On ISDN lines it's possible to prohibit the
presentation of caller id, what if I have a SIP
gateway, something like an Audiocodes Mediant
1000. How do I prohibit the caller id presentation
on that one?
Regards,
Kristian
--
Kristian Larsson KLL-RIPE
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided
beleive Dundi should be able to help you out
in situations like this.
Kristian
-Original Message-
From: Kristian Larsson [mailto:[EMAIL PROTECTED]
Sent: Monday, March 13, 2006 12:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
--
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com
to
bring the packets to userspace and back limiting
the performance by quite a lot.
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation
per box I instead have
several boxes and thus better redundancy.
One card per box also guarantess I won't have any
interrupt problems and it's capable of
transcoding.
Kristian.
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
.
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com
: Number Unavailable
Thank you
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
, and it's rather cheap too :)
Kristian.
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
On Fri, Jan 20, 2006 at 09:20:43PM -0500, Michael Miller wrote:
I have over 50 Asterisk servers geographically distributed in pairs all
connected via DUNDi. Contact me off list and I will be happy to describe
my experience.
I'm also interested in knowing more of this. Why
not write to the list
Hey!
I'm having a small problem. I'm using Realtime to
store SIP account information. Dialing works just
fine, but when dialing a person already on the
phone I don't get a busy tone.
Eg, Phone 100 calls 200 and they chat with each other
phone 150 calls 100, and gets a regular ringing tone
what I
On Wed, Jan 18, 2006 at 03:44:03PM -0800, calvis wrote:
Sorry about the OT thread, but I am sure that someone could give me some
advice. Nothing is more frustrated than doing a long cable run and then
finding your cable is defective.
OK, I have had it with the General Cable brand of
On Wed, Jan 04, 2006 at 02:46:36PM +, Alistair Cunningham wrote:
Peter Bowyer wrote:
I was thinking along the same lines, but for a dynamic setup it should
be possible to have SER/OpenSER load balance REGISTER requests
according to some strategy/metrics, and then forward INVITEs and other
I'm trying to set caller presentation to
prohibited and I'm having slight problems doing
it.
Using a machine that has a Sangoma facing my Telco
works but when using an asterisk that talks to the
first machine using SIP it does not work.
I suspect that SetCallerPres is not transitive, ie
it's not
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
?
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
On Mon, Jan 02, 2006 at 03:36:57PM +0200, [EMAIL PROTECTED] wrote:
On Mon, 2 Jan 2006, Kristian Larsson wrote:
I have an Avaya IP Office PBX connected to an
Asterisk system via a Sangoma ISDN PRI card.
Dialing from the as
terisk system into the avaya works just fine but
when
:51 +0100
From: Kristian Larsson [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Mime-Version: 1.0
Content-Type: text/plain; charset=iso-8859-15
Content-Disposition: inline
User-Agent: Mutt/1.5.8i
X-Virus-Scanned: by amavis at citynet.nu
X-Spam-Status
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote:
On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote:
...
And this is bad for us. With Gizmo we can talk. With google talk we have
stand a chance of talking. But we're blocked from Skype.
since you cite it, what
On Tue, Dec 20, 2005 at 09:54:50AM +0900, [EMAIL PROTECTED] wrote:
I sincerely believe that it's completely non-sense to make a channel for
Skype.
Skype is a *proprietary* protocol. If they(ebay) don't like the idea of
someone messing around their network,
they will change the protocol
Just the other day I tried connecting an Avaya
IP403 Office IP PBX to my asterisk.
The IP403 is currently used for all the phones at
our office and it is connected via it's own PRI to
the PSTN.
Now I have a Asterisk machine with three PRIs used
for our SIP services. To be able to utilize our
would I go about doing this with Asterisk
RealTime. I have quite a few users and it would be
really great if there could be some form of
automatic extension adding.
Regards,
Kristian.
--
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
themselves?
I would be very interested in hearing more of such
solutions and people experiences with it.
Regards,
Kristian Larsson
On Thu, Dec 08, 2005 at 10:00:01AM +0200, Zoa wrote:
Yes,
transcoding is not going to work for that density.
asterisk doesn't do g723, and even if it would your
I am having problems with echo, first let me
explain my setup:
I have a Gateway box, which basically is an
Asterisk with a PRI card. It's only job is to
interface with 2 incoming ISDN PRI connections.
Then there are two other asterisk boxes to which
my users are registered.
Dialing from a phone
not currently on the list and
thank you.
Regards,
Kristian Larsson
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I have a an Asterisk server running asterisk 1.0.3 and a TDM04b card.
I'm having a problem with my setup. Incoming and outgoing calls are working to
95%.
When the other party hangs up their phone after I've hang up mine it starts
ringing in my phone. example:
1. I get an incoming call
2. I answer
Cisco came up with PoE before the standard was set and so it differs.
The polarity is switched, so using a dumb power injector and a crossed cable
one could make it work anyway.
Quoting Julio Arruda [EMAIL PROTECTED]:
Keith Burns wrote:
I think you need to look at a few other factors.
...
32 matches
Mail list logo
