On Mon, 2003-11-10 at 05:54, Paulo Mannheimer wrote:
> Thanks Steven.
>
> I'll have to find a way to use bandwidth only when the call to the PSTN
> is completed on the other side.
Why does that matter? are you on a metered connection for bytes?
> [mailto:[EMAIL PROTECTED] On Behalf Of Steven
>
On Mon, 10 Nov 2003 22:34:54 -0500, Brian D Heaton wrote:
>Actually I use a USB pen drive. Its a lot tougher than a floppy and
>easier to carry around without damage.
Now that is a great idea
Now all we need is a usb handset (with ringer & maybe display) with
falsh as a drive.
carry around
i've just spent the pass 2 days trying to get AGI to work with PHP;
i made a lot of silly mistakes along the way which could have been
avoided if only there were some kinda howto or samples. at the risk
of looking stupid, i decided to shared my experience in hopes that
it might help some newbie g
On Fri, Nov 07, 2003 at 02:12:07PM +0530, ranga wrote:
> Can we use D-Link external modem with asterisk?
You are a bit vague on the details, so I will be vague on my answer:
maybe not. Modems with linux don't generally do full duplex voice.
Most users use Asterisk with Digium Hardware or other sp
Louis-David Mitterrand wrote:
Snip
The main problem with ipsec packets is the lack of TOS support: data and
voice traffic are agregated in one stream which is opaque to external
routers.
This is not the case with FreeS/WAN, below is an excerpt from the website:
Can I use Quality of Service ro
Hello,
We just did our own home-made, poor-man's Asterisk single-extension call
parking. It involves using several extensions that have MusicOnHold and then
using a client interface to send Manager:Redirect calls to send the parked
call to a specific extension:
exten => 871,1,Answer
exten => 871
Brian,
Its definitely something that needs some more thought. I agree with
Chris's points on client side implementation. I'm hoping Mark will
chime in here about what (if any) further thoughts he has about loading
some type of encryption into the IAXy.
THX/BDH
On Mon, 2003-11-10 at 21:54, Chris Albertson wrote:
> The PGP documentation suggestes that users cary their key
> in a floppy and never copy the key file to the hard disk.
> So your "little black plastic key" is a floppy with the write
> tab punched out.
Actually I use a USB pen drive. Its a lot
Pressing hold and the user hears music...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Posted At: Friday, November 07, 2003 9:56 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Putting call on hold
Subject: RE: [Asterisk-Us
> >
> I'd really like to see this. Maybe each user could have a little
> black
> plastic key they could insert and turn to go secure.
The PGP documentation suggestes that users cary their key
in a floppy and never copy the key file to the hard disk.
So your "little black plastic key" is a flopp
I agree with everyone's comments. I'm talking something a bit more light
weight to keep the casual network snooping from taking place. IPSEC
requires full control of both ends Not an ideal solution in some
cases. It was just a thought to see who all was intrested.
bkw
__
On Mon, 2003-11-10 at 20:25, Chris Albertson wrote:
> The below is all correct. In fact the US DoD has very restrictive
> and conservative rules about how some types of data are handled.
> Basically if it leaves a trusted area it will so through a hardware
> crypto box.
> Some of the rules are to
The below is all correct. In fact the US DoD has very restrictive
and conservative rules about how some types of data are handled.
Basically if it leaves a trusted area it will so through a hardware
crypto box.
Some of the rules are to ensure that data are protectied even if
the hardware is badly
This
is one of the best phones on the market. The only problem is getting a hold of
anyone at Polycom to get the specs through legitimate methods. I've tried for
the last several weeks to reach someone at Polycom to get into their developer
program, only to be sent from person to person and
I guess everyone already has these docs but me. Just in case.
http://www.loligo.com/asterisk/misc/GR_2BchanXfer.pdf
Generic Requirements for ISDN PRI Two B-ChannelTransfer
http://www.loligo.com/asterisk/misc/pri-tr41459_99.pdf
TECHNICAL REFERENCE 41459
AT&T Network
Integrated Services Digital Ne
Brian,
Mark was talking about it with JustinT at PN7. I caught the end of the
conversation. The question I asked then (and still ask now) is, (for
the IAX/IAX2 case at least) why load down the PBX with PBX-to-PBX
encryption?
If you look at the way most large organizations (mil
Mark,
On the first post, the include => default is two lines bellow [mainmenu],
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Farver
Sent: 10 November 2003 22:30
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Menu's & Sub-Menu's
O
On Mon, 10 Nov 2003 17:40:02 +0200
"Dan" <[EMAIL PROTECTED]> wrote:
> > and need 'callto:' support :-)
>
> Why you need this?
> Give me an example.
1) netmeeting and cuseeme are already support.
2) make a call easier with groupware and/or portal.
( 'tel:' is good too )
mack
__
Does any one have a Asterisk PBX running in the Portland, OR area? I'd
love to see one of these up and running?
-tyler
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I have a scenario that I just can't seem to
determine the best way to deal with.
We would like to have our PRI lines moved
to our datacenter with Asterisk directing calls between our 4 small
offices. Internet connections at the small offices are high-speed, but
*may* go down on a rare occa
Hello,
I have it working on distinguising just the local numbers of our 4 B channels
and the number assigned to the group. I have ordered an '100 in-dial range'
here in Australia and should have it available to me by the end of next week, I
can let you know how it goes.
Regards,
Matthew Enger
[EM
Hello all,
does anybody have experiences with DDI on ISDN BRI? I would like to
order DDI with 10 or 100 numbers from telco company, and use DDI
numbers for direct calling SIP/H323 clients. But I don't know if
Asterisk supports it, in archive I found only one answer about it, and
I'm not sure. I am
On Mon, 2003-11-10 at 15:28, David J Carter wrote:
> [insurance]
> exten => s,1,Background(insurance_thanks)
> exten => s,2,MusicOnHold(default)
> exten => s,3,Background(sorry_for_delay)
> exten => s,4,Goto(s,2)
> exten => s,5,Hangup
>
> if I enter an internal extension (range = 7001 to 7015) the
On Mon, Nov 10, 2003 at 03:22:43PM -0600, PJ Welsh wrote:
> On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote:
> >
> > I second that, and I think I remember hearing Mark talking about it too. But.
> >
> > What type of encryption can you do that does not introduce latency?
> >
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote:
>
> I second that, and I think I remember hearing Mark talking about it too. But.
>
> What type of encryption can you do that does not introduce latency?
>
> That said, I would like it to support hardware encryption cards.
>
> > Is it possible to incorporate iLBC codec, some hotels only allow 28.8
> > dial-up links, and then your product will be really useful on the
> > road
>
> How _does_ * work on dialup? I have never tried. I know you have an
> immediate 200-300ms lag but how is it otherwise?
>
I've used a
Hey guys - hate to beg, but my Cisco ID has expired (yes - I'm renewing) and I can't
get the latest ver 6.0 image for my SIP Phones - could anyone send me the .scp & .bin?
Of course this email never happened! :-)
Thanks!!
___
Asterisk-Users mailing li
Hi all,
I am trying to get a Menu system to work, and having probs with the internal
extensions from the prompts.
Below is the extensions.conf section.
[mainmenu]
;
;"main menu" context with submenu
;
include => default
exten => s,1,Answer
exten => s,2,Background(hello)
exten => s,3,Background(t
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote:
>
> I second that, and I think I remember hearing Mark talking about it too. But.
>
> What type of encryption can you do that does not introduce latency?
>
> That said, I would like it to support hardware encryption cards.
>
Facility IE seems to be the "ioctl" of the ISDN world. Even if you look
at the intel app note, it leaves out a lot of details about what's going
on. I'd love to get TBCT if anyone can find me a good, complete spec for
implementing it.
Mark
On Mon, 10 Nov 2003, Gene Kochanowsky wrote:
> Hi Gavi
> We are having problems with incoming calls on our X100P where callers
> try to dial 10, but the 1 gets detected twice and they end up on
> extension 11.
Since we're on the subject of DTMF, I'd like to ask how to solve "live"
dialpad problems of NON-SIP phones, such as anything hooked up to an F
Oh, cool. However, soxmix problem still remains. Soxmix is truncating
the input files!?!
-rw-r--r--1 root root70851 Nov 10 15:41 call5-in.gsm
-rw-r--r--1 root root75471 Nov 10 15:41 call5-out.gsm
-rw-r--r--1 root root 5082 Nov 10 15:42 call5-mix.
Hi all!
I´m testing an intracom sw netphone with asterisk,
someone have one netphone or have any experience to share about?
miklos
I use this on my 7960 to use blind xfer to parking.
exten => _2XX,1,Answer
exten => _2XX,2,Wait(1)
exten =>
_2XX,3,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|default,${EXTEN:1},1)
Ie if i'm on exten 11.. I blind xfer to 211. It waits 1 second.. calls me
back with the parking number
I second that, and I think I remember hearing Mark talking about it too. But.
What type of encryption can you do that does not introduce latency?
That said, I would like it to support hardware encryption cards.
I have done work with FreeS/WAN and it works, and yes it adds about 30-100ms of
You mean something like this:
backup-fs-1*CLI> show application directory
backup-fs-1*CLI>
-= Info about application 'Directory' =-
[Synopsis]:
Provide directory of voicemail extensions
I though digittimeout was the length of time to wait before expecting all digits to be
received during IVR type situations from the time the first digit is pressed.
The problem you may be having is sometimes especially cell-phones seem to have real
issues with asterisk and dtmf detection. Try re
On Mon, 10 Nov 2003, Mark Farver wrote:
> I see there is the DigitTimeout application that sets the maximum time
> between digits before asterisk will interpet.
>
> Is there any way to control the minimum?
>
> We are having problems with incoming calls on our X100P where callers
> try to dia
> >>>2) Put on hold and pick up on a different phone set.
> >>>
> >>>
> >>The right thing for this is call parking but it doesn't work to well
> >>with IP Phones..
> >>
> >
> >Could you clarify what doesn't work well? Is there a SIP deficiency?
> >
> It does not work so well becasue the IP Phone (
Hi,
- Original Message -
From: "Ariel Batista" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 10, 2003 9:09 PM
Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download
> -- Original Message --
> From: "Dan" <[EMA
No need to reverse the files now.. they are now padded out.
bkw
On Mon, 10 Nov 2003, David C. Troy wrote:
>
> Attempting to play with .gsm files generated by Monitor application, along
> the lines of what bkw suggested for merging channel files (reverse each
> channel, merge those, then reverse
I wonder if anyone else on the list has expressed any intrest in having
some type of native support for encryption for IAX? I hear IPSEC adds
some latency... I would like to side step that for something simpler to
setup.
bkw
___
Asterisk-Users mailing l
I see there is the DigitTimeout application that sets the maximum time
between digits before asterisk will interpet.
Is there any way to control the minimum?
We are having problems with incoming calls on our X100P where callers
try to dial 10, but the 1 gets detected twice and they end up on
--- John Harragin <[EMAIL PROTECTED]> wrote:
> I have a machine that crashes every so often. I believe the following
>
> I'm investigating this now. Is gdb the best (or easiest) tool for
> analyzing
> dumps?
Yes, likely all you need is to issue one "bt" (backtrace) command
to find the problem.
The Indianapolis Marion County Public Library has put out a
request for proposals on its website for a local dialtone/voice
system. I know there are several people on this list that
run consulting companies that specialize in implementing
Asterisk systems.
A PDF file describing the RFP proces
I have a machine that crashes every so often. I believe the following macro
is responsible (gotoif,$[${ARG3}] in particular). The macro works as
expected: if ARG3 is defined - hop over assignment. But my hunch is that it
gradually chews up memory.
; This macro is puts voicemail in an alternate mai
This Polycom
phone seems to be one of the best on the market for sound quality and features.
I have seen on the list that some people have gotten the IP 600 to work
with Asterisk. Does anyone have the details of how to get this working i.e.
XML phone config files, and any thing else I mig
Hi Gavin,
According to Sprint here in Tallahassee it is supported on the DMS100 for an NI2 ISDN
PRI. I am aware of the Intel tech note you refer to. You can find the code snippit it
contains spread all over the place. According to the docs it was tested on a DMS100
using an NI2 ISDN PRI. Howev
Source:
http://www.nanpa.com/number_resource_info/vsc_assignments.html
See also:
http://bugs.digium.com/bug_view_page.php?bug_id=071
Some (which?) of the codes below are hardcoded into Zap channels only.
Is there a European equivalent for this (or ITU / IETF)?
Greetings, Philipp
VERTICAL S
-- Original Message --
From: "Dan" <[EMAIL PROTECTED]>
>Hi,
>
>Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download
Dan,
Great to see the next version. I have now a problem with it. I can register without
problems and get calls and mak
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
Query2 ("code" fiel
At 10:33 AM 11/10/2003, you wrote:
Steve Underwood <[EMAIL PROTECTED]> wrote in news:3FAE487A.7000508
@coppice.org:
> Hi,
>
> I've kind of ported a DTMF text extry method I wrote some time ago for
> Dialogic. It is now a semi-working Asterisk app. I've still got to clean
> up some stuff in how Fest
TBCT (Two B Channel Transfer) would drastically reduce b-channel load in
some applications. Great for find me follow me Unified communicator type
services. Incoming calls answered on your cell phone etc. I am very
uneducated on this feature. How could I find out more about it? Does
anyone have
Steve Underwood <[EMAIL PROTECTED]> wrote in news:3FAE487A.7000508
@coppice.org:
> Hi,
>
> I've kind of ported a DTMF text extry method I wrote some time ago for
> Dialogic. It is now a semi-working Asterisk app. I've still got to clean
> up some stuff in how Festival is used to read back what
I have been having a lot of problems with SIP calls and
gotos within contexts as well as between contexts.
They work some of the time and fail some of the time
but the console reads the same either way. Am I the
only one having this problem? A little sample config
below.
[macro-stdexten]
exten =
On Wed, 5 Nov 2003, Dave Weis wrote:
> On Wed, 5 Nov 2003, Charles Hatchette wrote:
> > I am new to Asterisk and Digium card implementation issues. My VAR is
> > strongly recommending using Apple hardware and Yellow Dog Linux for my
> > telephony project, because of his familiarity with this OS. I
Attempting to play with .gsm files generated by Monitor application, along
the lines of what bkw suggested for merging channel files (reverse each
channel, merge those, then reverse the merged file). Anyway, that all
makes perfect sense.
The problem I have is with sox or with the .gsm files *
> SELECT *, length(code)
> FROM a
> WHERE code = '00442085673456'
>OR code = '0044208567345'
>OR code = '004420856734'
>OR code = '00442085673'
>OR code = '0044208567'
>OR code = '004420856'
>OR code = '00442085'
>OR code = '0044208'
>OR code = '004420'
>OR code
The cleanest way to do this would be to switch to
PostgreSQL and then define a "phone_number" type
and a "prefix_match" operator maybe calling it <>=
then you could use SQL like this:
Select data where Code <>= Number_dialed.
That's the best thing about Postgresql and other object/relational
D
Thorsten Lockert wrote:
SELECT *, LENGTH(code)
FROM a
WHERE code = left('00442085673456', LENGTH(code))
ORDER BY LENGTH(code) DESC
LIMIT 1;
Awesome, thanks Thorsten..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/li
rnc Info Lists wrote:
Is anyone running Asterisk under Fedora Core 1
(http://fedora.redhat.com/)?
If so, did everything with Asterisk work properly? I'm looking to migrate
from Red Hat 8.0 to Fedora this week.
Thanks.
Interesting question... Since RedHat will in the future have only their
Enterpri
Hi list!
I have done a H.323 - SIP Gateway. The SIP Register is within Asterisk, so all
the SIP phones are known and are all registed in Asterisk. Now it works ok,
because there are no so many phones connected, about four, but I would like to
know if when the SIP network would be bigger, if there
[EMAIL PROTECTED] wrote:
> exten =>> _1800XXX,1,SetCallerID(${CALLERIDNUM})
>
> That should be =>
Ah yes, search and replace without forethought or inspection
to include my previous email indented with ">>" for informational
purposes. Alas, it does not work even with "=>".
My experience with Fedora was a little unpleasant. I found it to have quite
a few problems that I could not overlook. When I checked up2date there were
no updates available yet. I would wait until some bugs have been worked out
before transitioning to Fedora.
Regards,
Mike
P.S. After my experienc
I have just completed a set of voice files in Dutch, plus a patch that
forces Asterisk to sane (i.e. Dutch ;-)) behaviour when composing dates,
times, numbers, etcetera.
The current release, 0.0.1, is a sort of pre-release - some known issues
have been identified, but I nevertheless would like to
F.G.Testa <[EMAIL PROTECTED]> said:
>Hi! I'm new here, and I was wondering if there is any newsgroup gateway
>to the Asterisk lists?
>
Yes, this list is gated at gmane.org. However, you'll need to be
subscribed to the mailing list in order to be able to post, AFAIK.
--
Cees de Groot
Hello,
Almost agree. I have fixed one or two problems by swapping motherboards. But changing
to APIC alone will not do it, you also have to enable those IRQ's for APIC. If you
don't it won't use it. You can double check this by "cat /proc/interrupts" and it will
still say XT-PIC next to all of
Hi,
- Original Message -
From: "Masakazu Nakano" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 10, 2003 4:26 PM
Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download
> > > strange function in call,when it stilln't registered
> > > ( like a no home
SELECT *, LENGTH(code)
FROM a
WHERE code = left('00442085673456', LENGTH(code))
ORDER BY LENGTH(code) DESC
LIMIT 1;
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Monday, November 10, 2003 9:33
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-U
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 10 November 2003 15:31, Eric Wieling wrote:
> > I now have an E100P, an (possibly defective) E400P and two TE410P cards.
> > Only the E100P works flawlessly. The E400P doesn't work with the newest
> > zaptel driver and the one TE410P card I'v
Hi,
I would like to test chan_sip with a bigger jitter buffer. Does anybody know
where in the code this is defined? I looked through it but could not find
where.
If anybody else can find it please let me know.
Regards,
Andres
___
Asterisk-Users mai
Anyone a list one what basic Debian packages are need to compile * and get
it up and running ?
I'd like to run * on a clean as possible system so looks like debian is the
way to go in that...
Greetings,
Tj
- Original Message -
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Aste
> Is anyone running Asterisk under Fedora Core 1
> (http://fedora.redhat.com/)?
> If so, did everything with Asterisk work properly? I'm looking to migrate
> from Red Hat 8.0 to Fedora this week.
>
> Thanks.
>
Interesting question... Since RedHat will in the future have only their
Enterprise versio
> So if you only have smp systems with ohci and no zaptel cards (because it's a
> sip/iax2 gateway) then you're screwed?
You can always get an x100p...
mark
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exten =>> _1800XXX,1,SetCallerID(${CALLERIDNUM})
That should be =>
On Mon, 2003-11-10 at 08:15, Barton Hodges wrote:
> Barton Hodges wrote:
> >> Hi,
> >>
> >> Does anyone know how to dial toll-free (800) numbers through FWD or
> >> Siphone?
> >>
> >> Using the configuration below, I can d
Reinhard Max wrote:
On Mon, 10 Nov 2003 at 13:34, WipeOut wrote:
I guess I am going to have to look through the table multiple times
dropping the last digit on each select until I get a result..
You could also try this one to see which one is faster:
SELECT *, length(code)
FROM a
WHER
Is there a way to have a Cisco SIP gateway register with Asterisk?
The current setup just drops calls into the sip.conf default context
which works fine but has some security risks since anyone who can
install XTEN and has access to my LAN can then use this context to drop
calls in
I'd lik
Try a different motherboard. More than a few people have solved their
problems by switching to a different motherboard. If you are having
problems with the ZapTel cards sharing IRQs then try enabling APIC on
UniProcessor machines (it's a config option when building the kernel)
On Mon, 2003-11-10
On Mon, 2003-11-10 at 07:10, Roy Sigurd Karlsbakk wrote:
> sox asdf.gsm asdf.wav
>
> ...and use your favourite sound clip editor
To expand a little on Roy's answer, the GSM format we use in asterisk
has no header on it. The sox line Roy supplied will put the raw gsm file
into RIFF format so your
On Mon, 2003-11-10 at 15:03, Roy Sigurd Karlsbakk wrote:
> Why don't you just move to Debian? A lot cleaner and better IMHO
IMHO stick with the devil you know.
RH 9 -> 10 (fedora) would just be evolution.
RH -> Debian would require a learning curve due to the different ways of
doing things. I'm
On Mon, 10 Nov 2003 08:08:00 +0200
"Dan" <[EMAIL PROTECTED]> wrote:
> Hi,
>
> - Original Message -
> From: "Masakazu Nakano" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, November 10, 2003 1:22 AM
> Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download
> Or you can spend the $100 and get an X100P to get zaptel timing and
> support Digium at the same time..
... provided that he has any spare slots.
Regards,
Andrew
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On Mon, 2003-11-10 at 06:49, F.G.Testa wrote:
> Hi! I'm new here, and I was wondering if there is any newsgroup gateway
> to the Asterisk lists?
No. Closest thing you will find to that is by reading the archive, but
you can't do it with normal newsgroup software.
If you do read the archive you wi
> You could also try this one to see which one is faster:
>
> SELECT *, length(code)
> FROM a
> WHERE code = '00442085673456'
> OR code = '0044208567345'
> OR code = '004420856734'
> OR code = '00442085673'
> OR code = '0044208567'
> OR code = '004420856'
> OR c
Dipak,
Look in /etc/services
This file has most common and any RH specific port number assignments.
Alternately, you can look at
http://www.iana.org/assignments/port-numbers
Which is a much more comprehensive list.
-sb
-Original Message-
From: DIPAK PAUL [mailto:[EMAIL PROTECTED]
Sen
Barton Hodges wrote:
>> Hi,
>>
>> Does anyone know how to dial toll-free (800) numbers through FWD or
>> Siphone?
>>
>> Using the configuration below, I can dial out to SIPphone.com users
>> by simply dialing their number (1747XXX) and can dial out to FWD
>> users by dialing 1383>
>>
>> Ho
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 10 November 2003 14:37, WipeOut wrote:
> >So if you only have smp systems with ohci and no zaptel cards (because
> > it's a sip/iax2 gateway) then you're screwed?
> Pretty much..
> Or you can spend the $100 and get an X100P to get zaptel timi
On Mon, 10 Nov 2003 at 14:55, Michael Bielicki wrote:
> Just as a tip you could do that with regex functions as well,
> omitting the quite costly length functions ..
Well, regular expressions can be costly as well, and it depends on the
DBMS implementation how costly length() is. If the actual le
Why don't you just move to Debian? A lot cleaner and better IMHO
On Mon, 2003-11-10 at 14:34, [EMAIL PROTECTED] wrote:
>
>
> Is anyone running Asterisk under Fedora Core 1 (http://fedora.redhat.com/)?
> If so, did everything with Asterisk work properly? I'm looking to migrate
> from Red Hat 8.0
On Mon, 10 Nov 2003 at 13:34, WipeOut wrote:
> I guess I am going to have to look through the table multiple times
> dropping the last digit on each select until I get a result..
You could also try this one to see which one is faster:
SELECT *, length(code)
FROM a
WHERE code = '00442085673
Greetings,
This may be a bit arcane but does anyone know what the contents of a facility message
should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on
a DMS100 for the last four months to no avail. The message I am currently sending
makes it to the switch but is
Just as a tip you could do that with regex functions as well, omitting the
quite costly length functions ..
On Monday 10 November 2003 1:11 pm, Reinhard Max wrote:
> On Mon, 10 Nov 2003 at 12:33, Reinhard Max wrote:
> > SELECT DISTINCT *,length(code)
> > FROM a
> > WHERE '00442085673456' LIKE
> > Is it possible to incorporate iLBC codec, some hotels only allow 28.8
> > dial-up links, and then your product will be really useful on the
> > road
>
> How _does_ * work on dialup? I have never tried. I know you have an
> immediate 200-300ms lag but how is it otherwise?
I have very sati
[EMAIL PROTECTED] wrote:
Is anyone running Asterisk under Fedora Core 1 (http://fedora.redhat.com/)?
If so, did everything with Asterisk work properly? I'm looking to migrate
from Red Hat 8.0 to Fedora this week.
Thanks.
From what I have seen in playing with it for the last day and a bit is
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is
provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and
ztrtc can't run on smp systems.
So if you only have smp systems with ohci
Reinhard Max wrote:
On Mon, 10 Nov 2003 at 12:33, Reinhard Max wrote:
SELECT DISTINCT *,length(code)
FROM a
WHERE '00442085673456' LIKE (code || '%')
ORDER length(code) DESC;
^
BY
Oops, that one got lost when I re-formatted the query.
cu
Reinhard
Hi Reinhard,
Th
Is anyone running Asterisk under Fedora Core 1 (http://fedora.redhat.com/)?
If so, did everything with Asterisk work properly? I'm looking to migrate
from Red Hat 8.0 to Fedora this week.
Thanks.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
htt
> So if you only have smp systems with ohci and no zaptel cards (because it's a
> sip/iax2 gateway) then you're screwed?
I tried to use IAX2 trunking, on a single CPU system with usb-uhci, and
neither zaprtc nor ztdummy made the 'zaptel timing needed' errors go
away :(
So I'd be interested in a
On Mon, 27 Oct 2003, Perry E. Metzger wrote:
> If your DSL link is the bottleneck, rather than earlier hops back
> through the providers network, the provider could also prioritize VOIP
> packets going up the DSL line. That requires a cooperating provider,
> of course.
Not strictly.
Other TCP tra
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is
provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and
ztrtc can't run on smp systems.
So if you only have smp systems with ohci and no zaptel cards (
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