One of my clients today had a POTS line with a bad punch, and no dialtone.
I used zap destroy channel x remotely to keep it from being used to send
outbound calls, which worked fine. Line repunched, ready again to use,
but how do I undestroy the channel?
In the end I kicked everyone off with
On Wed, 1 Oct 2008, Steve Kennedy wrote:
On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote:
More than 60% of our outbound calls are now to mobiles, so the time has
come to whack in a gsm channel bank.
Does anyone have any preference of bank ? Do you use a PRI or VOIP
On Wed, 1 Oct 2008, Daniel Hazelbaker wrote:
On Oct 1, 2008, at 11:39 AM, Jeff Peeler wrote:
Nope, that's the best you can do without restarting Asterisk. Is
requiring two restarts reproducible? I'd really like to see console
output with verbosity and debug set to 4 on chan_dahdi,
I would say miles. DSL limits for equiv bandwidth is around 3 miles if I
recall correctly.
j
On Fri, 3 Oct 2008, Eric Fort wrote:
without any other hardware than 2 bare ass pci based t1/e1 cards wired back
to back how far can one go between them? additional hardware defeats the
purpose.
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back and forth and
will probably show you that your NAT rules are applying when they
shouldn't. I agree with first turning off your
On Thu, 16 Oct 2008, GNUbie wrote:
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back
On Mon, 27 Oct 2008, Administrator TOOTAI wrote:
bilal ghayyad a ?crit :
[...]
What about Nokia Communicator? Any other Nokia Family that accept to
download fring on it?
Why do you want to use fring on a Nokia as they have a very good SIP
client ?
Speaking of fring, I just got my
On Mon, 27 Oct 2008, Andrew Kohlsmith (lists) wrote:
On October 27, 2008 02:01:43 pm Jeff LaCoursiere wrote:
Speaking of fring, I just got my brand new iphone 3G. Anyone have any
comments on how well fring or any other sip client (siphon?) works on
iphone?
I do not like fring. It's
On Tue, 28 Oct 2008, Alan Lord wrote:
I just wish there was a fanless version - one feature which I like in the
VIA boards I use.
Wow, that's an amazing price for the mobo. Though, like you, WTF do
Intel insist on using a chipset that needs fan cooling and draws about 4
times as much
distance, which in the Virgin
Islands is spotty at best ;)
I think much too big a deal is being made here and over-engineering is at work.
But then my
installations are not into heavy LAN use. I suppose as always it depends on
the situation.
--
Jeff LaCoursiere
JB Telenet, LLC
6501 Redhook
I do understand that this not free, but BillMax (www.billmax.com)
supports all of your requirements plus includes the source code. I think
you can get a demo that supports under 100 accounts for free... at least
you used to be able to.
j
On Wed, 29 Oct 2008, Jerry Jones wrote:
After spending
I've been playing with video phones over the past month or 2.
You've got 3 choices: Bottom-end is Xlite, etc. soft-phones.
Desktop videophones - currently Grandtream GXV3000 and ATL4000's.
Top of the range Polycom video conferencing units.
Starting with the top-of the range
I think everyone is missing the point of the question. He wants to know
if the user's shell is set to rasterisk, can they then use the CLI to get
a command shell.
The answer is yes, they can, and in that case it may not be such a
good idea. As someone else suggested, you can run a shell with
On Tue, 4 Nov 2008, Dima wrote:
The person I'm giving the access to is an admin of that asterisk. It's
up to him to do evil stuff with asterisk itself. as long as he can't get
a shell and do rm -rf / I'm safe.
Hmm, I wonder if you could run asterisk in a jail? Anyone done that on
FreeBSD
On Wed, 5 Nov 2008, Tzafrir Cohen wrote:
On Tue, Nov 04, 2008 at 04:02:40PM -0600, Jeff LaCoursiere wrote:
Hmm, I wonder if you could run asterisk in a jail? Anyone done that on
FreeBSD for example? That would solve your issues I think. It would
certainly be difficult for your admin
Can anyone recommend (offlist) a good IAX or SIP based 800 provider?
Intention is for high volume calling card traffic from the US Virgin
Islands and Puerto Rico.
Thanks,
j
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On Wed, 5 Nov 2008, Mark Michelson wrote:
Thomas Kenyon wrote:
John Todd wrote:
It's a legitimate mail from Digium.
Bit of a pooh survey.
1. Does your business use an Open Source PBX in North America?
\ Err well, no, like 96% of the world, I don't live in North
What about Mexico and Canada? Aren't they considered North America?
j
On Thu, 6 Nov 2008, Thomas Kenyon wrote:
Matt Riddell wrote:
On 6/11/2008 8:37 p.m., Thomas Kenyon wrote:
Jeff LaCoursiere wrote:
I didn't realize only 4% of the world's population lived in North America!
Learn
On Tue, 11 Nov 2008, Peter Evans wrote:
On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote:
hii guys:
i get the message from the asterisk:
Started music on hold, class 'default', on Local/[EMAIL
PROTECTED],1
[2008-11-11 14:32:41] WARNING[1781]:
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman Lesher wrote:
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
I used to use IDEFISK, but
Sorry again for the only marginal relation to asterisk, but the issue does
affect the voice performance I am experiencing, so I am soothing my guilt
with that.
Bet you don't see this every day:
ast% uptime
13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01
ast%
I
it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, November 19, 2008 1:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] puzzle
Sorry again for the only marginal relation to asterisk, but the issue
modprobe.
This of course assumes that the command is in your last 1000 commands.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, November 19, 2008 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, 19 Nov 2008, Tzafrir Cohen wrote:
On Wed, Nov 19, 2008 at 07:57:33PM +, Jeff LaCoursiere wrote:
ast% ps auxw | grep modprobe
root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe
-r ipt_state
modprobe -r is basically rmmod . rmmod and insmod and nowdays
PROTECTED] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, November 19, 2008 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] puzzle
A good idea! The modprobe command is actually in the ps below - it is
part of the /etc/init.d/iptables script
that).
Thanks for the suggestions, though!
j
On Wed, 19 Nov 2008, Steve Totaro wrote:
YUM update? service iptables stop service iptables start?
On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p
, 2008 at 7:19 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
Its not Centos - there is no 'yum'.
service iptables stop is what
produced the hanging process in the first place - I think my big problem
here is that a kernel module is broken, and there is no way to stop it,
and there seems
!
Did I miss anything?
Cheers,
j
On Wed, 19 Nov 2008, Steve Totaro wrote:
I was not implying that you upgrade anything but iptables.
What is the output of ls /etc/init.d/
On Wed, Nov 19, 2008 at 8:02 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
Hi Steve,
[EMAIL PROTECTED] ~]# ls -ltr
No... isn't that a GUI? This is a colo'ed server running a prepaid
calling card app.
Cheers,
j
On Wed, 19 Nov 2008, Steve Totaro wrote:
Are you using NetworkManager?
On Wed, Nov 19, 2008 at 8:29 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
Happy for all suggestions, of course
Hardware solutions are of course simply packaged software solutions.
Personally I would go with something that has this wonderful support base
and quick solutions versus dealing with a vendor. You did mention that
price was a consideration, right?
j
On Thu, 20 Nov 2008, Nitzan Kon wrote:
On Fri, 5 Dec 2008, Tzafrir Cohen wrote:
On Fri, Dec 05, 2008 at 12:46:26PM -0600, Danny Nicholas wrote:
Good programmers can diagram the most obfuscated code. It's part of the
job description.
Anybody with a dialplan that looks like a puppy?
Reminder from a previous thread: a really
I never did solve my puzzle as to how to kill a Linux process that seems
to be deadlocked in kernel space, but thought I would report to the list
that the server did manage to stay up and continue to process several
thousand calls per day:
ast% uptime
11:49:37 up 1000 days, 16:30, 1 user,
On Mon, 8 Dec 2008, Gordon Henderson wrote:
Pah! I take your 1000 days and raise you:
% uptime
18:18:11 up 1146 days, 5:20, 1 user, load average: 0.08, 0.03, 0.01
Other than as a test-bed some months back, this isn't an asterisk server
though.
No fair! Must be a server in active use
On Mon, 8 Dec 2008, Danny Nicholas wrote:
The 100,000,000 calls without a crash are more impressive to me than the
1000 days of uptime. Mine crashes on crazy things like dynamic conferences,
etc. :(
To be upfront the system is only running a prepaid AGI app and routing
calls for
On Mon, 8 Dec 2008, Kristian Kielhofner wrote:
That much uptime at The Planet in Dallas? I guess you're lucky:
http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm
On Mon, 8 Dec 2008, RE Kushner List Account wrote:
That's what happens when illegal aliens, er, Undocumented Americans, do
all your contracting work.
But they taste like chicken!
:)
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I built one in C using AGI. Would you consider licensing the source?
j
On Thu, 11 Dec 2008, Michael wrote:
I want to build my own calling card system on Asterisk.
I looked at this page -
http://www.voipinfo.org/wiki/view/CallingCard+Applications
and it has listed some applications that
On Sat, 13 Dec 2008, Michael wrote:
I'm not saying it can't be done - just be aware that the undertaking
you're proposing is very complicated, and the information would come
from innumerable data sources (a great deal of them commercial and
expensive) and a bewilderingly overlapping array of
On Sat, 13 Dec 2008, Michael wrote:
In general you don't need to worry about that, as when you go to buy your
routes, the splits are given to you. For example, though you have split
up New Zealand nicely I don't need that information, as the termination
provider I buy New Zealand from
On Sat, 13 Dec 2008, Michael wrote:
Hmm, I looked over your summary again against the route prefixes I just
gave and they seem to match. They aren't as detailed, but that isn't
important, as long as I can tell a cellular from a landline, which those
prefixes do accomplish. I don't really
On Sat, 13 Dec 2008, Michael wrote:
On Sat, 13 Dec 2008 16:45:11 Alex Balashov wrote:
Michael wrote:
Let's forget about USA/Canada for now as from my/most people's point of
view the routes are all so cheap (and blended) that it does not matter. I
think it is more important to focus on
On Sun, 14 Dec 2008, Tzafrir Cohen wrote:
Right. So for those of us who want to do simple things and avoid
complicated stuff such as telephony in shoddy continent of North
America, could you please provide data for your country?
So far we have AU, IL and NZ.
Not that I am trying to put
TTL is part of the UDP header (Time To Live). It isn't really about the
voice at all.
Length 345 is the number of bytes in the packet.
j
On Mon, 15 Dec 2008, michel freiha wrote:
*Dear All,
I run the below tcp dump on my asterisk server
tcpdump -i eth0 -n -s0 -v udp port 5060
I got
on asterisk server?
Regards
On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote:
TTL is part of the UDP header (Time To Live). It isn't really about the
voice at all.
Length 345 is the number of bytes in the packet.
j
On Mon, 15 Dec 2008, michel freiha wrote:
*Dear
or variable exist anywhere?
Regards
On Mon, Dec 15, 2008 at 10:55 PM, Jeff LaCoursiere j...@jeff.net wrote:
No. TTL in the header is about hop traversal. Each IP router that
forwards the packet will reduce this number in the live packet until it
reaches zero, when it will be dropped. I believe
Anyone know of any IT work in the Chicago area? I just moved up here and
am finding the economy has really stifled things.
Will do IT mgmt/Unix/Networking/VoIP/C for food...
Cheers,
j
http://www.jeff.net/resume.pdf
___
-- Bandwidth and Colocation
Use the AGI script to collect the digits instead of doing it in your
dialplan.
j
On Wed, 17 Dec 2008, Michael wrote:
I want to take series of user entered (via phone keypad) options/numeric entry
fields and use these with an AGI script. I have looked through voip-info and
I can't find any
Surely only list members should be allowed to post unmoderated?
On Thu, 18 Dec 2008, ad...@viagra.com wrote:
Dear asterisk-us...@lists.digium.com!
Lovers package at discount price!
Discount price store: ID 406858
http://tba.dojmoquj.cn?faz
Pfizer is a licensee of the TRUSTe Privacy
On Thu, 18 Dec 2008, John Todd wrote:
Dear asterisk-us...@lists.digium.com!
Lovers package at discount price!
Discount price store: ID 406858
http://tba.dojmoquj.cn?faz
Pfizer is a licensee of the TRUSTe Privacy Program.
? 2001-2008 Pfizer Inc. All rights reserved.
That is correct -
Well a good hanging might bemore satisfying, anyway. Details... ;)
On Thu, 18 Dec 2008, David Gibbons wrote:
Last I checked, Lynch mobs don't shoot people.
snip
I wonder if there would be interest in organizing a bounty for a lynching
mob, that would track down these !...@#$# silly
On Tue, 23 Dec 2008, Brent Davidson wrote:
Dave Fullerton wrote:
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
I'm
What does Audiocodes release under GPL?
j
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
AudioCodes blatantly violates the terms of the GPL by not distributing
the source code even after requesting it. Please don't use their
hardware.
On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not and
later they admitted it did, but refused to supply the source code
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not and
later they admitted it did, but refused to supply the source code
Is it ready for prime time? I am about to install a new server that will
be processing about 3M minutes per month and running a custom AGI program
for prepaid calling cards. Need to choose between 1.4x and 1.6...
Cheers,
j
___
-- Bandwidth and
On Wed, 7 Jan 2009, Matthias Apitz wrote:
El d?a Wednesday, January 07, 2009 a las 10:55:55AM -0500, Singer XJ Wang
escribi?:
As much as I'm an open source guy, but the OpenMoko phones are worthless
IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it
too much to ask
Does your fring work over the 3G network also or just the wifi?
Cheers,
j
On Wed, 7 Jan 2009, Eric Moniz wrote:
TianLun,
I should have know you would have wanted a Blackberry SIP client to connect
to an Asterisk box. Sorry my bad!
I knew there was a reason why I didn't choose Truphone as
On Thu, 8 Jan 2009, Brent Vrieze wrote:
Thczv F. Thczv wrote:
[snip]
and not
have to use a timeout when dialing long distance.
[snip]
I think you are over thinking this. We set our Asterisk server up with
multiple outgoing dial rules to handle local and long distance. Keep in
mind
I'm pretty sure he was asking about a minimum of two boxes. 4 quad T1
cards would only be 16 x 24 = 384 lines. I agree to a point though - if
you have a service that is utilizing close to 800 lines and half of your
service suddenly bites the dust you would probably be in a world of hurt.
I
[also posted on Trixbox trunk forum]
I am also working with Sangoma directly to debug this, but so far no real
luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE
3.2.6 (3.2.7 is out, but nothing has changed that would affect this
problem). The system gets about 200 calls
On Fri, 9 Jan 2009, Andres wrote:
[snip]
I have the full logging enabled, and here is an excerpt of a call that
was terminated. You can see the conversation lasted about forty seconds
before it was hungup.
What you need to do is figure out who is ordering the call to be
hangup. For
On Fri, 9 Jan 2009, Steve Totaro wrote:
It looks normal to me. I think two dropped calls a day is reasonable
and I would start looking for commonalities.
I tried that logic - they don't buy it :) The sad part is I replace a
Nortel system that did NOT have the issue (according to their
On Fri, 9 Jan 2009, James Noble wrote:
I had the same problem with a sangoma card and a clean install of asterisk
as well as a trixbox set up. I finally started using a vegastream to handle
the T1 connections and was able to get rid of the problem.
James
$5K for a sinlge T1? Thats an
On Fri, 9 Jan 2009, Steve Totaro wrote:
$5k for a single T1 is/was pretty much the norm. Go price non-used T1
cards for big proprietary phone systems.
Thats a copout. Big proprietary phone systems are expensive by default -
certainly not to be considered the norm.
I say it is an
On Sat, 10 Jan 2009, Tzafrir Cohen wrote:
Its not a PRI. Its an RBS T1 with EM Wink. I will try enabling the SIP
debug, though, that is a good idea. Is there any kind of extra debugging
for RBS T1?
No idea, but the driver is much more aware of the specifics. So maybe
their driver has
On Sat, 10 Jan 2009, Kevin P. Fleming wrote:
John Todd wrote:
Desired procedure: A public key signature method would be publicly
available via an SSL web page or various keyservers. Individuals
could sign messages with the public key. Signed messages sent to
security@ would then be
On Sat, 10 Jan 2009, Tzafrir Cohen wrote:
No idea, but the driver is much more aware of the specifics. So maybe
their driver has extra debugging information for that case.
For starters, have you enabled full debugging in Asterisk? Make sure you
log 'debug' and set debug to at least 5 .
On Mon, 12 Jan 2009, David fire wrote:
hi again mybe this info is usefull to solve this problem
*box1---*box2*box3
box2 originate 1 call to box1 and to box 3
using sip/box1/1
extension 1
context default
exten = 1,1,dial(sip/box3/1)
box1 and box3 will exec musiconhold when they
On Thu, 15 Jan 2009, Geoff Lane wrote:
On Thursday, January 15, 2009, Drew Gibson wrote:
[snip]
However, SLA is functionally almost the same as call parking. In that
system, I transfer the call to extension 700 and the parking system
tells me the number (usually 701) I need to dial to
On Thu, 15 Jan 2009, Geoff Lane wrote:
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's bad enough playing find the phone when a
cordless handset
On Thu, 15 Jan 2009, Tilghman Lesher wrote:
On Thursday 15 January 2009 13:02:32 Geoff Lane wrote:
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's
I would be more worried about the ATA gateway failing than the switch, as
you have found yourself. How about two gateways and two phones on
everyone's desk :)
j
On Fri, 16 Jan 2009, Adam Moffett wrote:
I don't know of any ATA like that except the grandstream.
The service provider grade
Agg, I felt bad about being pedantic. How about splitting the load and
reducing the single point of failure? Instead of one big ATA how about a
number of smaller ones (two port) split between your switches?
j
On Fri, 16 Jan 2009, Jeff LaCoursiere wrote:
I would be more worried about
On Mon, 19 Jan 2009, Joseph wrote:
On Mon, 19 Jan 2009, bilal ghayyad wrote:
Hi All;
Anyone knows an IAX IP Phone works fine and tested?
Does polycom support IAX IP Phone?
Regards
Bilal
How about IAX2 adapter from digium?
I've been uing it and it works very well.
Wow, that has
Anyone have an example XML file for the PAP2T?
Cheers,
j
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Wed, 21 Jan 2009, Stefan Schmidt wrote:
Tom Moore schrieb:
I'm not sure if this trick will work with this device, but I was able to
pull down a spa8000's config by connecting to:
http://ipaddress/admin/spacfg.xml
Tom
Hello,
The spacfg.xml link doesnt work on a Pap2T but you could
On Wed, 21 Jan 2009, Stefan Schmidt wrote:
I know it's pretty much a given, but don't forget to edit/remove the
provisioning info. I'd hate to see someone's open device locked.
Tim
___
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On Wed, 21 Jan 2009, Andres wrote:
Why don't you just edit the Trixbox endpoint manager files. They
produce basic XML files for Linksys and Polycom phones. It is trivial
to add support for any Linksys ATA as well.
File is: /var/www/html/maint/modules/11_endpointcfg/endpoint_linksys.php
know if you need to know how to use it
Regards
On Wed, Jan 21, 2009 at 1:04 AM, Jeff LaCoursiere j...@jeff.net wrote:
Anyone have an example XML file for the PAP2T?
Cheers,
j
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Usually a SIP DID provider will let you pay for extra ports for each DID
number you have incoming (in fact a lot of them offer unlimited). Are
you saying your provider is only allowing one call to your DID at a time
with no option to increase it? There should be no reason to try and make
a
Try www.ipcomms.net . I have been using their inbound 800 service for
some time, and I know they do US/Canada DIDs. You can spec the number of
channels you want.
j
On Wed, 21 Jan 2009, Alfred Monticello wrote:
Ideally, I'd like one number that can handle 5 calls or more. The provider I
This is *really* not the place for this...
On Thu, 22 Jan 2009, Andrew Joakimsen wrote:
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your service is still up and working,
Because Suzanne Bowen has better judgment than you.
You did charge back on the payment
to inform this list of bad service.
j
On Thu, 22 Jan 2009, Alex Balashov wrote:
Au contraire.
Jeff LaCoursiere wrote:
This is *really* not the place for this...
On Thu, 22 Jan 2009, Andrew Joakimsen wrote:
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your
Can you configure the LAN port on the back of a 2102 to be bridged
rather than routed to the WAN port?
Cheers,
j
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To UNSUBSCRIBE or update options
On Thu, 22 Jan 2009, Brian J. Murrell wrote:
On Wed, 21 Jan 2009 19:02:01 -0500, Steve Totaro wrote:
Why not just get a softphone and use a USB soundcard or even the onboard
sound card as your ATA? Like a MagicJack and SJphone or Xlite or
whatever it is that works with it.
Please forgive
On Fri, 23 Jan 2009, Kristian Kielhofner wrote:
On Fri, Jan 23, 2009 at 9:18 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Fine using ATT's (at least I think they belong to ATT) DNS servers
(Also do NTP). I will make this thread useful to someone.
Listed below because they are
To be fair they did specify underground ;)
j
On Sat, 24 Jan 2009, Don Kelly wrote:
For fiber installations, be sure that your loops are not placed where
flashes will distract drivers or people performing potentially dangerous
activities.
--Don
Don Kelly
PCF Corp
People Come First
Wilton Helm wrote:
[snip]
My conclusion after installing a worthless * demo (that actually does
allow two SIPs to talk to each other) is that Asterisk is not of any
value to anyone other than a person who makes a full time career out
of running Asterisk systems. I've installed and
I'm actually having trouble understanding what people would order BRI for
over POTS lines. The only thing I ever used ISDN for was net access, and
it was trumped by DSL a decade ago. Do you get some extra service with
your 2B service over ordering two POTS lines?
j
On Wed, 28 Jan 2009,
Here is the bomb:
http://www.clarityproducts.com/products/listing/item3200.asp
95Db :)
Plug this into a cheap ATA as was suggested earlier. Solution should be
about $100.
j
On Wed, 28 Jan 2009, Brent Vrieze wrote:
If you know anyone with electronic experience you could take the speaker
On Wed, 28 Jan 2009, Mike wrote:
My previous question brings me to this:
I know there are plenty of SIP ATA, but is there one that is particularly
recommended that answers (as many of) the following needs:
1) As cheap as possible
2) Allows for auto-provisioning/configuration using a
On Wed, 28 Jan 2009, Karl Fife wrote:
One problem with BRI adoption has no doubt been the need for external power
to the NT1 or TA.
Obviously analog loops are powered by the CO, so much of the benefit of
ISDN-BRI as the first voice circuit is eroded away for a large percentage of
the
On Thu, 29 Jan 2009, Daniel Johnson wrote:
pdha...@optusnet.com.au wrote:
Funnily enough, most people install phones with BLF lamps, on install
something like hudlite/FOP/etc so you know if the person is on the phone
before you call them..
PaulH
Hi Paul,
Yes I have seen these tools.
Nope - 286 didn't have protected memory access mode, which is key to *nix
kernels.
j
On Thu, 29 Jan 2009, Danny Nicholas wrote:
If you're not GUI-ing, you could theoretically run * on a 286 since Linux
doesn't have the overhead of Windows.
_
From:
This thread made me nostalgic - see this:
http://en.wikipedia.org/wiki/MINIX
I took a course based on MINIX (as did Linus Torvald) back in 1989 and
recall building symbolic links into its kernel as part of a class project.
On a 386SX I built in my dorm room.
j
On Fri, 30 Jan 2009, Jeff
Funny how a topic will come up that you have never dealt with before, and
suddenly it comes up from multiple directions at the same time. I was
recently involved in a meeting where TAPI (which I understand only
vaguely) was proposed as way to link a custom application to Asterisk for
/thirdlane-dialer
This seems to be just the ticket...
Cheers,
j
On Fri, 30 Jan 2009, Jeff LaCoursiere wrote:
Funny how a topic will come up that you have never dealt with before, and
suddenly it comes up from multiple directions at the same time. I was
recently involved in a meeting where TAPI
This would work if you only care that you get very rough phonetic
spellings as Don implied. If you think about it humans cannot do any
better. I know personally - I have to spell my name all the time.
Perhaps your app could ask them to spell their name, which actually has a
shot at
I am confused as to what you are trying to accomplish. Can you be more
specific? It seems that you are making this too complicated. You say
that the remote end is providing you two SIP trunks that will come from
the same IP address. To distinguish them simply have them authenticate
with
, it will give me all the schema related to this account. Sometimes
I need to use another schema for some calls, I am not able until send for the
provider from another IP.
Did u get what I need?
Regards
Bilal
--- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote:
From: Jeff
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