.
For future reference...
A better subject may yield better answers.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731
,hangup()
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On Sat, Sep 29, 2012 at 6:49 AM, Markus unive...@truemetal.org wrote:
Am 29.09.2012 10:49, schrieb resea...@businesstz.com:
[tz-ivr01 ~]# uptime
11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57
Sharing is caring
Is that a Quad Core CPU in your box?
PS: Yes,
to make it do what
you want :)
A multi-threaded AGI or externalivr() sound easier to me if it meets your
needs.
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Newline
to consider is the externalivr() application.
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ended' pattern could also expose you to unexpected
outcomes.
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clues.
Judging from 7+ years lurking on this list and never seeing anything like
this, it's something weird about your install.
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be a useful tool to trigger a script to create an Asterisk
call file.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760
Un-top-posting...
On Fri, 14 Sep 2012, RAJNI VANZA wrote:
I was wondering if anyone has any experience for recording user voice
while play background music?
On Fri, Sep 14, 2012 at 11:13 AM, Steve Edwards wrote:
What methods have you tried?
On Fri, 14 Sep 2012, RAJNI VANZA wrote:
I
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On Fri, 14 Sep 2012, RAJNI VANZA wrote:
I was wondering if anyone has any experience for recording user voice
while play background music?
What methods have you tried?
--
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,
-
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Newline Fax: +1-760-731-3000
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On 1 September 2012 09:08, Olle E. Johansson o...@edvina.net wrote:
31 aug 2012 kl. 13:13 skrev Steve Davies davies...@gmail.com:
On 31 August 2012 07:49, Olle E. Johansson o...@edvina.net wrote:
24 aug 2012 kl. 16:18 skrev Steve Davies davies...@gmail.com:
Hi SIP Gurus,
I've tried
.
Thoughts please?
Many thanks,
Steve
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On 24 August 2012 15:34, Faisal Hanif fai...@vopium.com wrote:
Steve Davies davies...@gmail.com wrote:
Hi SIP Gurus,
I've tried to find the relevant RFCs, but am struggling. I can find
the odd opinion online, but was wondering if anyone could give a
definitive answer.
If a SIP call is initiated
if you find that they are digital.
Thanks,
Steve Totaro
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On 08/19/2012 11:45 AM, Lee Howard wrote:
On 08/17/2012 04:58 AM, Steve Underwood wrote:
On 08/17/2012 06:08 AM, Eric Wieling wrote:
Has anyone experimented with increasing the DAHDI chunk size in
improve fax reliability? If so, did it help, hurt, or not make any
difference?
I haven't
system for
some reason it can badly affect the reliability of FAXes.
Steve
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in the actual FAX exchange?
Steve
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?
Both sipp and sipsak can send REGISTER packets.
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in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
,
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Newline Fax: +1-760-731-3000
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will issue a new set every few months.
As always, if you need custom prompts you can order them from Allison on the
IVR prompts page on Digium.com.
Thanks,
-S
Steve Sokol
Asterisk Marketing Director | Digium, Inc.
31762 Timberlake Drive, Crystal Lakes, MO 64024
Direct: +1 256 428
=openmeetingshttps://www.google.com/#q=openmeetings+site:youtube.comsa=Xei=vfEWUJjAB-200QGI4oH4CAved=0CKgBENsBhl=enbav=on.2,or.r_gc.r_pw.r_cp.r_qf.,cf.osbfp=1f871a50839302aebiw=1280bih=576
Looks like it will be pretty cool.
Thanks,
Steve T
configuration is documented
here:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
One thing that page doesn't mention is only spandsp supports T.38
gateway right now. The Digium FAX module does not.
Regards,
Steve
a
different SIP header and the correct bits twiddled in it's humongous
configuration files.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
Please don't top-post.
On Tue, 10 Jul 2012, bilal ghayyad wrote:
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Newline Fax: +1
in advance,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
on the shared device and then
moving the call file to the outgoing spool directory.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
Hi David,
The old app_fax code, which allowed spandsp to be used with Asterisk
before Digium introduced the new modules supported the features you
want. Maybe someone can go through that code and port the feature into
the current res-fax code.
Steve
On 07/03/2012 09:57 AM, David Cunningham
responses.
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Newline Fax: +1-760-731-3000
card's E1 interface from
the PSTN.
Steve
From the asterisk console I can see the receiving fax session running,
but halfway it stops due to timeout or hangup.
Below is a fax session output which was marked as failed:
-- Channel 'DAHDI/i1/-4' receiving FAX
'/var/spool/asterisk/fax/fax-65126150
, but I don't
know if the Asterisk module exposes that facility.
Steve
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I am looking for a CDR report tool that will link extensions to the
user's names... are there any that offer this feature? We are using
trixbox 2.8.
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New
.
Steve
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asterisk-users mailing list
easily :)
http://archives.manageengine.com/vqmanager/7011/ManageEngine_VQManager.bin
Reverts to 'free' version after 30 days.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
and T2_TIMEOUT.
Any suggestions on how to improve the fax receiving rate?
I have a problem. Can you fix it? is not really a meaningful question.
Steve
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Newline Fax: +1-760-731-3000
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a letter absolving me of liability signed by the CEO in my back
pocket.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax
,dial(sip/security)
exten = *,n,hangup()
This shows 'Tora! Tora! Tora!' as line 1 and 'Main Lobby' on my cisco
7960.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1
the prompts on an occasional basis and we will get them set up
for download.
Thanks,
-S
Steve Sokol
Asterisk Marketing Director | Digium, Inc.
31762 Timberlake Drive, Crystal Lakes, MO 64024
Direct: +1 256 428 6101
Mobile: +1 816 806 8844
Ask me about Asterisk
]any in case a host smokes.
I like storing CDRs in a database (MySQL is my choice) so '1' is easy and
'2' may be handled with database replication.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1
when your AGI takes too long?
*) If I get a chance to re-code it, I'd just write a database record and
then cobble up a daemon or cron job to do the heavy (and time consuming)
lifting.
--
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-
Steve Edwards
an AGI
(meetmeadmin-by-index) before returning the admin to the conference.
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Newline
with the magnitude of the delay and
that most people manage to carry on conversations without noticing.
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
On 05/17/2012 02:47 PM, gincantalupo wrote:
Hi Steve,
you are telling me there is no way to set a particular speed on my
iaxmodem in order to force the sender speed?
I have some problems with a customer who gets malformed faxes even if
no error occurs. Since I cannot tell the sender to lower
their results.
Steve
On 05/18/2012 09:38 AM, Sebastian Gutierrez wrote:
Rusty,
thanks for the reply, the issue seems a spandsp issue, I changed to
digium free asterisk fax and works much better, has still some issues
that not all faxes pass ok, but does the work.
thanks!
On May 17, 2012, at 1:06
.29+V.17 are OK. So, if you allow the
7200bps mode of V.29 you are compelled to allows the 9600bps mode too.
Steve
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in progress to finish.
Then you can install patches, replace failing disks, etc, etc, etc.
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Newline
) teams.
**) I skip hosts that are supposed to be exact clones of other hosts.
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Newline
,
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,
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On Fri, 4 May 2012, Jonas Kellens wrote:
Terminated with signal 6, what is that ?
kill -l
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Newline
or commetrex on Asterisk ?
Does it really matter whether I use Asterisk or FreeSWITCH ?
regards,
Anita
Steve
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On 25 April 2012 18:05, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 11:54 AM, Steve Davies wrote:
A further question... It appears that for SIP endpoints, this facility
only updates RPID and PAI headers? I have found that there appear to
be 4 different SIP CID-update mechanisms
the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?
Thanks,
Steve
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of these other methods? If
not, I will almost certainly add them for my own use, and submit the
code.
Regards,
Steve
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from a 'can do' perspective. I'd still vote for
separate boxes if the budget can support it.
Hey OP, better details yield better suggestions :)
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Steve Edwards sedwa...@sedwards.com Voice: +1
layer as R2 and ISDN. It is an
alternative to them, not another layer.
Steve
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MyTrunk to = ${myglobalvar}
set(${l_databaseVariableName}=${${l_databaseVariableValue}})
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Newline
/Mean_opinion_score
There's lies, damn lies and mean opinion scores. The chart on that
wikipedia page is mostly for humour value.
Regards,
Steve
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New
that they are
confusing mysql() somehow.
Then, I'd crack open another beer and reach for a book on c or PHP or
Perl.
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Newline
/ and are named asterisk.ctl
and asterisk.pid
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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On Wed, 4 Apr 2012, Vik Killa wrote:
Please don't feed the trolls.
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Newline Fax
to 'friend' and use
username/password based authentication.
Buyer beware - I believe the above to be true, and I hope it makes sense!
Regards,
Steve
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'command: meetme list meetme-name' and then parse the output and then do
something with it like set the 'user number' as a channel variable.
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in Mediatrix boxes. They usually
work OK.
Regards,
Steve
On 03/27/2012 08:02 AM, Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
- -BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
I'm having difficulties when receiving faxes from the PSTN with this
relatively simple
if the 240 channels are distributed to a bunch of
separate 'meetmes' or is the assumption that all 240 are in a single
meetme?
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Newline
of thing? Any words of wisdom?
murf
--
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and relate it to your
application -- congratulations :)
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Newline Fax: +1-760-731
G729a is used?
Significant.
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in advance,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On Sat, 10 Mar 2012, Steve Edwards wrote:
The retrieved row contains a bunch (up to about a hundred) of columns, most
s/row/rows/
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468
DAHDI did.
I may fire up a Debian Lenny VM and see if the fork with the patches match
up and work, and then if app_rpt and app_radio compile or throw an error.
The latest all in one ISO uses CentOS 5.7.
Thanks,
Steve Totaro
2012/3/9 Paul Belanger pabelan...@digium.com
On 12-03-09 03:18 AM, Márkus Béla wrote:
how can I add/enable app_rpt module to Asterisk 1.8?
Make sure DAHDI is installed. However, there is a patch on
reviewboard[1] that will see this module be removed from asterisk.
The code is out-dated
On Fri, Mar 9, 2012 at 8:52 AM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
2012/3/9 Paul Belanger pabelan...@digium.com
On 12-03-09 03:18 AM, Márkus Béla wrote:
how can I add/enable app_rpt module to Asterisk 1.8?
Make sure DAHDI is installed. However, there is a patch
.
Obviously, these security issues should be patched, but I feel that in my
implementations, things are very secure.
Thanks,
Steve T
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channel variables in the blink of an eye. The channel
variables then control the flow of the script - what kind of questions are
asked and what to do with the responses.
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digging.
Thanks,
Steve T
-- Forwarded message --
From: Shea Caughron scaugh...@digium.com
Date: Thu, Mar 8, 2012 at 8:20 AM
Subject: Do you know how Asterisk came to be?
To: stot...@asteriskhelpdesk.com
**
View this email on your mobile device or
onlinehttp
,
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New
advise me, why to use all this ling argument and not using only
call_log?
Maybe a Vicidial mailing list would yield more 'on target' answers.
*) Google.com
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,
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:-)
Could you help me solve this problem?
Thank you!
Steve
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On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Yes, I have had no problems with Grandstream first gen ATAs, configured
with
server and credentials and shipped off, they just
to the
device as unsolicited and drop it. That is a function of the router but
keep alives from Qualify on the Asterisk side, and setting the device to
register every few minutes will keep that mapping open and alive, letting
traffic pass as solicited.
Thanks,
Steve Totaro
On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote:
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Agreed with one exception, the endpoint behind the NAT DOES need to be
setup
correctly to keep the router from seeing inbound
ones. Then it turned
into a pissing contest, like you say, it happens in every list with
the topic this or that.
Again, as I pointed out to Steve above, and after reading all of your
responses, our SIP/NAT woes seem obviously ignorance on our part, but
that doesn't shadow the fact that IAX2
only
use SIP and use OpenVPN.
I build Asterisk from source and menuconfig, I remove all that is not
needed, including IAX2. I do download all the sound files in different
languages and codecs.
It just works. I like things that just work.
Thanks,
Steve Totaro
On Tue, Feb 28, 2012 at 5:17 PM
Roger That, I am an IC. I contract with the Government to little ten phone
shops. From VA/MD/DC area, I have been contracted and flown in to many
large call center locations that were CONUS and OCONUS.
My facebook is Steve Totaro in Reston VA. LinkedIN is more accurate, but
my resume speaks
. These days it offers no real
advantages in our opinion.
On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
People around here either hate me or love me. I post experience and am
accused of bragging or whatever. As a reader, I want to know who is
giving
me
They said the same thing in 2005, 2008, now Every release.
You never answered the question as to why you don't want to use SIP. Is
there a reason, or do you just want to torture yourself?
Thanks,
Steve T
On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote
combined into one stream.
Without trunking, you only have the single port thing. It is quite easy to
open the correct ports for SIP, some just have GUI with a SIP checkbox,
IPTables is simple and there are tons of howtos.
Thanks,
Steve T
On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro
stot
that the word Ghetto means anything above, or a racial
slur should look up the true definition.
It isn't even an insult to the Asterisk Community. By definition, the
Asterisk Community is an online Ghetto.
Just wanted to clear that up before someone tries to label you.
Thanks,
Steve T
On Tue, Feb 28
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