On Thu, Feb 18, 2010 at 03:05:14PM -0500, Ken D'Ambrosio wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest problem, as
Thanks for the review, I was wondering if snom's mass deployment tools
they have
On Fri, Feb 05, 2010 at 01:21:38PM +, Nikhil Nair wrote:
Hi again,
OK, I've now installed a local caching nameserver, but don't see any
change at all.
Just to add to the discussion, my setup I was using a local bind9 server
for local/authorative and recursive queries
I think from
On Thu, Feb 04, 2010 at 09:52:35PM -0600, Warren Selby wrote:
On Thu, Feb 4, 2010 at 9:20 PM, Nikhil Nair nn...@pobox.com wrote:
No, again, I can cut off the internet altogether with ifdown eth1, and
the SIP phones (via eth0) continue to work fine, as does the Zap channel.
It's only if
On Sat, Jan 23, 2010 at 08:08:28AM +0100, Philipp von Klitzing wrote:
Hi!
I was wondering if you can use the base station as a outbound pots
connection for asterisk.
I currently have a tdm410 to do fxs/fxo ports and would like to get rid of
it, I used to use a spa3102, but it only
On Fri, Jan 22, 2010 at 05:06:17PM -0300, Andrew Latham wrote:
I have worked on many snom phones over the years I have never had
a snom phone go bad...
I have had about 10 in the last 12-18 months, I had 1 with a fault hand
set plug - the reseller replaced it. Other wise they have been
Hi
I was wondering if you can use the base station as a outbound pots
connection for asterisk.
I currently have a tdm410 to do fxs/fxo ports and would like to get rid
of it, I used to use a spa3102, but it only had 1 fxo (telephone
connector). I like the idea of the siemans but I would like to
On Tue, Dec 29, 2009 at 11:30:21PM -0500, C F wrote:
Before I start I am a Panasonic certified dealer AND I have installed
over 100 Asterisk systems that are in production.
That said for your application use Panasonic, DONT use Asterisk.
Use the Panasonic KX-TDA50G. Supports up to around 50
Hi
I use one of these http://www.soekris.com/net5501.htm fairly cheap to
buy and to run, I have a tdm410 in there and it has worked flawlessly
I am running debian i386 on the box - it also doubles as my
firewall/router/vpn/adsl box.
I do have one problem with the box (but I have seen on other
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote:
IAXDIAL is free on app store works great on WiFi even true NATs but seem
blocked for GPRS.
ta
HB
[snip]
Well I have a 3gs - will tell you how that goes.
installed (non cracked), but I am on wifi now, easy to configure and
On Tue, Dec 15, 2009 at 09:14:16PM +1100, Alex Samad wrote:
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote:
[snip]
My only concern with it - it's not just a voip client, its many other
things as well. not sure if I want to be a fring user as well as all the
other memberships I have
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
Gavin Spurgeon gspurg...@dageek.co.uk writes:
iSip (£2.39)
http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality from iSip, at least from
the
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote:
Fring, it's free and works perfectly with an Asterisk server..
thanks
On 13 Dec 2009, at 10:15, Alex Samad wrote:
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well
.
--
Sent from mobile device
On Dec 14, 2009, at 6:57 PM, Alex Samad a...@samad.com.au wrote:
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote:
Fring, it's free and works perfectly with an Asterisk server..
thanks
On 13 Dec 2009, at 10:15, Alex Samad wrote
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
Alex
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On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote:
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
[snip]
2. Run from the external shell prompt:
asterisk -rx 'help whatever' | less
Or, you can use the script command to capture the output to a file
On Mon, Nov 16, 2009 at 08:17:27AM -0600, Kevin P. Fleming wrote:
Alex Balashov wrote:
As far as I know, Asterisk has no way to restrict the content of the
domain portion of the Contact URI. However, most commercial SBCs
should have a way to filter this, and it is highly recommended
Interesting on my asterisk box I have installed Virtualbox and I run my
firewall/router in a vm, stripped down linux box with iptables, I have
snapshoted the image to a working image. it only does ip
forwading/vpn/iptable stuff ends up being a low foot print, 256M + 8G /
Alex
On Tue, Oct 13,
Hi
I look after a site which is using asterisk and a vsp for its primary
telco needs, so I am on holiday for a week and of course some jack arse
has decided to reboot the server and something has gone wrong with the
remote access. Now they don't have any internet and i can't fix it
remotely.
Hi
I have a site that has asterisk install with a tdm410 one port is
connected to a pstn that is used as a backup outbound line when/if the
internet/voip is unavailable.
Currently my dial plan for this line is to ignore it, I just basically
do a
s,1,noop
s,n,wait (60)
s,n,hangup
what I would
On Thu, Sep 24, 2009 at 05:32:24PM -0500, Michael Graves wrote:
On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote:
Hi,
Given that the Digium transcoding card has no external connections
(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
Does such a beast exist, or
On Wed, Sep 23, 2009 at 09:39:09AM -0700, mgra...@mstvp.com wrote:
I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.
I
Hi
This is the output from show dialplan dial-sipmnf-sippt-pstn
[ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ]
's' =1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config]
2. Dial(SIP/${ar...@${sipmnf},${ARG2},${OUTBDIAL})
[pbx_config]
On Tue, Sep 22, 2009 at 07:57:56AM -0400, Leif Madsen wrote:
Alex Samad wrote:
4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok)
[pbx_config]
i believe i have captured the relevant logging from the console. my problem
is with Gotoif statement
-- Executing
On Thu, Sep 17, 2009 at 12:02:16PM +0300, Tzafrir Cohen wrote:
On Thu, Sep 17, 2009 at 08:18:13AM +1000, Alex Samad wrote:
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call
On Wed, Sep 16, 2009 at 12:24:22PM -0700, Steve Edwards wrote:
On Wed, 16 Sep 2009, Danny Nicholas wrote:
I'd try this:
- exten = 4000,1,Dial(SIP/4000,20,ikKtT)
- exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten = s-NOANSWER,2,Voicemail(4000)
- exten =
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
Alex
--
Drug therapies are replacing a lot of medicines as we used to know it.
- George W. Bush
10/18/2000
St. Louis, MO
Hi
I am in the process of move a company from pstn to an asterisk setup.
They had 2 pstn lines - only really needed a max of 2 previously.
Now I have installed a tdm410 to handle the cross over from pabx to voip
handset. this has been done, the tdm is now just used to provide a
backup pstn
On Sun, Aug 30, 2009 at 06:49:06PM -0700, Kyle Kienapfel wrote:
It's been my experience that when asterisk does a dns lookup, for externhost
or to do a SIP register, it blocks the whole server. Not sure if 1.6 has
that problem or just 1.4 though as my internet has been stable while im
awake
On Tue, Aug 25, 2009 at 07:30:08PM +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 18.50 skrev John A. Sullivan III:
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
[snip]
mode
in Linux on any old switch and it works reasonably well
Hi
I had a working system, until recently - its asterisk 1.6.1 from debian
- not the lastest as the last doesn't seem to work.
but somebody who rang me said my voice mail announcement was all
stuttery. so i dialed my voicemail box and its really stuttery...
so I have done a reboot and its just
On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote:
On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote:
Hi
I had a working system, until recently - its asterisk 1.6.1 from debian
- not the lastest as the last doesn't seem to work.
but somebody who rang me
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Kevin P. Fleming wrote:
Jeff LaCoursiere wrote:
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
[snip]
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print
On Sat, Aug 15, 2009 at 10:58:07PM +1000, Lee, John (Sydney) wrote:
I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
problems since Dec last year. We are using Digium TE412P to connect to
[snip]
Pid: 0, comm: swapper
EIP: 0060:[,C0417911.] CPU: 1
EIP is at
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote:
Here's how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten = 101,n,Playback(vm-goodbye)
exten =
On Mon, Jul 20, 2009 at 05:58:51PM -0400, Brian McEntire wrote:
Thanks for the reply Alex. I'm not too scared of the soldering iron (I
own one, but my work with it isn't pretty ;-)
But can you confirm, are you just using the small power header on the
board to supply power to the pci card? I
On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote:
Hello -
I've been running Asterisk (quite happily!) for several years now
using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM).
I'm also running another old PC running m0n0wall as a firewall.
Between these two
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
--
There is no instance of a country having benefited
from
On Mon, Jul 13, 2009 at 11:50:00AM -0500, Brent Davidson wrote:
Alex Samad wrote:
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
My problem is that when you call something like internet banking they
want #, but when
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
My problem is that when you call something like internet banking they
want #, but when # is pressed asterisk gets it instead. is there a way
around this ?
I haven't been able to get
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote:
Hello,
i have just installed asterisk 1.6.0.10 on debian 5.0 like:
./configure;make menuselect; make;make install
any reason to not use the deb files ?
There are no erorrs, but folder /usr/lib/asterisk/modules is empty.
What am i
/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder
Best regards,
Loïc Didelot.
On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote:
Hi
I am trying to setup asterisk to do a mass deploy of some snom phones. I
can't find where i configure
Hi
I was wondering if any one has used these cards, I am looking at this as
a replacement for the tdm410, I have some issues with installing the
tdm410 in a small case because of the power plug being at the end of the
board.
I am in australia seems like we have a different setup for out fxs
)
[1083340.340492] Port 2: Installed -- AUTO FXS/DPO
uses complex impedance (220+820Ohm resistors with a 120nF capacitor)
whereas the US uses a straight resistor.
Did yo buy from the us or local ?
Alex
Alex Samad wrote:
Hi
I was wondering if any one has used these cards, I am looking
On Tue, Jun 23, 2009 at 11:32:08AM -0500, Shaun Ruffell wrote:
Alex Samad wrote:
I am having some problem forcing my tdm410 to alaw over ulaw...
You will want to set the alawoverride module parameter to 1. i.e.
'modprobe wctdm24xxp alawoverride=1' or alternatively, edit your
/etc
Hi
I was reading this article on installing asterisk 1.6 + debian
http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian
and I noticed they suggested to recompile to 1000Hz enable kernel, I
currently have a 250Hz stock
On Wed, Jun 24, 2009 at 01:02:08AM +0300, Tzafrir Cohen wrote:
On Wed, Jun 24, 2009 at 07:10:15AM +1000, Alex Samad wrote:
Hi
I was reading this article on installing asterisk 1.6 + debian
http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr
Hi
I am having some problem forcing my tdm410 to alaw over ulaw, I have
1.6.1.0 asterisk (debian i486)
dahdi1:2.2.0 built with the hardware echo canceller firmware
/etc/asterisk/chan_dahdi.conf
alaw=1-4
but I have this in the general section, before any channel definition
dahdi
Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set them to AUS zone
I have got this though
options wctdm24xxp opermode=AUSTRALIA
thanks
--
See, we love -- we love freedom. That's what
On Fri, Jun 19, 2009 at 11:08:49AM +0300, Tzafrir Cohen wrote:
On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote:
Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set
Hi
I am trying to setup asterisk to do a mass deploy of some snom phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set the notify reply.
I was hoping to not have to use dhcp options
alex
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On Thu, Jun 18, 2009 at 07:34:38AM -0400, Alex Balashov wrote:
I thought TFTP (and therefore, DHCP option 66) is the only
autoprovisioning method Asterisk supports?
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address
of *.
The doco seemed to suggest after I press flash I should heard a dial
tone ! which i don't
Alex
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad
Sent: Wednesday, June 17, 2009 9:41 PM
To: asterisk
On Thu, Jun 18, 2009 at 11:57:20PM +0200, Philipp Kempgen wrote:
Alex Samad schrieb:
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address (224.0.1.75) and the
listener is
meant to respond with a notify which has
On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where
Hi
I have 2 digium cards (tdm410) with combination of fxs + multiple fxo
ports.
I have had a quick look at sangoma B series cards. I was wondering if
there is a card out there with
hardware echo canceller
say max 4 ports (mix of fxs/fxo)
g729 encoding onboard
Alex
--
More and more of our
On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote:
After a kernel update (but before rebooting) Is there a way to recompile
Zap/Dahdi against the new kernel?
My objective is to eliminate the additional downtime that occurs while
recompiling/installing zap/dahdi after booting into
On Wed, Jun 17, 2009 at 09:34:55AM -0400, Matt Florell wrote:
On 6/17/09, Gordon Henderson gordon+aster...@drogon.net wrote:
On Wed, 17 Jun 2009, Steve Totaro wrote:
Hi,
[snip]
Gordon
The TC400B is up to 120 channels of G729a now:
Hi
I am trying to get transferring of calls working, I place a call from
ext 101 = 103 and then from 101 I try and transfer the call to 102
(such that it will be 102=103), I have tried flash and *2 and nothing
seems to work.
I have allowed transfers in sip.conf, I am expecting a dial tone when i
On Sun, Jun 14, 2009 at 03:10:03PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware package (sadly it has not made it into the
archive yet
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
Although I think I did see it download the firmware
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware package (sadly it has not made it into the
archive yet). As the firmware
Hi
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error
loading module 'chan_dahdi.so':
On Tue, Jun 16, 2009 at 07:03:06AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30] WARNING
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote:
[snip]
dahdi_genconf generates configuration. It is a tool intended to help you
and not a required step.
It defaults to using mg2[1]. You can tell it to use
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
be activated regadrdless of the software EC you use.
(Not sure exactly how. Anybody?)
It's automatic; nothing needs to be
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
some question I have now is when i do a dahdi show channel 1 i get these
interesting results
Echo Cancellation:
128 taps
currently OFF
I have a hardware echo can and I have asked
On Wed, Jun 17, 2009 at 07:16:53AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
[snip]
Default law: ulaw
I have a alaw:1-4 in the conf file, but it doesn't seem to take
That is not valid syntax for /etc/dahdi
On Wed, Jun 17, 2009 at 01:23:19AM +0300, Tzafrir Cohen wrote:
On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
Hi
I would like the option to set the codec used on a call by call basis.
I have a tdm410 2fxs + 1fxo.
when I make calls to my vsp, they go through as ulaw, I am guessing
because I have allowed if for the vsp (g729, alaw and ulaw).
I would prefer to use g729 from the fxs to the vsp but I would
Hi
I have come across a problem, with my tdp410 and soekris board
(basically pc on a chip amd geode cpu).
I am using the box as a firewall/asterisk box. The problem occurs when I
drop ppp and I get dead loop dectiotn going, I seem to lose interrupts
and get lots of messages in syslog from
On Mon, Jun 15, 2009 at 08:19:33PM -0500, Lyle Giese wrote:
Alex Samad wrote:
Hi
[snip]
as you can see with the interrupts the wctdm24xxp0 is above eth0 (local
lan) and eth3 (my adsl)
eth1 is wireless and not heavily used
So any one had this problems, any other possible
On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote:
To get this to work can i simply
apt-get source dahdi-linux
modify debian/patches/series
to comment out no_firmware_download
then
dpkg-buildpackage
On Sat, Jun 13, 2009 at 01:10:33PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 07:51:54PM +1000, Alex Samad wrote:
On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote:
To get this to work can i simply
On Sat, Jun 13, 2009 at 05:10:34PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 09:46:23PM +1000, Alex Samad wrote:
On Sat, Jun 13, 2009 at 01:10:33PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 07:51:54PM +1000, Alex Samad wrote:
On Sat, Jun 13, 2009 at 11:58:40AM +0300
not tar
balls easier to maintain.
My only issue has been that because of debian rules the firmware for the
hw echo cancellor isn't provided
Alex
\erik
Date: Sat, 13 Jun 2009 09:51:24 +1000
From: Alex Samad a...@samad.com.au
Subject: Re: [asterisk-users] Help building dahdi for debian
On Sun, Jun 14, 2009 at 08:20:18AM +1000, Alex Samad wrote:
[snip]
It merely packages (most of the) the source tarball in the dahdi-source
binary package, which is later built with dahdi-linux.
that makes sense, I had a time constraint, will look at it next weekend,
when I have some
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
On Sun, Jun 14, 2009 at 12:23:41PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 08:20:18AM +1000, Alex Samad wrote:
[snip]
It merely packages (most of the) the source tarball in the dahdi-source
binary
Hi
I am in the process of installing a new box and using dahdi. I have a
tdm410 + hardware echo canceller.
I have just read in the read me for dadhi that VPMADT032 support has
been removed and unlike with the zaptel stuff i could just download and
install the firmware I can't with dahdi
what
On Fri, Jun 12, 2009 at 05:40:16PM +0300, Tzafrir Cohen wrote:
On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote:
Hi
I am in the process of installing a new box and using dahdi. I have a
tdm410 + hardware echo canceller.
I have just read in the read me for dadhi
On Sat, Jun 13, 2009 at 01:40:48AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote:
any chance of getting digium to host a digium debian repo (sort of how
virtulbox doit), that way they could have a fully build package ?
Or resolve the issues
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote:
On Wed, 10 Jun 2009, Alex Samad wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote:
At 02:01 PM 6/10/2009, you wrote:
http://www.cyberguys.com/product-search/?keyword=molex
doesn't look like it, really need a 90 degree plug and I am in OZ not
usa so postage is going to kill me
I'd buy a standard one, pull the pins,
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the motherboard is too small.
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8
On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote:
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card -
On Wed, Jun 10, 2009 at 05:49:22PM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8 just behind the molex connector on the tdm410.
can any one here confirm this, or have any info
Hi
I am trying to setup asterisk at home, I have 1 in bound VSP (I have a
register cmd setup for that in asterisk). At home I have a cordless
phone with 2 line capability - I currently have 2 spa3102's in place to
handle the 2 lines ( I am in the process of buying tdm410 to handle to
handle this
On Wed, Jun 03, 2009 at 08:23:13PM +1000, Rob Hillis wrote:
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Hi
i have just recently installed asterisk 1.4 server with a digium card 410, i
used the zaptel packages in debian.
now I have notice the move to dahdi which seems to be a rename and some
changes as well.
is it a easy change from zaptel to dahdi ? any sort of gotchas to watch
out for ?
Alex
Hi
My setup is
Internet - firewall - asteriskbox
- spa3102a
- spa3102b
the spa's can talk to the firewall directly. The firewall does NAT.
The current asterisk flow for outgoing calls is
phone = spa3102 = asterisk = vsp
and vis versa for inbound
On Thu, May 28, 2009 at 10:49:38AM +0200, Stefan Schmidt wrote:
David Backeberg schrieb:
On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote:
all server are in one rack in our datacenter and are connected to an HP
Procurve 2650 switch, which has been setup around 3 months
On Thu, May 28, 2009 at 02:15:08PM +0200, Stefan Schmidt wrote:
Alex Samad schrieb:
Hi
Hi Alex,
I am new to asterisk so my suggestions might be a bit silly.
Why not setup a iax2 connection bettween the asterisk servers, because
its a lower overhear and more efficient.
We
On Mon, May 25, 2009 at 09:29:54AM +1000, Paul Hales wrote:
Alex Samad wrote:
Hi
I have an account with mynetphone (australia), which gives me two voip
(sip) accounts, which i used to have connected to a spa9000.
this is behind a firewall, so on the spa9000 I would listen on another
Hi
I have an account with mynetphone (australia), which gives me two voip
(sip) accounts, which i used to have connected to a spa9000.
this is behind a firewall, so on the spa9000 I would listen on another
port apart from 5060. so on the firewall 5060 would go to voip1 and
5061 to voip2.
I
On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote:
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
[snip]
I left the busy after dial because this is what the original poster
had. In this case, if the channel does not get hungup
On Tue, May 19, 2009 at 10:38:24AM +1000, Paul Hales wrote:
Not true. I am always wrong.
(wait...is that a paradox?)
only on the 42nd time
PaulH
[snip]
ContactTel Business wrote:
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On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
What you have here should work just fine except:
exten = _1866NXX,1,Dial(ZAP/g1/${EXTEN}) -- note the change from n to 1.
I also don't understand why you have an Answer after your Dial statements.
I would do this:
Hi
I am
On Fri, May 15, 2009 at 12:12:23PM +0300, Tzafrir Cohen wrote:
On Fri, May 15, 2009 at 02:47:30PM +1000, Alex Samad wrote:
Hi
I have a fxs (tdm410 ) connected to a pstn that is primarily used for
faxing, it is meant to be a just in case line.
How do I tell asterisk to ignore
On Thu, May 14, 2009 at 07:46:26AM +0200, Marco Sambo wrote:
FXO channels shuld have FXS signalling, and FXS channels shuld have FXO
signalling, so:
# FXO channels are 1,2,3
fxsks=1,2,3
# FXS channel is 4
fxoks=4
yep turned it around and tested it out, worked, had to fxs tune to get
the
Hi
I have a fxs (tdm410 ) connected to a pstn that is primarily used for
faxing, it is meant to be a just in case line.
How do I tell asterisk to ignore the line completely - ie don;t pick up
when it rings ?
Alex
--
I will have a foreign-handed foreign policy.
- George W. Bush
Hi
I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.
I borrowed a spa9000 to place between the PABX and the PSTN lines. I
have had this going for a while (5 months) and it has been working fine
(some issues with echo and other minor things),
to the FXO and I have my pabx attached to
2 FXS ports, which signal as fxo into asterisk (I could be wrong about
that).
An analogue passthorugh setup _is_ doable, just not overly recommended.
PaulH
Alex Samad wrote:
Hi
I am in the middle of move a small business over from legacy
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