Func_odbc is your friend.
Check out func_odbc.conf for odbc access from the dialplan
Check out res_odbc.conf to allow you to use odbc as a realtime source
Julian.
Callum McGillivray wrote:
I was hoping for something more along the lines of the Asterisk CMD
MySQL().
I could always resort to s
nik600 wrote:
i have a PRI connected to a TE205P.
Actually, can i send and receive FAX through Asterisk using stable
solutions?
Or shall i connect an ATA to Asterisk and then a modem with Hylafax?
I would suggest that using IAXmodem with HylaFAX would be more stable
than using an ATA-co
I was hoping for something more along the lines of the Asterisk CMD MySQL().
I could always resort to something like that.. but I don't want to "run"
it on a windows server and I really don't want to go to the bother of
writing FastAGI scripts to make it all happen.
I just want to write a qui
Oh. Got it now. Well, in this case I think you are looking at it
backwards. I imagine most users with this requirement write AGI
scripts that talk to their databases then communicate back. You can
use FastAGI and run your code on a windows server, or you could use
any other programming language (P
Oh Microsoft SQL Server for those unfamiliar with the term M$ ;)
mitcheloc wrote:
I've never heard of M$ SQL Server?
On 4/22/07, Callum McGillivray <[EMAIL PROTECTED]> wrote:
Hi all,
Has anyone successfully set up asterisk to query a M$ SQL Server?
I'd like to be able to query one in the
I've never heard of M$ SQL Server?
On 4/22/07, Callum McGillivray <[EMAIL PROTECTED]> wrote:
Hi all,
Has anyone successfully set up asterisk to query a M$ SQL Server?
I'd like to be able to query one in the dial plan and use the results to
tamper with call priorities / CLID etc.
If someone co
Crazy Boy wrote:
If IPhone is released in India, Can you tell me any Apple authorized
showroom in Hyderabad (Andhrapradesh, India)?
Oh gosh... another troll... Google IS your friend:
http://www.google.com/search?q=apple+iphone
___
--Bandwidth and Colo
Well this is the user's list. Apparently I'm a jacka**. Time for bed.
On 4/22/07, Sean Bright <[EMAIL PROTECTED]> wrote:
Try the A$teri$k user'$ li$t.
On 4/22/07, Callum McGillivray <[EMAIL PROTECTED]> wrote:
>
> Hi all,
>
> Has anyone successfully set up asterisk to query a M$ SQL Server?
>
Try the A$teri$k user'$ li$t.
On 4/22/07, Callum McGillivray <[EMAIL PROTECTED]> wrote:
Hi all,
Has anyone successfully set up asterisk to query a M$ SQL Server?
I'd like to be able to query one in the dial plan and use the results to
tamper with call priorities / CLID etc.
If someone could
Hi all,
Has anyone successfully set up asterisk to query a M$ SQL Server?
I'd like to be able to query one in the dial plan and use the results to
tamper with call priorities / CLID etc.
If someone could point me to a howto / guide or relate their experiences
with this, that would be great !
Hi FOlks,
I am using for research purposes Kvin's codecs available at
http://kvin.lv/pub/Linux/Asterisk/
G729 is working very well but g723 has a very poor audio quality.
I recompiled everything with gcc4 and the distro used is Slackware 11.
Anyone with some experience on that?
Thanks in adv
On Mon, Apr 23, 2007 at 01:49:12AM +0200, Hans Witvliet wrote:
> Hi all,
>
> Just curious,
>
> Quite a while a go, i was checking for supported SW-platform.
> AFAIR, it was RHES and SLES
>
> Now it's only RHES-4 and FC-3 or FC-4.
> Not a single syllable about CentOS or SLES-9 or SLES-10
>
> It
On Sat, Apr 21, 2007 at 08:59:27AM +0100, Senad Jordanovic wrote:
> What about creating a configuration file on server for each soft phone
> extension automatically and then importing that file into the soft phone?
>
> In another words, user receives a link to the setup program and the
> configur
meaning...?
Is it an option?
2007/4/22, Brandon Kruse <[EMAIL PROTECTED]>:
Please go to bugs.digium.com and make a bug, but no, it writes the agents to
users.conf
make sure to "activate changes" also
There is also a current overhauling of the queues/agents in the gui anyways.
-bkruse
Hi all,
Just curious,
Quite a while a go, i was checking for supported SW-platform.
AFAIR, it was RHES and SLES
Now it's only RHES-4 and FC-3 or FC-4.
Not a single syllable about CentOS or SLES-9 or SLES-10
It probably just runs fine, but any chance of getting support for their
*-enterprise ver
Hi,
I have spend allot of time searching a solution: We have different SIP
accounts that our Asterisk registers to, for example:
[general]
port=5060
disable=all
allow=[...]
srvlookup=yes
pedantic=no
context=start
language=de
register => 0123456789:[EMAIL PROTECTED]/
Problem 1:
Incoming calls
[EMAIL PROTECTED] wrote:
Date: Sun, 22 Apr 2007 19:38:04 +1000
From: Rob Hillis <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Softphone that supports central
provisioning?
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[EMAIL PROTECTED]>
Content-Ty
> "SU" == Steve Underwood <[EMAIL PROTECTED]> writes:
SU> G.729 isn't the best. Its just the one you need to be compatible
SU> with the other end. G.729 is the lock-in choice, not the quality
SU> choice.
What is the best codec with asterisk on a slightly lossy link (0.1%
packet loss), if band
Chris,
You might want to rethink your dishonest strategy.
What happens if the legitimate buyer of that card tries to register
his new card and get the HPEC and Digium declines because the sn was
already used?
Tom
At 02:07 PM 4/22/2007, you wrote:
> It seems that I need the serial number to
> It seems that I need the serial number to get a free copy of HPEC... but
> unless someone can convince me otherwise, I have a feeling it would just
> be easier to shell out the $10 per channel to avoid the downtime and
> drive out there.
Not that I'd normally encourage cheating the system, but c
Hello,
I'm at a loss for a way to find the serial number of a Digium analog
card without physically removing it from the server. The only time I
have physical access to this particular installation is during business
hours and that's obviously a bad time to be taking a server down.
It seems
Arun Kumar wrote:
> I've configured my exten.conf for few exten. But I'm curious to know how
> long can be my exten like (exten => XXX.). Is there any limit for
> this or not.
For asterisk the answer is in include/channel.h:
---cut---
#define AST_MAX_EXTENSION 80 /*!< Max lengt
On Fri, 2007-04-20 at 00:43 +0200, Remco Post wrote:
> Hans Witvliet wrote:
>
> > The only obstacles currently, are the ISP's.
>
> Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well as
> an ipv4 address.
>
> > afaik, all dsl-modems currently can only work with v4.
> > (correct
On Sat, 2007-04-21 at 11:30 -0700, Ira wrote:
> At 02:31 PM 4/19/2007, you wrote:
> >Fridge (sending snmp traps if a dork leaves the door open ;)
>
> But will it tell you when the person who put in the box that's
> holding the door open now slammed the door on that box instead of
> putting it in
Maybe you have too short a digit timeout.
l.
In data Sun, 22 Apr 2007 11:39:59 +0200, Poul Moller
<[EMAIL PROTECTED]> ha scritto:
Getting better... however still l didn't managed to transfer a call from
my
ATA. As you see some digits new gets recognized but never the full
extension
(100
Hello there,
you should check how the ATA encodes DTMF tones (eg rfc2833), and that you
have the same setting in sip.conf.
l.
In data Sat, 21 Apr 2007 18:21:34 +0200, Poul Moller
<[EMAIL PROTECTED]> ha scritto:
You are right kind of.
I tried from an IP SIP phone and it worked. The
Hi,
I've configured my exten.conf for few exten. But I'm curious to know how
long can be my exten like (exten => XXX.). Is there any limit for
this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my
hard phone to make calls. when my exten length is 14 then calls goes
Getting better... however still l didn't managed to transfer a call from my
ATA. As you see some digits new gets recognized but never the full extension
(1002 in my case). The # however always correctly triggers the transfer IVR.
/Poul
SIP/1003-08e68090 is ringing
-- SIP/1003-08e68090 answered
I put such a request for enhancement in sometime, and as is seeming to
be frustratingly common for CounterPath, it was completely ignored.
Were it not for the Plantronics CS-50 headsets that we bought that have
support in a /very/ limited number of softphones, I'd be dumping EyeBeam
/and/ X-Li
Alain Degreffe wrote:
Why do you use Ulaw as codec ?
Try another codec ( g729 is by far the best but isn't free ).
G.729 isn't the best. Its just the one you need to be compatible with
the other end. G.729 is the lock-in choice, not the quality choice.
Steve
__
On 21 Apr 2007, at 18:37, Salvatore Giudice wrote:
Most large enterprises (25k+ employees) would rather have a product
backed
by a real vendor, are not willing to switch office workers to
linux, and do
not see IAX as a viable option. Many customers avoid things like
IAX form
fear of being
Hi all,
I want to pass the incoming SIP callerid in Dial application:
Asterisk 1.2.13
sip.conf:
register => user:[EMAIL PROTECTED]/ext
extensions.conf:
exten => ext,1,Dial(SIP/phone1&SIP/phone2)
on phone's display I see the 'ext' number, not the incoming SIP callerid as
can be seen on incom
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