On Thu, Aug 25, 2011 at 4:36 AM, Skyler skchopper...@gmail.com wrote:
Steve,
On Thu, 2011-08-25 at 00:39 -0400, Steve Totaro wrote:
...
For fax, I use Hylafax and for text, I use Kannel. These are WAY more
powerful than Asterisk apps. With Kannel, I used the Bluetooth GSM
modem
Normally a call into SIP has one of 3 paths:
1) Unauthenticated, so use the default
2) Identifiable username
3) Identifiable IP address
In the above example, we have a BLOCK of IP addresses instead of a
single address. Can this be made to work?
Thanks for any pointers.
Regards,
Steve
, but for me,
personally, DISA would be a an added cost and complication.
The only purpose I can think of for myself could be accomplished by spoofing
caller id.
Thanks,
Steve T
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NBX100 by Polycom. True plug and play. Can be IP but uses MAC on the LAN
except for the PBX.
It may be discontinued but there is plenty on Ebay. Cheap, scales well,
tons of options.
On Wed, Aug 24, 2011 at 10:34 AM, C F shma...@gmail.com wrote:
The 824 is NOT discontinued.
On 8/23/11,
:
Try put the php in the AGI directory and call it via AGI() in Asterisk.
If the script does not conform to the AGI protocol, do not call it with
agi().
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-
Steve Edwards sedwa...@sedwards.com
Gao wrote:
Try put the php in the AGI directory and call it via AGI() in Asterisk.
On Wed, Aug 24, 2011, Steve Edwards wrote:
If the script does not conform to the AGI protocol, do not call it with
agi().
On Wed, 24 Aug 2011, Tim King wrote:
What does this mean? The suggestion from Jian
=${EPOCH})
Aside from allowing you to specify the parameters in any order (I'm big on
'alphabetization'), it is much easier to maintain when you pass several
parameters.
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-
Steve Edwards sedwa
On 11-08-24 03:48 PM, Steve Edwards wrote:
You can pass multiple parameters on the command line like:
exten = *,n,system(echo ${EPOCH} ${CHANNEL} /tmp/foo)
(When I wrote this example, I used the EXTEN channel variable instead of
EPOCH. Extra points for the first reader to explain
for a really bad set up of
questions.
Thread subject, Looking for ideas for nice **Asterisk** home phone system
Actual subject?!?!
On Wed, Aug 24, 2011 at 3:15 AM, Per Jessen p...@computer.org wrote:
Steve Totaro wrote:
On Wed, Aug 24, 2011 at 2:42 AM, Per Jessen p...@computer.org wrote
ten SMS per second
using ten phones. Kannel is very well developed. Chan_mobile is
incredible.
The same is true with HylaFAX.
Thanks,
Steve T
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New
phone call, followed by a an inbound
call from the server that you initiated with call back.
Inbound to a cell is generally less expensive that oubound on a cell,
sometimes completely free.
Corp identity shows just fine.
Thanks,
Steve Totaro
On Wed, Aug 24, 2011 at 3:15 AM, Per Jessen p
capabilities in conjunction with Asterisk ?
Invest a couple of days reading http://nerdvittles.com/
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
. This one must be the new record for asking for something for
nothing.
Thanks,
Steve Totaro
Forwarding to biz list
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. This one must be the new record for asking for something for
nothing.
Thanks,
Steve Totaro
Forwarding to biz list
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EOL, 1.8 etc etc, it needs revisiting.
As far as I know, 1.8/1.10 has the same issues.
Regards,
Steve
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will only
allow you to set it to a number allocated to you, some will not let you
set it at all.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
to be pretty good stuff.
[un]Fortunately, I've never had the need to try either.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax
On Sun, 14 Aug 2011, Tahar .H wrote:
can some one please explain to me what this one stands for :
Exten = 1234,1,read(numtodial,enternum,10,skip,1,10)
[core] show application read
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in setting up the AGI environment.
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Newline Fax: +1-760-731-3000
On Tue, Aug 9, 2011 at 5:21 PM, C F shma...@gmail.com wrote:
On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan sas...@gmail.com wrote:
Hi,
I would like to make sure I got it right:
1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
fax from the PSTN and have anther FAX
inband passthrough
and you are using 711u on your VOIP legs you should be fine... YMMV of
course.
-Ryan
On Tue, Aug 9, 2011 at 6:00 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Tue, Aug 9, 2011 at 5:21 PM, C F shma...@gmail.com wrote:
On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan
On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard fax...@howardsilvan.com wrote:
Ryan McGuire wrote:
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38. The cheapest and
easiest approach that I have found is to buy yourself an FXS
that isn't borken.
-Ryan
On Tue, Aug 9, 2011 at 6:00 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Tue, Aug 9, 2011 at 5:21 PM, C F shma...@gmail.com wrote:
On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan sas...@gmail.com wrote:
Hi,
I would like to make sure I got it right:
1
On Tue, Aug 9, 2011 at 8:13 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
On Tue, Aug 9, 2011 at 6:15 PM, Ryan McGuire rdmcguir...@gmail.com wrote:
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38. The cheapest
On Tue, Aug 9, 2011 at 8:31 PM, Lee Howard fax...@howardsilvan.com wrote:
Steve Totaro wrote:
On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard fax...@howardsilvan.com
wrote:
Ryan McGuire wrote:
Unless your network is under load and you are seeing dropped packets
and high jitter, I would
as a fax endpoint (hylafax, app_fax take your pick)
should work in most cases if the handoff to asterisk is TDM and could
work if it's T.38
On Tue, Aug 9, 2011 at 9:28 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
On Tue, Aug 9, 2011 at 8:31 PM, Lee Howard fax...@howardsilvan.com wrote
Off topic but cool tech. One of the defunct Asterisk Appliance's was
based on the Altix 350.
http://www.sgi.com/products/servers/altix/numalink.html
On Wed, Aug 10, 2011 at 12:12 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
I do love this stuff. Being self taught, Asterisk actually
and I'm too lazy to try to
figure it out.' Guess how many files have '777' on any freshly installed
OS?
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
and
nobody knows what patent issues might come crawling into view. iSAC has
been around for a while. The source wasn't open until recently, but
licenced users have had it for a long time. There has been much more
opportunity for patent issues to show up.
Steve
} -c 1 -s -w -r 8000 /tmp/$$.wav
What does your sox command line look like?
Can you post a link to 'before' and 'after' files?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
way past it's 'sell date.'
The OP has his religious beliefs, many have tried to show him the light,
but he just keeps digging in deeper.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760
?
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
cannot
live without, DBAN the disk and start over.
Before you re-install the OS, read up on what you should have done the
first time.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468
On Thu, 21 Jul 2011, Robert Huddleston wrote:
When I get hacked I typically run a rootkit checker
http://www.chkrootkit.org/
How often do you get hacked?
How are 'they' breaking in?
--
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Steve Edwards
in sh running
sh running foo.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
.
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On Wed, 20 Jul 2011, Danny Nicholas wrote:
I?m putting Asterisk in to replace an existing IVR and that PBX system
uses * to terminate number input instead of #.
On Wed, 20 Jul 2011, Steve Edwards wrote:
How about an AGI executing some mix of get data, get option, stream
file, or wait
Lee
I've not had a problem here with 1.6.2.19.
What are you reloading that causes the issue, and can you post the
usual gdb backtrace somewhere? Perhaps on the bug tracker.
Regards,
Steve
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need is T.38 - Asterisk 1.6 supports this
to a limited degree, and your ITSP will need to support it.
The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.
Regards,
Steve
On 18 July 2011 13:00, Lee Archer lee.arc...@thebigword.com wrote:
Hi Steve, I think it's related to my ODBC connection. I started with a
random hang where it looked ODBC related which led me to try a few things.
Reloading the config a few times is causing core dumps which 1.6.2.18.2 just
, then it should still be fixed.
At least I believe that's the rules.
Regards,
Steve
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in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
you find a solution?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards
asterisk@sedwards.com wrote:
I have a TDM400p with 3 fxs and 1 fxo daughter cards.
It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p
is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive.
I'm getting a bunch
On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards
asterisk@sedwards.com wrote:
I have a TDM400p with 3 fxs and 1 fxo daughter cards.
It's in a mini-itx case with a 'right-angle' PCI riser card so the
TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard
drive.
I'm getting
On Wed, 13 Jul 2011, Eric Wieling wrote:
cat /proc/interrupts will tell you if the card is sharing IRQs with
anything else.
The card is on it's own on interrupt 66.
--
Thanks in advance,
-
Steve Edwards sedwa
for (in alphabetical order) Gordon Henderson, Kevin P. Fleming,
Steve Underwood, Tilghman Lesher, Tony Mountifield and many others. We sit
at the feet of giants.
But you're out of line using other people's code as examples of bad
programming techniques. Unless my code is in a directory on your desktop
On Mon, Jul 11, 2011 at 06:45:08PM -0700, Steve Edwards wrote:
while read line; do
epoch=`echo $line | cut -d '|' -f 1`
if [ $epoch -ge $start_epoch -a $epoch -le $end_epoch ]; then
echo $line
fi
done /var/log/asterisk/queue_log
[snipping snippy comments about improving
about using an actual TDM card since that can be handled by
Redfone or a USB device.
Thanks,
Steve Totaro
On Sun, Jul 10, 2011 at 4:11 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Fri, Jul 08, 2011 at 07:28:39PM -0400, Doug Lytle wrote:
Can you say a Virtualized Asterisk with a PRI card
On Mon, Jul 11, 2011 at 8:22 AM, Doug Lytle supp...@drdos.info wrote:
Doug Lytle wrote:
I've been searching the Jira issue tracker and found a ticket:
What I ended up doing was to copy the app_meetme.c out of the 1.4.30 source
and compiled it into my current Asterisk setup. I now have PIN
El 08/07/11 12:50, Steve Edwards escribió:
*) You can execute hundreds of AGIs written in C in the time it takes
to load the Perl interpreter and parse your script.
On Fri, 8 Jul 2011, Miguel Molina wrote:
Just curious... have you timed this to demonstrate?
What? You want to confuse
to
play with.)
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On 12/07/11 9:29 AM, Steve Edwards wrote:
Many times, I've made the statement that you can execute hundreds of
AGIs written in C in the time it takes to load an interpreter and parse
a script written in PHP or Perl.
Well, now that I know better, let's not perpetuate an ancient claim
learning to appreciate PHP and how it lets me represent some
programming problems clearly, quickly and sufficiently efficient.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards
asterisk@sedwards.com wrote:
Many times, I've made the statement that you can execute hundreds of AGIs
written in C in the time it takes to load an interpreter and parse a script
written in PHP or Perl.
I can see I'm going to spend the rest
On 11/07/11 23:42, Steve Edwards wrote:
'Standalone' AGIs still have advantages in lower complexity and less
impact on failure. If a bug takes out your fastagi daemon it can affect
all calls.
On Tue, 12 Jul 2011, Vincent Sweeney wrote:
I'm pretty sure if you have a bug in your AGI code
On Saturday, 9 July 2011, Gordon Henderson gordon+aster...@drogon.net wrote:
On Sat, 9 Jul 2011, Steve Davies wrote:
On 9 July 2011 12:34, randulo rand...@randulo.com wrote:
Go ahead and lambast me for this post, it isn't specific to Asterisk, but:
G+ has only been open at all for a week
suggest a good way of finding/following appropriate
VoIP/Asterisk people once on Google+? How do you then group them? Just
in a Circle, or some other mechanism?
Thanks,
Steve
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, deserve neither Liberty nor Safety.
I guess you could do it the old fashioned way until you open a ticket on
Bugtracker?
Sorry, not much help, but I believe that Bugtracker is the proper way to
report a bug.
Thanks,
Steve Totaro
that are are
not other roadblocks yet to be seen.
Thanks,
Steve Totaro
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in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On Fri, Jul 8, 2011 at 7:28 PM, Doug Lytle supp...@drdos.info wrote:
Can you say a Virtualized Asterisk with a PRI card!
http://www.phoronix.com/scan.**php?page=news_itempx=OTY0OQhttp://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ
Doug
--
Ben Franklin quote:
Those who would give up
assigned to different VMs?
Thanks,
Steve Totaro
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On Thu, 7 Jul 2011, Tim Nelson wrote:
On occasion, I have calls coming into an Asterisk 1.2.x system where
the ${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the dashes from the number, leaving me with
'1234567890'?
On Thu, 7 Jul 2011, Steve Edwards wrote:
I would do
have done and the hoops they have to crawl
through to achieve an inferior solution and decided I should stick to what
I know.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
The argument to chanspy is a pattern and not an exact match.
--
Jim Dickenson
mailto:dickenson at cfmc.com
CfMC
http://www.cfmc.com/
On Jul 2, 2011, at 3:48 PM, steve casto wrote:
asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use
flash operator panel 2.0
(from
,
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Newline Fax: +1-760-731-3000
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New
1 19:06:48] VERBOSE[15242] logger.c: == Done Spying on channel
Zap/41-1
[Jul 1 19:06:48] VERBOSE[15242] logger.c: == Spying on channel Zap/4-1
[Jul 1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Zap/31-1 to Zap/4-1
thanks list
Steve
.
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Newline Fax: +1-760-731-3000
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote:
How to create the conference feature in Asterisk?
RTM, keeping your eyes open for references to 'meetme.'
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On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote:
I am referring to 3-way conference
With a little reading, you would discover that meetme can handle lots of
participants.
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by all in any way (BSD
licence style). I believe that prior to that there was a requirement
to register every commercial use of the codec with the licence holder,
or some-such thing?
Regards,
Steve
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On 22 June 2011 17:14, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
On 06/22/2011 03:32 PM, Steve Davies wrote:
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc
,
Steve
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, extension patterns and/or an AGI can all be useful tools to solve
your problem.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
think about it.
Steve
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On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Two requests, not from me but the community.
1. Don't top post
*cough*
2. When you find your solution, reply to this thread
in Google.
Thanks,
Steve T
On Mon, Jun 20, 2011 at 1:44 AM, virendra bhati virbh...@gmail.com wrote:
Hi Steve,
Thanks for share your knowledge. I will revert back to you after testing
with asterisk.
On Sun, Jun 19, 2011 at 6:46 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Sun
.
I just found this article that should answer most if not all of your
questions. It work just fine for me at many locations.
http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN834
Thanks,
Steve Totaro
Thanks,
Steve Totaro
On Sun, Jun 19, 2011 at 8:49 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Sun, Jun 19, 2011 at 5:13 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I have installed Kannel server into my Linux server. I have asterisk
installed into the same server. Now I want to connect
get more than 1% failures when sending FAXes
to and from your own equipment you should be looking into the cause.
Steve
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in a heavy
handed sort of way.
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Newline Fax: +1-760-731-3000
and strong
passwords to reduce the chance of nasty surprises.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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wait= yes
}
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On Sun, Jun 12, 2011 at 3:18 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 12 Jun 2011, bilal ghayyad wrote:
How can I see the the files are now loaded? Normally in the windows, I can
see that a request reached to the TFTP server and the files now are loaded
(upload or download
purpose.
Better safe than sorry.
Thanks,
Steve Totaro
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On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote:
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
On Thu, 9 Jun 2011, Hans Witvliet wrote:
I went originally from a almost working machine running:
asterisk180-1.8.3.2-87.1
To a machine that continuously
set it in sip.conf
Thanks,
Steve T
Thanks
Steve Totaro
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,
Steve T
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New
to asterisk ?
Does this help?
https://issues.asterisk.org/jira/browse/ASTERISK-17866
Regards,
Steve
Thanks steve,
But you know if i connect X-lite softphone my asterisk sending NOTIFY .
But its not sending NOTIFY to polycom 501 phone ? Do you think i need to
subscribe my phone
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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On Thu, Jun 9, 2011 at 7:32 PM, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 9 Jun 2011, Hans Witvliet wrote:
I went originally from a almost working machine running:
asterisk180-1.8.3.2-87.1
To a machine that continuously restarts asterisk (+core dumps) running:
asterisk180-1.8.3
asterisk not sending NOTIFY message ? Do i need
to subscribe my phone to asterisk ?
Does this help?
https://issues.asterisk.org/jira/browse/ASTERISK-17866
Regards,
Steve
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/Zap - - Extensions.conf
What will be the VoIP calling call flow in Incoming and outgoing calls?
Eth[x] - sip.conf - extensions.conf
--
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
to here the IVR sound files.
What is the problem in this case please help me..
NAT is a frequent culprit. Firewalls and iptables are also suspect.
Better details, better answers.
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