Take a look on the Digium site for the requirements of the card. Most
likely the old computer can't handle the card.
M.Hockings wrote:
I just bought an TDM400P card with one FXS port and a X101P FXO card
to try and put together the beginnings of a PBX here. The computer
they are going in
Hi Alex,
Thanks for the clarification.
As a user i was just trying to give a caution before the group lands
up in trouble.
Not all that shines is Gold. Just trying to be good.
Cheers
Bye
On 19/05/06, Alex Robar [EMAIL PROTECTED] wrote:
Hmm, I don't know... A user just looking for a DID
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Cisco have released V8.3 sip firmware for the 7940/60 (currently in the
.cop format for CCM 5). I tried this briefly this afternoon only to
find that the phone fails to register with Asterisk (1.2.6
realtime/mysql). As I did not have enough time to
This worked for me yesterday: -Please replace your actual extension number where it says extensionnumber and password in passwordOn asterisk:[extensionnumber]
Hi,
An Cisco 7960 ipphone has been set to SCCP firmware
by one of our students.
I want to set it to 7.5 SIP firmware and I've been
unsuccessful yet.
Firmware versions are SCCP 3.0 (Source:
On Fri, 19 May 2006, Alexander Lopez wrote:
For me the answer is simple:
1 Not open source, IT is a thrill to be able to say what you did
without Closed SW.
The drivers are GPL'ed open source. If your are referring to the firmware,
then yes, it is closed, but why would you need that
You changed your default SIP (bindport) port to 5061 at the server, so
your client needs to look there.
Try like these
register = sipteszt:[EMAIL PROTECTED]:50/sipteszt
bindport=5061 ; UDP Port to bind to (SIP standard port
is 5060)
Adrià Vidal
Does anybody know of any IP phones (ideally SIP based) that have
interfaces to plug into a pro audio system (eg for phone interviews).
Something can probably be hacked up with a headset connector or the 1/8
jacks on a 7970 but I'm wondering if there's something better out there.
Thanks,
Julien
Hi All,
I am happy to offer $1000USD for the fix of the g726-32 in Asterisk.
What's wrong with it? It currently gives a very distorted sound as
though the gain is set to high. Lowering the gain on endpoints helps
but this is not a fix just a poor workaround. We require g726-32 to be
of the same
I could not find anything so I whipped up something myself using an
M-Audio FireWare Box, It gives my 4 distinct analog audio channels as
well as two inputs. It also has SPDIF (optical and Coax). I then paired
it with the SNOM softphone. I am also able to record to an external DAT
or MD, or feed
On 5/18/06, Remco Barende [EMAIL PROTECTED] wrote:
Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy
a dual port ethernet adapter which will use only one irq to free up an IRQ
on another slot. This just totally sucks and irq sharing in a box with
only 3 pci slots is
Hi All,
I am happy to offer $1000USD for the fix of the g726-32 in Asterisk.
What's wrong with it? It currently gives a very distorted sound as
though the gain is set to high. Lowering the gain on endpoints helps
but this is not a fix just a poor workaround. We require g726-32 to be
of the same
Olivier
Complete these steps:
Go to the main Date/Time screen.
Press **# to unlock Network Configuration on the phone.
Press Settings.
The Network Configuration lock symbol should be unlocked. If it is not,
exit to the main screen and press ** # again.
Press 3
Hello all!
I have a problem with ringing indication when calling from h323 (oh323+open
phone client) to sip users. The phone rings and users can talk to each other
with no problems but the calling h323 user hear silence unless sip user picks
up the phone.
Calling to pstn no problems. Calling
On 17 May 2006, at 22:48, Andy Kirby wrote:
I am new to the group but have searched the doc's FAQ's etc before
posting here.
We are attempting tie our asterisk server/service to the
building's PBX, the building is in the UK and the local PBX is a
meridian option 11 installed and
I'm looking for a method to signal an
insideextension (asterisk extension with external dialing appl.) with a
DTMF "A" tone to indicate when Asterisk has completed dialing and the voice
path has beencut-though on a ZAP T1 Trunk.
If this can be done, I'd also like to know if there
is a
Personally, I have to agree with Alexander on this...The single PRI card is $6k, I might as well get online and buy a used cisco as5300/5400, I can get those for $5-7k on ebay usually, they have DSPs, and take all the load off the server itself, and I'd have the added benefit of the thing being
On Wednesday 17 May 2006 12:29, Jan Pringels wrote:
I'm trying to compile Zaptel driver with the ztdummy. I have no hardware
cards from digium.
I tried following steps:
http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation
2006/5/20, Cory Andrews [EMAIL PROTECTED]:
Olivier
Complete these steps:
Go to the main Date/Time screen.How do you get to that screen ?It's crazy but when I plug my phone, the only thing I can see is :a black strip on top with 2 lines bellow showing :
Cisco Systems, Inc.Copyright
I believe its trying to get an IP. Put
it on a network, have a DHCP server up to give it a possible address, and see
if that changes its behavior.
Terrelle
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier Krief
Sent: Saturday, May 20, 2006 9:53
AM
To:
Kevin:
Thanks for the info, I think I will buy the video phones
Erick W.
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 19, 2006 6:18 PM
Subject: Re:
It passes Configuring VLAN.It gets its own and gateway IP addresses from DHCP server but remains stuck in Configuring IP...No FTP server is defined.I believe Network Configuration menu is locked because Configuring IP process in going on.
If I could find a mean to stop this process, maybe I could
As far as I know, you will still need a
tftp or ftp server in order to do that.
Is this phone setup to configure itself
from tftp/ftp server each time the phone is restarted? Then that would explain
why its stuck because the tftp/ftp server is removed.
Usually they are setup to get the
It passes Configuring VLAN.
It gets its own and gateway IP addresses from DHCP server but remains stuck
in Configuring IP...
No FTP server is defined.
I believe Network Configuration menu is locked because Configuring IP
process in going on.
If my memory serves me right from last time I did
You may need to use ethereal or something to see
what TFTP server address it is looking for, unless your are sure it's
undefined.
Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY 14225++voice - 716.630.1555
X22email - [EMAIL PROTECTED]AIM - B2CORY
-
Cory, Do you have the Nokia E70 and or the E60 ? If not are you guys gona get it in anytime soon ?DovidCory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote,wireless handsets via DECT.Cory AndrewsExecutive Vice
Ok, eventually I gave up on the version of asterisk in Ubuntu breezy
1:1.0.9.dfsg-1 and compiled it from CVS (asterisk, libpri and zaptel driver)
and all problems went away. It just looks like the code included in Breezy is
a bit broken.
Thanks for all the help.
Pieter
On Wednesday 17 May
You can check that info in www.asterisk.org or voip-info.org
If you have problems applying the patch let me know, may be I can make
you a patch for the 1.2.7.1 specially.
Regards
On 5/19/06, Obelix [EMAIL PROTECTED] wrote:
Quoting Moises Silva [EMAIL PROTECTED]:
Hi,
I am ready to try out
Quoting Moises Silva [EMAIL PROTECTED]:
You can check that info in www.asterisk.org or voip-info.org
I downloaded the trunk from SVN early this morning - I think I should have kept
the revision number. Is there some way of knowing whether it contains the
patch?
When I checked the Asterisk
On Fri, May 19, 2006 at 09:30:33AM +0100, Chris Hastie wrote:
I've just received an OEM Wildcard X100P FXO card. Installing into my
FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since
it's the first time I've put a PCI card in this machine I've just
dropped a Netgear
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Cisco have released V8.3 sip firmware for the 7940/60 (currently in the
.cop format for CCM 5). I tried this briefly this afternoon only to
find that the phone fails to register with Asterisk (1.2.6
realtime/mysql). As I did not have enough time to
I thought the price of E1 in Australia was quite reasonable, at least
compared to analog.
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
- Original Message -
From: Craig Guy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
The new grandstream video phone has rca-style audio jacks
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
- Original Message -
From: Julien Goodwin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Cosmin Prund wrote:
Take a look on the Digium site for the requirements of the card. Most
likely the old computer can't handle the card.
M.Hockings wrote:
I just bought an TDM400P card with one FXS port and a X101P FXO card
to try and put together the beginnings of a PBX here. The computer
In my attempt to setup a single FXS line I have been following the
instructions for Telephony Card Drivers on the asteriskdocs.org site.
I have managed to checkout, make and install the zaptel code and can
load the zaptel module but when I attempt to load the wcfxs module it
tells me:
Last time I checked with Telstra about 3 months ago, at 7 channels (eg
3.5 BRI services), a 10 channel E1 service (OnRamp10 from Telstra) is
cheaper than BRI in terms of monthly line rental (a fair bit more
expensive to install though). So if you actually need 4 ISDN ports / 8
lines to connect to
Snom 200 if you can get ahold of them has standard 1/8 headset jacks you
would use the Monitor() app to record.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, May 20, 2006 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
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