Brian Capouch wrote:
ODBC only, no other DB access methods?
No, ODBC only. There is little benefit to using any direct access
methods for something like this (actually, even the Realtime drivers
work just fine with ODBC as well).
Once I get 1.2 out and can start working on res_storage, it
Kevin P. Fleming wrote:
Brian Capouch wrote:
ODBC only, no other DB access methods?
No, ODBC only. There is little benefit to using any direct access
methods for something like this (actually, even the Realtime drivers
work just fine with ODBC as well).
If ODBC just worked with the
Somewhere in the Wiki I read that the best way to adjust the rxgain and
txgain is to dial a type 102 milliwatt test line.
This line is usually found in xxx-958- or xxx-959- ranges.
I'm in area code 323 in Los Angeles.
Does anybody know the test number here??
The number assigned
On Thu, 17 Mar 2005 12:05:27 -0800, Kerry Garrison
[EMAIL PROTECTED] wrote:
Please ignore, intelligent messages will come later
Doesn't look like intelligent messages blah blah blah. You could
have waited to see the messages coming in, you didn't have to test it.
It's an idiotic thing to do.
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
This is a test, please disregard
Next time you post, make sure you turn off HTML.
And not that I expect anything productive to come of this part of a
rant... If you(collective mass of people sending test messages lately)
are testing
And what does that do?
I am not testing no filter, I am testing the change of my name as it shows
up on the list. I can't think of any other way. However, if turning off
HTML has the desired result, I thank you for the tip.
BTW: I can only dedicate 5 minutes to this issue. Thanks
CS
Discussion
Subject: Re: [Asterisk-Users] test-ignore
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
This is a test, please disregard
Next time you post, make sure you turn off HTML.
And not that I expect anything productive to come of this part of a
rant... If you(collective mass
, January 11, 2005 7:49 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] test-ignore
Dante and the windmill?
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent
]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, January 11, 2005 7:49 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] test-ignore
Dante and the windmill?
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax
]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] test-ignore
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
This is a test, please disregard
Next time you post, make sure you turn off HTML.
And not that I expect anything productive to come
On Tue, 2005-01-11 at 19:45 -0500, C. Savinovich wrote:
And what does that do?
I am not testing no filter, I am testing the change of my name as it shows
up on the list. I can't think of any other way. However, if turning off
HTML has the desired result, I thank you for the tip.
.
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
-Original Message-
From: C. Savinovich [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 11, 2005 7:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] test-ignore
On Thursday 28 October 2004 04:15, Steve Totaro wrote:
i think he meant numbers that would not be billed for completing a call.
No, any number is just fine.
Preferably a mix of mobile and fixed numbers for as many countries/regions as
possible.
So often a customer will say something like I've
Richard Bennett wrote:
On Thursday 28 October 2004 04:15, Steve Totaro wrote:
i think he meant numbers that would not be billed for completing a call.
No, any number is just fine.
Preferably a mix of mobile and fixed numbers for as many countries/regions as
possible.
So often a customer will say
/LookieLoo/VoicePulse/1 of our 6 Voice T1/PRI's,
whichever works best.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Thursday, October 28, 2004 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk
Paul Rodan wrote:
Yeah, I'd definitely be interested in that list too.
I thought I was one of the only find the right carrier game. Sometimes
customers can't place calls or have quality issues to certain countries,
sometimes even certain Area codes within the U.S, so I then route those
calls
On Thursday 28 October 2004 15:12, Andrew Thompson wrote:
When you get a decent size list, you will post it on the wiki(if you're
not doing it now), or at least mail it to any other interested parties,
right?
Yeah, the wiki would seem like the right place, so people can add numbers
Richard Bennett wrote:
Hi,
I was wondering if there already exists a list of worldwide test telephone
numbers for us to use to test if we can terminate that destination?
If not i'd like to suggest setting this up on the wiki.
I was thinking of a list of numbers sorted per country/type/operator,
Andrew Thompson wrote:
Richard Bennett wrote:
Hi,
I was wondering if there already exists a list of worldwide test
telephone numbers for us to use to test if we can terminate that
destination?
Hotels, restaurants and any other public place similar, searched for in
google
will provide you
--On Wednesday, October 27, 2004 10:51 +0200 Richard Bennett
[EMAIL PROTECTED] wrote:
For instance, if you want to test US termination you can dial the
movie-phone on 12127773456, or if you need to test Belgium mobile
Mobistar you can use 32494413965...
Any ideas?
I'm willing to setup a 'time
i think he meant numbers that would not be billed for completing a call.
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Wednesday, October 27, 2004 8:43 PM
Subject: RE: [Asterisk-Users
I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.
But just 5 minutes ago I moved a user over to g729a. Changed the
SIP000.cnf file for the Cisco phone, but forgot to change the
On October 19, 2004 10:47 am, Roy Sigurd Karlsbakk wrote:
does anyone know if it is possible to test-drive G.729 without paying
Digium for it?
You're too cheap to blow $20 (I assume you need 2 licenses) on a test?
Seriously it's not that expensive and if it doesn't work it doesn't work -- I
Deon Rodden wrote:
I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.
uLaw *is* telephone quality. Its what the PSTN uses. G.729 is much
inferior, but at 1/8th the size (not
/SNIP/
Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 different implementations of it. One
being where you paid license per channel fees, one that was
free/open source, and another I can't remember. Check the WIKI.
/SNIP/
There is no such
On Tue, 19 Oct 2004 11:13:02 -0400, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
/SNIP/
Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 different implementations of it. One
being where you paid license per channel fees, one that was
On Tue, 19 Oct 2004 15:59:03 -0400, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
Remember that we discussed this at length over about 3000 posts. I again
reiterate that it is called a HACK and I am Ok with what you are
doing. Kevin and Others in this forum has made the licensing point
Kanuri, Seshu (Company IT) wrote:
/SNIP/
Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 different implementations of it. One
being where you paid license per channel fees, one that was
free/open source, and another I can't
Nicholas Bachmann wrote:
Olle E. Johansson wrote:
The decision is to base the future 1.0-release on the CVS head tree.
The current stable-1.0 tree will be released as something intermediary,
maybe 0.91, and at that point it will be considered end-of-life.
At some point when we have cleared the bug
Youre making it now..
At 07:02 5/18/2004, you wrote:
none of my messages are arriving on the list... just testing.
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To UNSUBSCRIBE or update options
tmpm wrote:
Youre making it now..
Sorry... I actually didn't expect it to work.
My first resend from yesterday came through (twice) but my second
one doesn't (including the output from sip debug) - it seems the list
quietly drops long messages (14K in this case). I put it on
OMG I want to say If I find the person that has the mail server Never
mind its fixed now. But please check your mail filters to make sure they
bounce mail properly.
bkw
On Fri, 30 Jan 2004, Brian West wrote:
test
___
Asterisk-Users mailing
Hi, a Virus was sent from this account to the Asterisk-Users mailing list...
scan your computer for virus!!!
Ronen
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 7:29 PM
Subject: [Asterisk-Users] Test
The message contains Unicode
That is no longer a problem.
bkw
On Wed, 28 Jan 2004, Tazman wrote:
Hi, a Virus was sent from this account to the Asterisk-Users mailing list...
scan your computer for virus!!!
Ronen
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26,
On Wed, 2004-01-07 at 14:16, M. Matt Colgin wrote:
Test Post
Ohh noo, I read it. Anyone have the antidote?
--
Steven Critchfield [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 4:42 PM
Subject: Re: [Asterisk-Users] Test Post-Do Not Read
On Wed, 2004-01-07 at 14:16, M. Matt Colgin wrote:
Test Post
Ohh noo, I read it. Anyone have
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls from around
world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
sip:[EMAIL
Walker Haddock wrote:
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
Hi Miklos,
I have the same as Walker.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walker Haddock
Sent: 24 November 2003 18:02
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas
Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 4:02 PM
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test
Adding canreinvite=no to your sip.conf for that phone should do it..
I did have that in there. Here's the stanza in sip.conf. I set up device [90]:
[90]
context=from-sip
type=friend
insecure=yes
host=sipserver.com.br
fromdomain=home.datacrest.com
canreinvite=no
reinvite=no
nat=yes
: [Asterisk-Users] test call request
Hi Miklos,
I have the same as Walker.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walker Haddock
Sent: 24 November 2003 18:02
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] test call request
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 2:46 PM
Subject: Re: [Asterisk-Users] test call request
Adding canreinvite=no to your sip.conf for that phone should do it..
I did have that in there. Here's
On Mon, Nov 24, 2003 at 05:38:13PM -0200, listas iPfone wrote:
Hi !
Thank you for the call
I think that you have to Put reinvite=no in your sip.conf for the given
friend/user/peer to keep * from trying a native bridge.
I tryed to call you ( sip:[EMAIL PROTECTED] and
sip:[EMAIL
On Mon, Nov 24, 2003 at 03:59:14PM -0500, Andrew Thompson wrote:
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 2:46 PM
Subject: Re: [Asterisk-Users] test call request
Adding canreinvite=no to your sip.conf
Does anyone know if there is a dialplan in place that would allow
me to dial out via iaxtel (with a 700 number) and back into my
fwd number?
I've tested fine in the opposite direction, but would like to verify
the fwd incoming call success.
Rich
700-99-X where X is the 5 digit
Does anyone know if there is a dialplan in place that would allow
me to dial out via iaxtel (with a 700 number) and back into my
fwd number?
I've tested fine in the opposite direction, but would like to verify
the fwd incoming call success.
Rich
700-99-X where X is the
On 17/06/2003 at 10:23 Rushowr wrote:
Is it possible to set up Asterisk without any of the cards? I'm
interested in setting it up for the company I work for, but I would like
to set it up and see how difficult it will be before I start having the
company spend a chunk on equipment.
Yes, you can
Rushowr wrote:
Is it possible to set up Asterisk without any of the cards?
Absolutely. Use IP Phones and a VoIP termination provider... Hell, you
may not even need your own Asterisk box if your just looking to see how
simple it is to make an IP phone work.
Jeremy McNamara
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