
exten = 2944006,1,Dial(SIP/2944006)
Every time I touch realtime I hit obstacles.
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 20, 2006 5:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Apache for FastAGI
On Sun, Aug 20, 2006 at 04:18:11PM -0600, Douglas Garstang wrote:
I'm not sure there's
ngrep is also good if you only want to see SIP traffic and filter all the lower
level stuff.
-Original Message-
From: Brandon Galbraith [mailto:[EMAIL PROTECTED]
Sent: Mon 8/21/2006 8:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Over the last few months, I've peeked at the web site a few times and there was
_two_ jobs (the same ones every time). Now, there's _zero_. Is that likely to
increase any time soon?
-Original Message-
From: Matt Gibson [mailto:[EMAIL PROTECTED]
Sent: Sun
Message-
From: Anders Nygren [mailto:[EMAIL PROTECTED]
Sent: Fri 8/18/2006 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Apache for FastAGI
On 8/18/06, Douglas Garstang
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Friday, August 18, 2006 5:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Recent additions to the Digium
Asteriskdevelopment team
I also want to add that I saw
Can someone tell me what this is about? Asterisk seems to be 'losing' peers.
Usually when a peer isn't known (such as when you first start Asterisk),
Asterisk will do a database lookup and find the peer, and then seed them.
I tried to dial 3254101, and I get the error below. I ran an ngrep and
Here's an idea...
Rather than writing your own multi-thread socket server for use with FastAGI,
has anyone tried to use an Apache web server instead? After all, it does all
that for you. I just gave it a shot, but Asterisk tries to send all the agi
params to the web server, which it doesn't
I tried this...
[test-in]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
It didn't work. The select that Asterisk sent to the database was:
SELECT * FROM extensions_table WHERE exten = '1001' AND context = 'test1' AND
priority = '1'
So, it obviously ignores the second
, Douglas Garstang [EMAIL PROTECTED] wrote:
Here's an idea...
Rather than writing your own multi-thread socket server for
use with FastAGI, has anyone tried to use an Apache web
server instead? After all, it does all that for you. I just
gave it a shot, but Asterisk tries to send all the agi
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 11:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote:
Oh, and I see nufone caters
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote:
Good luck with supporting enterprise
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote:
That's funny. I remember asking
Does realtime support include = yet?
Thanks,
Doug.
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote:
What's not specific about
I'm curious, have always been curious... why does realtime repeat the same
query multiple times when lookup up extensions? This is one of the reasons we
chose not to use it, but I'm still wondering why it does it?
[14:[EMAIL PROTECTED](pbx1):asterisk]# ngrep -d eth0 port 3306 | grep SELECT
-Original Message-
From: Douglas Garstang
Sent: Thursday, August 17, 2006 2:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk 'Hosting'
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 2:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 'Hosting'
On Thursday 17 August 2006 16:04, Douglas Garstang wrote:
Yet again you have
-Original Message-
From: Douglas Garstang
Sent: Thursday, August 17, 2006 2:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk 'Hosting'
-Original Message-
From: Douglas Garstang
Sent: Thursday, August 17
PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, August 17, 2006 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk 'Hosting'
-Original Message-
From: Douglas Garstang
Sent: Thursday, August 17, 2006 2:17 PM
Ok, maybe I'm having a brain fart, or maybe I've never gotten quite this far,
but, if you call a fast AGI script, how do you RETURN data from the fast AGI
back to the dialplan???
Doug.
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-Original Message-
From: Justin Tunney [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Return data from Fast AGI
On Thursday 17 August 2006 17:12, Douglas Garstang wrote
mailx?
-Original Message-
From: Damien Gabrielson [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Sending Email From A Dial Plan
Hi,
I'm looking for a simple way to send
-Original Message-
From: Anders Nygren [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Return data from Fast AGI
On 8/17/06, Douglas Garstang [EMAIL PROTECTED] wrote
-Original Message-
From: Roger Schreiter [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 4:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Return data from Fast AGI
Douglas Garstang schrieb:
...
It doesn't even have
*lol* The cryptic replies have been exactly my problem as well!
-Original Message-
From: kjcsb [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
-Original Message-
From: kjcsb [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
Absolutely. The SER/OpenSER documentation is terrible, and if
How
did you find out about 468*??? It's sure as poop not documented in the Polycom
Admin Guide anywhere.
-Original Message-From: Dovid Bender
[mailto:[EMAIL PROTECTED]Sent: Tuesday, August 15, 2006
11:16 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject:
Has anyone ever tried to run multiple instances of Asterisk on a single system,
running each with a different username, and each in a separate base directory?
Something like /home/pbx/business-1, home/pbx/business-2 etc?
Did it work? I assume for every service that Asterisk runs, on each
-users] Asterisk
'Hosting'You might be able to use virtual NICs to
eliminate the problem with "non-standard" ports for a company's SIP
phones. Or real NICs using a couple of multi-homed cards.I
haven't tried it, though.
On 8/16/06, Douglas
Garstang [EMAIL PROTECTED]
wr
-Original Message-
From: Matt Riddell (NZ) [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote:
Has anyone ever tried to run
, if one asterisk box
goes down, you don't have 50-100 clients completely
down.-brandon
On 8/16/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Well, we're talking about
several dozen, maybe 100, companies, per Asterisk box
here.
-Original
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