devolution into a flamewar is unfortunate, but such things are
inevitable whenever a 'this' vs 'that' question is posed.
For instance, is the Yugo really any worse than the competing Trabant?
The only correct answer is to fling them both with a Trebuchet
On 2012-02-28 21:22:44 +, Kevin P. Fleming said:
On 02/28/2012 03:08 PM, Troy Telford wrote:
[myprovider]
type=friend
username=
secret=
context=somecontext
host=provider_server
qualify=1000
disallow=all
allow=g729
allow=ulaw
auth=md5,rsa
requirecalltoken=yes
trunk=yes
A serious bug with
niSTIM, etc."
Digium's fax driver doesn't work with IAX2... even in ulaw passthrough mode.
So if I can find that yes, it's so much a problem with my configuration
but a bug in the software, then I'll be s
ery likely.
For the record: 1.8.8.2~dfsg-1 (via Debian packages).
I've tried "trunk=no", and it might have made a difference (I'll have a
better idea after some more testing.)
--
Troy Telford
--
_
-- Bandwi
hich is pretty compelling.
- Aforementioned locations can't get IAX to work well.
- So they hire Steve to get IAX to work properly, and he makes money.
At least, that's my take.
--
Troy Telford
--
_
-- Bandwidth and Colocation Pro
or that the
"single-channel" nature of IAX2 may have something to do with it. IAX2
"talks" on 1 channel, SIP uses "twisted pair" connotation on two channels
(as I understand it).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
e
callers) fixed the call quality issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy
Telford
Sent: Tuesday, February 28, 2012 4:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Same provide
Outgoing connections are fairly typical for a NAT setup - anything can go out.
Any other ideas before I give up on using IAX?
Thanks
--
Troy Telford
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
N
isn't needed/useful anymore?
--
Troy Telford
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.
43
Our goal is to make these apps deployable without needing to know much about
voice infrastructure, and hopefully that will encourage others to add phone
calls, SMS, and video to their services.
Cheers,
Troy
--
_
-- Bandwidth
n the free accounts include SIP. If your registrations to
Cloudvox also time out, it's probably the PIX.
Troy
--
Cloudvox -- http://cloudvox.com/
"Asterisk in the cloud" -- AGI, HTTP/JSON, SIP, REST, live in minutes
--
___
Hey folks,
I am involved with a group that is going to use Twitter, SMS, iPhone,
and Asterisk to field-monitor the upcoming US elections.
The group is pretty large scale and you can find out more here:
http://votereport.pbwiki.com
We need some help with SIP telephony infrastructure. Specif
ngs... of course make sure IAX is selected too with FWD.
-Troy
Shane D wrote:
> I get the following output:
>
> Jan 7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
> Registration of '886036' rejected: 'Registration Refused' from:
> '192.246.69.186&#
sk box. My friend has done this successfully for free. Does
> anyone know of a service for this?
IPKALL.com
There may be others too.
-Troy
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asterisk-users mailing list
To UNSUB
x2.c:5252 register_verify: No registration for
> peer 'home' (from )
Could it be something simple, like missing registeriax=yes for the
extension?
-Troy
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asterisk-users mailing list
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he appropriate "extra settings" for
the modem (IE +ms=v34 or -v90=0 etc) to force a non 56K/v90 connection?
-Troy
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asterisk-users mailing list
To UNSUBSCRIBE or updat
uot; I say it looks like spam to me too. I won't be
trying your product anytime soon, partly because of the way the matter
was handled.
-Troy
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Co
;
> what is the problem with phone ???
>
> add param special???
>
> Note: i am mark number phone and wait ... sesonds and call.
>
>
> thank you.
>
Are you hearing "stutter dial-tone"? If that is the case, maybe turn
off MWI (Messa
It can be "fixed" with the patch from
http://lists.digium.com/pipermail/asterisk-dev/2007-June/028093.html
Cheers, Troy
On 26/06/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:
> This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
> dial from macro it shows &
ten = s-CONGESTION,1,Playtones(congestion)
exten = _s-.,1,Goto(s-CONGESTION,1) ;Treat anything else as no answer
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
--
Brisbane (07) 3018 2840
Fax (07) 3105 5987
Disclaimer - This e
I would have been convinced if you had not top-posted! heh
Rob Schall wrote:
> Tom,
>
> I disagree with your argument for a number of reasons. Each of these
> reasons should be more than enough to convince you I'm correct and you
> should do it my way and only my way.
>
> And for the record, V
Doug wrote:
At 12:51 5/31/2007, Bruce Ferrell wrote:
>Troy Ayers wrote:
>> My asterisk box doesn't recognize DTMF from my analog phone, plugged
>> into my ATA(linksys pap2 version2).
>>
>> I can make/receive calls fine... it's just that, for example, I can
cles via google where some people have this
trouble, but have not seen suggestions on how to fix. I presume this is
an ATA problem, and I expect not much is tweakable on the Asterisk side
regarding this, but still I am looking for suggestions for either
Asterisk or t
default requested but no musiconhold loaded.
Is that the correct way to disable it
Is there a way to get rid of this warning
Thanks
Patrick
Not sure if it's the "right" way or not, but a simple way would be to
put in a silent audio f
this problemn before?
greetz
Try https:// not http://
-Troy
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
busy/una.
I'm nearly done, but I had a thought: before I re-invent the wheel, does
anyone know if this has already been done? My searches only saw basic
answering machines examples.
-Troy
___
--Bandwidth and Colocation provided by Easy
that
though to see if it would fix it). I am fairly sure it is not a
transcoding issue - as the server transcodes for the inbound/outbound
calls.
Has anybody come across this before?
Regards. Troy
Disclaimer - This email and any files transmitted with it
strange
being that none of the other providers we are connected to exhibit
that behavior.
It does it with all the devices we are using (admittedly they are all
from the same company - sipura erm Linksys)
If anybody has any ideas?
Cheers, Troy
again ;)
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
--
Australia :: +61 7 3018 2840
USA ::+1 800 924 1347
Japan ::+81 904 956 8327
On 23/12/06, Dave Schardin <[EMAIL PROTECTED]> wrote:
Unfortunately I don't think there is a way to just do a pass
I use Vitality Communications.
http://www.vitelity.net/
I have no problems with the call quality. I have been with them since
they were SixTel.
They have very responsive customer service, a nice provisioning system
and a fairly easy to understand / manage interface.
Cheers, Troy
On 20/12/06
I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the
tones to be recognized during the background( ) the playback and background
files play, but asterisk doesn't do anything when I start pushing keys -
I've tried it from softphones and pstn line phones
Can anyone tell me wha
thout getting slammed by queue calls.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium
Can you please also send to phpkidhotmail.com.
Rplace the of course.
Thanks.
TRoy
Rajesh kumar wrote:
Please send them to me at [EMAIL PROTECTED]
regards,
Rajesh
- Original Message -
From: Technical Support
To: asterisk-users@lists.digium.com ; 'Roman'
Sent: Frida
e. Once their
code had been completely rewritten, they did an audit and found that
they were no longer using the original code base and made the decision
to move to the GPL. Why they wanted to move to a more restrictive
license is beyond me (and this thread), but they did it.
--
Troy Sett
d by a LOT
of people for a long time now, I'm glad it finally happened.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
Kevin Walsh wrote:
Brian C. Fertig [EMAIL PROTECTED] wrote:
Further info. The domain is registered to Marc Olivier Chouinard. He
has p
s are primarily used for dialup traffic.
Also, on the TNT, I see calling name information coming in from the PSTN
(Lucent 5E), but the TNT will not pass it through the PRI to my * box.
Am I understanding correctly that calling name information also does not
work with SIP?
Thanks,
--
Troy S
t
it does best, telephony.
Please join in the Astmanproxy discussion and let us know what features
you'd like to see added!
Regards,
David Troy
--
David C. Troy
President/CEO
popvox, LLC
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Aste
t.
Thanks, and for you fellow yanks out there, have a great holiday!
Regards,
Dave
--
David C. Troy
President/CEO
popvox, LLC
[EMAIL PROTECTED]
Phone: +1-410-647-5812
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http://lists.digium.
;s Astricon in Madrid and would like to gather some
momentum in the developer and user community for adding new features and
enhancements to the code.
Cheers,
Dave
--
David C. Troy
President/CEO
popvox, LLC
[EMAIL PROTECTED]
Phone: +1-410-647-5812
_
, this may be a good approach for you.
Code is here: http://www.popvox.com/astmanproxy-latest.tgz
Regards,
Dave
--
David C. Troy
President/CEO
popvox, LLC
[EMAIL PROTECTED]
410-647-5812
===
Developing web-based realtime applications
cdr.h that AST_MAX_USER_FIELD
is define as 256. I am using CVS HEAD 3/28/05. Any ideas would
be appreciated TIA.
Troy L. Swaine
Systems Administrator / Engineer
___
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Asterisk-Users@lists.digium.com
http
viour ?
Please anyone who can shed some light ? or offer a workaround ?
T
On 06/01/2005, at 10:22 AM, Troy wrote:
Hi list,
I am having some difficulty implementing a certain dialplan where the
following
happens. If the first Dial() is not answered, I want to play a small
greeting then
ask the caller to e
ser entered ''
== Spawn extension (blah, s, 9) exited non-zero on 'vpb/1-1'
== vpb/1-1: Hangup requested
== vpb/1-1: Ending record mode (1/yes)
== vpb/1-1: Ending play mode on vpb/1-1
== vpb/1-1: Hangup complete
Any ideas what could be wrong ?
Cheers
Troy
_
nt, I have friends who are tops, and friends who
are bottoms. Every one of them seem to get extreme satisfaction from their
relationships with the other.
-- Troy
Settle Pulaski Networks http://www.psknet.com
866.477.5638
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
LAN or
behind a firewall. If you do not take some precautions you may be opening
up a completely unauthenticated proxy to your * box!!
Let me know if you have questions/thoughts/comments about this.
Thanks,
Dave
--
David Troy
popvox, llc
___
Asterisk-
be ready for
inclusion in the CVS head. Just edit apps/Makefile to include
app_waitforsilence.c in app list.
This app can be used for good or for evil; please use only for good.
Dave Troy
___
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[EMAIL PROTECTED]
http://li
The owner of the connection to the PSTN (Telco) must insert the NAME
portion for Call Display. There is no way around that since its their
database
the NAME is located in. Someone correct me if I am wrong .
Yes, I think it's fair to say that the ILEC/CLEC to whom the phone number
is routed is res
n/listinfo/asterisk-users
--
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Expect More!410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925 ww
on the wire when it's
originating?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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To UNSUB
,Company B - Sales
exten => 3023,1,NoOp,Company B - Customer Service
Any of these calls might come in on any of your lines, so how does setting a
different context for different zap channels help?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Mes
ties don't work though (or maybe they work too well?)
SIP/100, penalty 1
SIP/200, penalty 2
Call comes in, SIP/100 picks up
Call comes in, SIP/100 is busy, but SIP/200 NEVER rings...
*sigh*
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Mess
ng all 4 groups until
the call is answered.
What do I need to do to get this behavior? If the answer involves $$, tell
me about it, I'm not afraid to spend some cash to help streamline my
business.
Thanks,
--
Troy Settle
Pulaski Networks
http://www.psknet.com
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andrew Kohlsmith
> Sent: Tuesday, July 20, 2004 9:14 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Echo on a PRI
>
> On Tuesday 20 July 2004 09:04, Tr
Sprint a new one and tell them
to fix their trunking, but...
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steven Critchfield
> Sent: Tuesday, July 2
those curious, my single, biggest beef with mailing lists, is the
inclusion of a list tag in the Subject: line. I know it's Asterisk-Users,
because it says so in the To: line. It also says so in the List-ID: and
Sender: lines.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.
A: Because we read the question in the previous message.
> Q: Why should I post my reply above the quoted text?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
that top-posters are lazy? I say yes, we are. We
don't want to have to scroll through pages of quoted material just to get to
the new stuff.
I say that the bottom posters are lazy. They want a bottom post so that
they enter into a thread 12 messages later, and not have to read the thread
'b
Can you plug a regular telephone into the same port on your 'hardware' pbx
and use it?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Peter Boot
>
2793.1604 MHz.
cpu: 0, clocks: 1995112, slice: 997556
CPU0
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andrew Kohlsmith
> Sent: Tuesday, June 01, 2004 9:16
in dmesg/lspci is 82562EZ. It works fine with
linux 2.4, and FreeBSD 4.8 and 5.x.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bond
> Sent: T
> -Original Message-
> From: Troy Settle
> Sent: Monday, May 31, 2004 6:49 AM
>
> First time around, I just unloaded/reloaded the modules. The
> box locked up tight. On reboot, I get this:
>
> general protection fault:
> CPU:0
> EIP:0
f61d4008 f61d4000
Call Trace:[] [] [] []
[]
[]
Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9
<0>Kernel panic: Aiee, killing interrupt handler!
BTW, this is kernel 2.4.25-gentoo-r2
--
Troy Settle
Pulaski Networks
http://www.pskn
I just disable call waiting on all my sip phones and on all zap interfaces.
No problem.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Julien Levi
>
Running spandsp 0.0.1k, tiff 3.5.7.
I put some audio log files in the same directory.
Thanks,
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steve Unde
Dunno about not being able to generate a tiff, I got rxfax to do that, but
they're badly malformed.
http://roanoke-voip01.psknet.com/fax/
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
>
> -Original Message-
> From: Gregory Junker
>
> On Mon, 2004-04-12 at 11:28 -0400, Troy Settle wrote:
> > At this point, I'm using straight Asterisk, with a a PSTN
> gateway at a data
> > POP passing calls via IAX to my PBX here in the office.
>
nnel bank will not do the
trick).
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Gregory Junker
> Sent: Wednesday, April 07, 2004 7:28
Andreas,
The documentation you seek is on the Asterisk website and the Tiki.
If you jump on IRC, I'm sure there will be plenty of people around that can
answer questions, or for a small fee, perform your initial configuration for
you.
--
Troy Settle
Pulaski Networks
http://www.pskne
en a Cisco 7206 and a 3640. When the total bandwidth
pushes much past 50%, I start getting some crazy distrotion (jitter?),
making it impossible for one or both parties to understand the other.
TIA,
--
Troy Settle
Pulaski Networks
http://www.psknet.com
86
There have been several bites on the bounty, but nobody's hooked yet, so
I don't think a duplication of effort is an issue at this point.
I do wish that someone would get this done though, as I don't exactly
get thrilled over maintaining linux boxes.
*sigh*
--
Troy Settle
P
able to set CIDName to be something
different than the reverse lookup name? My goal is not to spoof the
White House, btw, but it makes a fun example.
Thanks,
Dave
=====
David C. Troy [EMAIL PROTECTED]
Perhaps someone is writing, or has written, an AGI script to fetch
current weather conditions and spit it out to callers?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
.../sounds/en/numbers/6.gsm
.../sounds/en/letters/a.gsm
.../sounds/en/letters/m.gsm
.../sounds/en/t/r/troy.gsm
.../sounds/en/w/r/wrote.gsm
(sorry if I'm a little off from when I actually press the send button
:D)
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477
I can't reproduce this either, but I do have the gsm codec installed (though
WMP won't play a .gsm file).
I play the wav49 files in Winamp with no issue.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business O
them committed to FreeBSD. I'm not up on
Net/Open, so don't ask me about those, but I do want them included in this
bounty.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
end up with a non-digium person
attacking this?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Beh
s before I post the bounty? I will recommend that those with
suggestions on the requirements and those that offer additional bounties
for this will sit in committee to determine when the requirements of the
bounty have been met.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866
MWI functionality comes from chan_sip; can anyone verify this for me?
Anybody have 7960 MWI working with mysql voicemail?
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast
y|*|*|25|dec
include => holiday|17:00-23:59|*|31|dec
; these are the days we're open
include => day|09:00-19:59|mon-fri|*|*
include => day|10:00-14:59|sat|*|*
; if we're not open, we're closed (duh!)
include => night
--
Troy Settle
Pulaski Networks
http://www.ps
usiness for you to investigate. =D
Our CLEC here, KMC Telecom, does the hybrid T1 thing as a matter of
course. I can have a 6x6 system delivered to my customers for less than
$400/month (.09/local connect), or unlimited local outbound for less
than $500/month.
KMC even provides the customer with a
it
leary of doing this and selling it as a supported service.
I've got the voice stuff down I think, my primary interest is in how you
accomplished the data portion.
Thanks,
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pu
abandoned
calls.
My latest patch (1.04) is available here:
http://bugs.digium.com/bug_view_page.php?bug_id=214
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast
Do it with MySQL CDR and you should avoid all of those issues; just add a
field to store the cost to the schema and compute it whenever makes sense.
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ty is warbly and the wav files unnecessarily
large.
Is anyone successfully using soxmix to merge two existing gsm files into a
third gsm file? If so, what version of soxmix, etc?
Cheers,
Dave
=
David C. Troy [EMAIL PROT
-mix.gsm
The mixed version contains only a second or so worth of audio. Also, why
aren't the two files the same size?
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go
st sox compiled from source (sox.sourceforge.net).
Dave
=====
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast?410-544-1329 FAX
570 Ritchie Highway, Severna
?bug_id=214
http://bugs.digium.com/file_download.php?file_id=304&type=bug
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast?410-544-1329 FAX
> -Original Message-
> From: Martin Pycko
> Sent: Thursday, October 02, 2003 4:13 PM
>
> use quit or ctrl-D
>
> Martin
>
>From what I can tell, * doesn't honor EOF, at least I've had no luck with
it.
--
Troy Settle
Pulaski Networks
ing
different than simply displaying the configuration file. By the time
we're done, I think it would be ideal to have abstracted the entire *
configuration and store it in some sort of organized fashion (flat-file,
RDBMS, XML, whatever).
--
Troy Settle
Pulaski Networks
http:/
ipes or
sockets or whatever to maintain a seperation. Doing it like this would keep
Asterisk at arms length from the encumbered code, allowing Digium to keep
the dual license option while also allowing the rest of the world to explore
the possibilities.
--
T
Also,
http://npanxx.darkrealms.net/
http://members.dandy.net/~czg/
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL
ge, which forces you to click over to another page to read another
reference, etc... Of course, you can also use the BBCode [quote], but
that's also a pretty ugly solution as well.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.99
> -Original Message-
> From: Roderick Montgomery
> Sent: Tuesday, September 30, 2003 8:24 AM
>
> According to Troy Settle:
> >
> > Why do they do that? Quite possibly because they, like myself, hate
> > having to scroll through pages and pages
, and stops at the
signature and/or citation.
PS, this is /way/ off topic for this thread and this mailing list, and
is best dropped. In fact, I feel really bad about hitting the send
buttin in about 2 seconds...
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477
I
was thinking about how some companies do release binary only modules for
the Linux kernel, and wrongly assumed that this was provided for in the
GPL.
All I can say, is sorry if I mislead anyone, and thank God I'm not a
software developer and don't have t
ling to grant commit privileges to more people? It
doesn't make much sense to create a new repository for this when Digium
already has the facilities available and a new repository for non-core
items.
BTW, I'm not a coder, I'm just an idea man.
--
Troy Settle
Pulaski Networks
h
if you're writing an application that you would like to
sell, your IP lawyer should be able to easily decipher the GPL and
advise you as to which parts of your code need to be made public.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Cha
asy to use with/without the mouse.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of costas
> Sent: S
ng is many "UNKN"
format SIP channels sitting dormant.
I am running CVS-09/04/03. Has anyone identified a firm cause/fix for
this problem?
Dave
=====
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
o the Zhone (almost everyone uses cordless phones anyways).
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
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