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), w
attached 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
> >
> >
On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
> Alex Samad wrote:
> > On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
> >
> >> I think you have your line types mixed up - FXS is for phones, FXO is
> >> for lines.
> >>
>
On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
>
> Have you tried plugging analog phones into the FXS ports in the Asterisk
> box?
good ideal, but trying to find an old style phone the site has a
commander PABX with digital handsets. I will see if I can track one down
:)
A
>
>
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
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
09/2
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.
> &g
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
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, i
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:
signature.asc
Description: Digital signature
___
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 move
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 fir
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 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 ago,
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 asteri
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 inb
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
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
> 8
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
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 jus
On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote:
> On Wed, Jun 10, 2009 at 7:17 AM, Alex Samad wrote:
> > Hi
> >
> >
> > recently bought a soekris net5501 and a tdm410 to place in there.
> >
> > I am having some issues attaching 12V p
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 con
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 th
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
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 i
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.
> >
> &
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 ?
&
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
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:
> > >
> &
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:
> > > >
do things the debian way - use deb's 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
>
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 i
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 (
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 wctdm24
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
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
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 fir
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': /usr/lib/asteris
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 l
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 ca
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 need
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
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 ta
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 thi
On Wed, Jun 17, 2009 at 09:34:55AM -0400, Matt Florell wrote:
> On 6/17/09, Gordon Henderson wrote:
> > On Wed, 17 Jun 2009, Steve Totaro wrote:
> >
> > > Hi,
[snip]
> >
> > Gordon
>
> The TC400B is up to 120 channels of G729a now:
> http://www.digium.com/en/products/voice/tc400b.php
wow if
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
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
signature.asc
Description: Digital signature
_
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 (22
uggest 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:
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
> >
On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote:
> > On Jun 18, 2009, at 7:25 AM, Alex Samad 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 th
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 in
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 the
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
> > Americ
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 s
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 alte
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 stand
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-configuri
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
volta
er on slot 2 (89V peak)
[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 won
etwork structure.
I have found this to be the simplest way to do it
https://svn.amooma.com/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
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
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 asterisk
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 someth
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 pr
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 tw
. Can you partition a CF card? (ie, one
> partition for the monowall "firmware" and the other for the stripped
> down linux install to run Asterisk?)
>
>
> On Mon, Jul 20, 2009 at 4:44 PM, Alex Samad wrote:
> > On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire
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)
>
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 smp_call_fu
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 hav
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 a
On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote:
> On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad 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 wor
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 reason
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 line
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 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 => s-BUSY,1,Dial
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
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
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
>
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.
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 be
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
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. Ba
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, 200
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 ' | less
>
> Or, you can use the "script" command to capture the output
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 recomme
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
signature.asc
Description: Digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
as
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
stern US, as far as I can tell. So I can't speak to
> > whether voice works over 3G.
> >
> > --
> > Sent from mobile device
> >
> > On Dec 14, 2009, at 6:57 PM, Alex Samad wrote:
> >
> >> On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wr
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 con
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 m
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
> Gavin Spurgeon 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 "other end" so to
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 bo
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
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 g
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 c
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
On Thu, Feb 04, 2010 at 09:52:35PM -0600, Warren Selby wrote:
> On Thu, Feb 4, 2010 at 9:20 PM, Nikhil Nair 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 eth1 is
1 - 100 of 102 matches
Mail list logo
