-users] asterisk config file online editor
No problem, hope it gets you where you need to be :)
Moj
Anton Krall wrote:
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division
to the file.
Moj
Anton Krall wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an
interface),
any recommendations?
Anton Krall
Direccion
some speech files.
Hope this helps
AK
-Original Message-
From: bilal ghayyad [mailto:[EMAIL PROTECTED]
Sent: miércoles, 16 de enero de 2008 08:07 a.m.
To: asterisk-users@lists.digium.com
Cc: Anton Krall
Subject: Re: app_voicemail for spanish
Hi AK;
I would like to ask a question
Will do
AK
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: lunes, 14 de enero de 2008 11:48 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] app_voicemail for spanish
No features are
Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish
prompts that can handle for example, instead of saying trabajo mensjes
would say mensajes de trabajo o mensajes trabajo (inverse)? Also can
handle singular and plural (mensaje vs. mensajes)?
Anton
recall which
right now)
But if further language support is needed you should file a bugreport.
On Jan 14, 2008 5:04 PM, Anton Krall [EMAIL PROTECTED] wrote:
Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish
prompts that can handle for example, instead of saying trabajo
mensjes
Thank you for the example Isaac. I did as you mentioned and now it seems to
be working perfectly.
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: jueves, 13 de septiembre de 2007 10:33 p.m.
To: asterisk-users@lists.digium.com
Thank Isaac, Ill try it this way.. Im currently using this before entering
the queue so calls from the queue are recorded:
exten =
s,n,SetVar(MONITOR_FILENAME=/var/spool/asterisk/${TIMESTAMP}-${UNIQUEID}-${C
ALLERIDNUM}-Queue-Ventas)
exten = s,n,SetVar(TRANSFER_CONTEXT=internalphones)
So I could
GUys.. I dont know if this is a known bug or not but I just tested and
replicated this one over and over again.
It involves call transfer from calls that entered the pbx via a queue.. say
a call comes in and its thrown in a queue, somebody answers the call but
then wants to transfer the call to
me is still better fo you.
Why do you want to move away from meetme?
On 9/1/07, Anton Krall [EMAIL PROTECTED] wrote:
Hi Moises.
So, would you recommend app_conference over meetme? Knowing what you know
about it?
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto
the time/skills to fix it.
Moy
On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote:
Is app_conference designed only for 1.4? I tried compiling against 1.2.24
and but get a no such file while looking for autoconf.h which is a file
only
used in 1.4... anybody running app_conference on 1.2
Is app_conference designed only for 1.4? I tried compiling against 1.2.24
and but get a no such file while looking for autoconf.h which is a file only
used in 1.4... anybody running app_conference on 1.2?
___
--Bandwidth and Colocation Provided by
Is app_conference designed only for 1.4? I tried compiling against 1.2.24
and but get a no such file while looking for autoconf.h which is a file only
used in 1.4... anybody running app_conference on 1.2?
___
--Bandwidth and Colocation Provided by
to call
I am using the free
http://www.babarnazmi.citril.com/forum/viewtopic.php?t=7sid=fd8047cffb13074
969d3418064f4eb31
It is working as you described.
It appears to be working well.
--
--
Steven
http://www.glimasoutheast.org
Anton Krall [EMAIL PROTECTED] wrote in message
news:[EMAIL
Jun 2007, Anton Krall wrote:
So Guys, no go on this topic?
I trialled a click-to-dial application recently. It generated a lot of
controversy on the list (search the archives) because various people said
it couldn't be done/wouldn't work, etc. Then there were whinges about the
commercial nature
calls.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Friday, June 01, 2007 10:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
Thank you for the explanation Dean, you are right on the money and could be
more precise.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Sábado, 02 de Junio de 2007 04:34 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
So Guys, no go on this topic?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Jueves, 31 de Mayo de 2007 10:58 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] click to call
The idea
I have been looking around for examples or code on making a click to call
application for web sites... has anybody had any luck on this topic? Is
there any open source code out ther that could do this?
Regards
AK
___
--Bandwidth and Colocation
de Mayo de 2007 10:18 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] click to call
Anton Krall wrote:
I have been looking around for examples or code on making a click to call
application for web sites... has anybody had any luck on this topic
Hi Guys..
I want to see what the R2mfc community has been up to. Anybody moved to 1.4?
what have you done regarding unicall? Any updates or are you stuck with
1.2.X too?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
Microsoft blocking access to
|that phone speaker.
|The claim that allowing the developers to access it would allow for
|invasion of privacy (like recording phone calls).
|So unless someone can work around this, softphones for WM will remain
|quite useless.
|
|Timothy.
|
|Anton Krall wrote:
| Guys
Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen
devices
|
|I've been trying the SJPhone with no luck. Where did you download the
|Xten version from?
|
|On 1/16/07, Anton Krall [EMAIL PROTECTED] wrote:
| Guys, anybody has seen
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work on Wm5 but they are designed for standard screens, anybody using
anything on square ones?
___
|
|-BEGIN PGP SIGNED MESSAGE-
|Hash: SHA1
|
|Eric ManxPower Wieling wrote:
| Anton Krall wrote:
| This is exactly one of the things that Steve and I discussed a bit
ago...
| when did asterisk turn from an open source project with very good
| developers
| into a business that only focuses
- Non-Commercial Discussion
|Subject: Re: [asterisk-users] no unicall on 1.4
|
|On 1/5/07, Anton Krall [EMAIL PROTECTED] wrote:
| I think you are misunderstanding several points here Moises.
|May be
|
| I do give Digium a break like you said, that's why you have options
|I dont understand this. How
: Thursday, January 04, 2007 10:14 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] no unicall on 1.4
|
|Anton Krall wrote:
|
|This is exactly one of the things that Steve and I discussed a bit ago...
|when did asterisk turn from an open source project
in asterisk 1.4 and 1.2?
On 1/4/07, Anton Krall [EMAIL PROTECTED] wrote:
Well Moises, if you do, please drop me a line and I will gladly test it.
I was mentioning digium because AFAIK, the guys at digium are in touch
with
the programmers and contributors so I thought maybe they would
! :)
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Moises Silva
|Sent: Friday, January 05, 2007 9:41 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] no unicall on 1.4
|
|On 1/5/07, Anton Krall [EMAIL
: Wednesday, January 03, 2007 5:22 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call
Announcement...)
|
|On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote:
| And probably wont be as Steve Underwood explained to me that he
This is exactly one of the things that Steve and I discussed a bit ago...
when did asterisk turn from an open source project with very good developers
into a business that only focuses in $$$?
Thats why openpbx was born I guess
For example, samba is still free, and people are making a
And probably wont be as Steve Underwood explained to me that he is now
supporting openpbx and has stopped support for unicall on asterisk 1.4
Can anybody at digium confirm? Is unicall going to be left out of 1.4?
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
No update on unicall and 1.4?
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Anton Krall
|Sent: Tuesday, December 26, 2006 6:15 AM
|To: asterisk-users@lists.digium.com
|Subject: [asterisk-users] 1.4 and unicall
|
|Guys, anybody knows
|complete on the incoming part of the protocol, but the outgoing logic is
|kind of crude.
|I wonder if Steve Underwood is still actively working on it.
|
|BarZ
|
|Anton Krall wrote:
| No update on unicall and 1.4?
|
| |-Original Message-
| |From: [EMAIL PROTECTED] [mailto:asterisk-users-
| |[EMAIL
Guys, anybody knows if 1.4 has support for unicall or if/which version of
unicall will compile on it?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Too bad Cepstral hasnt still made a decent Spanish voice, the ones they
have still sound too computer like, not like the English ones they have
which sound great!
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Julian J. M.
|Sent:
Guys.
I was able to put asterisk on debug for a while and another shutdown took
place, here are the logs... I was wondering.. I notice that we are using a
lot of manager actions and can it be that if you place a lot of manager
actions in a short period of time, that it makes asterisk do a
PROTECTED] On Behalf Of
|Anton Krall
|Sent: Friday, June 30, 2006 9:11 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] asterisk shutdown
|
|No log entries yet that might show whats happening and you are
|correct, I cant run under strace as it would hit
, Anton Krall wrote:
| So, no answers? Nobody knowd why this might be happening?
|Nobody else
| experiencing this?
|
|Is this a reproducable issue? Have you turned on verbosity and
|debug and log them (e.g. the full log)?
|
|If still no messages and this is reproducable, consider
|running
So, no answers? Nobody knowd why this might be happening? Nobody else
experiencing this?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Anton Krall
|Sent: Wednesday, June 28, 2006 7:03 PM
|To: 'Asterisk Users Mailing List - Non-Commercial
Guys.
Ive seen on my asterisk messages log that asterisk has shutdown itself about
12 times in 5 days... The logs show nothing but:
[Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call
[Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call
[Jun 28 09:40:02] VERBOSE[3172]: [Jun 28
Same version, same problem...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Doug Lytle
|Sent: Wednesday, June 28, 2006 11:42 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] asterisk shutdown
|
|Anton
Im my case, the box is closed down so I dont think its an
intruder issue... Im puzzled...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
PiperSent: Wednesday, June 28, 2006 4:41 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
|Subject: Re: [Asterisk-Users] sangoma unicall m2rfc
|
|Anton Krall wrote:
|
|Uys, Steve Underwood
|
|I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for
|R2MFC, I get the far and local end unblocked but as soon as I try to
|make a call I get dialing and then protocol failure
Uys, Steve Underwood
I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I
get the far and local end unblocked but as soon as I try to make a call I
get dialing and then protocol failure..
Do you guys know if there are any issues with sangoma and unicall? Anybody
has an
Are you around Steve?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Anton Krall
|Sent: Monday, June 19, 2006 11:58 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] sangoma unicall m2rfc
|
|Uys, Steve
|Subject: Re: [Asterisk-Users] sangoma unicall m2rfc
|
|Anton Krall wrote:
|
|Uys, Steve Underwood
|
|I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for
|R2MFC, I get the far and local end unblocked but as soon as I try to
|make a call I get dialing and then protocol failure..
|
|Do you
other files that you can download.
|Check http://moy.ivsol.net/
|
|Regards
|
|On 6/19/06, Anton Krall [EMAIL PROTECTED] wrote:
| Any particular tips I should consider? Its very weird why I get
| protocol failure with sangoma and using the same config with digium
| cards it works ok.
|
| BTW how can
Guys, is there a way to set CDR vards like SRC, I tried using set but
asterisk complains they are RO vars. What Im trying to do is a small way to
let users make calls from someone elses extension but auth using a password
and seitch credential to their own so the call appears on CDR as made from
Guys.
I have a couple of agis that when trying to dial a local call, LD, etc. ask
the user for a password and then checks against a DB to see if they can call
or not.
My newi dea here is to allow users to roam between extensions, for example,
user 1 can go to users 2 phone and when ask for the
Guys.
I have a couple of agis that when trying to dial a local call, LD, etc. ask
the user for a password and then checks against a DB to see if they can call
or not.
My newi dea here is to allow users to roam between extensions, for example,
user 1 can go to users 2 phone and when ask for the
DEFAULT_T1 2
Espero te sirva.
On 5/30/06, Anton
Krall [EMAIL PROTECTED]
wrote:
Steve
Underwood:Steve, why do some numbers give protocol errors? Ive
noticed here in Mexicothat certain numbers when dialed return protocol
failure and a busy tone.Any idea why this happens
me configuration on
Polycom Soundpoint501phone
Could not find your post for 4 months ago.
--
Original message ------ From: "Anton Krall"
[EMAIL PROTECTED] Yes, check a post that I
made about 4 months ago, I posted the cofig for setting th
principio, de ahora en adelante en todas la instalaciones qe
hago codificamos ese parametro
On 6/2/06, Anton
Krall
[EMAIL PROTECTED] wrote:
Muchas
gracias Felix, voy a probar a ver que tal jala.
Tu
tuviste ese miusmo problema?
From
Steve Underwood:
Steve, why do some numbers give protocol errors? Ive noticed here in Mexico
that certain numbers when dialed return protocol failure and a busy tone.
Any idea why this happens and why with only certain phone numbers?
___
--Bandwidth
Im using the fifo approach.. working great so
far!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
SavoySent: Friday, May 05, 2006 8:57 AMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Dumping queue_log to MySQL
Yes, check a post that I made about 4 months ago, I posted the cofig for
setting the speaker, handset and ring volumes ..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Jerry Jones
|Sent: Thursday, May 04, 2006 3:15 PM
|To: Asterisk Users Mailing
Do you know if you can disable soft keys like the blind xfer key that shows
on the screen?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Derek Listmail Acct
|Sent: Thursday, May 04, 2006 6:52 PM
|To: Asterisk Users Mailing List - Non-Commercial
Anyway to check out Corraleta? :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Dean Collins
|Sent: Wednesday, April 26, 2006 1:12 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] I am looking for a
Deadagi?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Tony Mountifield
|Sent: Tuesday, April 25, 2006 5:17 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Background asynchronous AGI
|
|I have been writing a lot of AGI
there recently.
|
|Regards
|
|On 4/22/06, Anton Krall [EMAIL PROTECTED] wrote:
| Are you sure its from today?
|
| The file has dates
|
| libmfcr2-0.0.3.tar.gz 30-Mar-2006 09:06 346K
|
| Also inside th tar the changelog has nothing inside and the
|news file
| has nothing too.
|
| How did you see
|consultory. If you still have problems let me know and i may
|be able to help you through SSH.
|
|Best Regards
|
|On 4/21/06, Anton Krall [EMAIL PROTECTED] wrote:
| Moises, how can I find out which version Im running, on
|Steves ftp all
| say
| 0.0.3 or the date also says the same date
Moises, how can I find out which version Im running, on Steves ftp all say
0.0.3 or the date also says the same date.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Moises Silva
|Sent: Friday, April 21, 2006 9:43 AM
|To: Asterisk Users Mailing
Can you send the output of zttest ? Whats your average and what kind of
hardware are you using?
That will give people pointers of what to use/expect.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Boris Bakchiev
|Sent: Thursday, April 20, 2006
a wide range of signal levels.
|
|Steve
|
|Anton Krall wrote:
|
|Do you know if you can tweak gains if using unicall? I tried it once
|and if you move the gains on zaptel using a te110p with
|unicall on E1,
|when gains are +2 or -1, calls do not complete, forget even about
|faxing
If the voice distortion sounds like clack clack clack las if you had a fan
right next to you (remember when you talk directly to a fan in front of you,
the other side gets your voice like in intervals), if thats the case,
exactly, your frame size should be 20ms, sipura and some other atas come by
Guys, this is a weird question but has anybody disabled the blind button
that appears on polycoms or know if you can disable the use of blind
transfers on polycoms to make any transfer attended?
Thx!
___
--Bandwidth and Colocation provided by
Do you know if you can tweak gains if using unicall? I tried it once and if
you move the gains on zaptel using a te110p with unicall on E1, when gains
are +2 or -1, calls do not complete, forget even about faxing :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
: Xeon or Opteron?
|
|I have used many sangoma cards, and have not had *any* irq issues
|
|Anton Krall wrote:
| Has anybody used the sangoma fxo cards with asterisk? Anybody using
| multiple cards? Problems with irq and such (same as with
|digium ones)?
|
|
|
| |-Original Message
: [Asterisk-Users] polycom blind transfer button
|
|I could be wrong but off the top of my head I think that it is
|in the features section of the config file.
|
|-Jonathan
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Anton Krall
|Sent: Tuesday
I don't know if this only works with multiple cpus but I have HT enabled and
it shows cpu0 and cpu1 .. I tried the first part of this email and still the
kernel boots and shows 2 cpus.. Will this only work with 2 real cpus?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
PM
|To: asterisk-users@lists.digium.com
|Subject: Re: [Asterisk-Users] te110p and interrupts
|
|On Monday 17 April 2006 12:39, Anton Krall wrote:
| I don't know if this only works with multiple cpus but I have HT
| enabled and it shows cpu0 and cpu1 .. I tried the first part of this
| email
really discriminate
|between their T1 and A200 cards.
|I found one defect in their FXS driver, which they have now fixed.
|
|Overall seems to be a good product, slightly more affordable
|and less of a problem child than the Digium/TigerJet TDM400
|
|John Novack
|
|Anton Krall wrote:
|
|Has anybody
I must agree with you. I too buy Digium cards because I want to support the
development of asterisk. Asterisk is a great product but digum cards are a
pain, they say they don't support faxing but a lot of people that are
implementing asterisk demand or need faxin as a day to day service on
their
Problem is, how to make sure you system WILL have 100% on zttest before
buying the cards.. You need to have stability, compatibility and certainty
that what you buy is going to work :(
Anybody had similar problems or success stories with sangoma cards?
|-Original Message-
|From: [EMAIL
Aaron, have you tried using 1 te110p and 2 tdm04b on the same server?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Aaron Daniel
|Sent: Thursday, April 13, 2006 7:19 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re:
disappointing !
|
|Anton Krall wrote:
| I must agree with you. I too buy Digium cards because I want to
| support the development of asterisk. Asterisk is a great product but
| digum cards are a pain, they say they don't support faxing but a lot
| of people that are implementing asterisk demand or need
).
|
|
|Anton Krall wrote:
| My main concerns would be, can you have multiple cards like
|this on a
| system, for example, I now have a te110p and 2 tdm04b and Im getting
| irqmisses on the te110p (according to zttool and zttest) which makes
| fax receiving on the te110p almost impossible.. Plus, voice
. The plug-n-play approach will have a
|very high failure rate.
|
|
|Anton Krall wrote:
| I must agree with you. I too buy Digium cards because I want to
| support the development of asterisk. Asterisk is a great product but
| digum cards are a pain, they say they don't support faxing but a lot
Of
|Rich Adamson
|Sent: Friday, April 14, 2006 8:37 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Digium cards, so disappointing !
|
|Anton Krall wrote:
| Problem is, how to make sure you system WILL have 100% on zttest
| before buying the cards.. You
Carlos, have you tested your te110p (or any T1/E1 card) to see if you are
missing irq, frame slips?
Ive tested a te110p with unicall (depends which version are you using) and I
am able to receive a few raxes after a very long time, Im getting frame
slips wich makes it very hard but Im trying to
Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple
cards? Problems with irq and such (same as with digium ones)?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|John Novack
|Sent: Wednesday, April 12, 2006 10:29 AM
|To:
Hi Andrew...
Thank you very much for the info.
I didn't recompile the kernel, Im using a generic 2.6 kernel but its worth
taking a look at what you said.. Where can I find (which file) the Hz the
kernel was precompiled to?
Also, Im running 1 te110p and 2 tdm cards, probably I'll disable 1 card
and interrupts
|
|On Mon, 2006-04-10 at 19:25 -0500, Eric ManxPower Wieling wrote:
| Anton Krall wrote:
| I will try that and see what happens...
| This server is a supermicro one.. Anybody else had issues
|like this
| on supermicro? Any hints on how to resolv them?
|
| If I remember
Guys. I have an issue with a te110p card and also some tdm04b cards on the
same system:
Zttest returns this for the tdm04b cards:
[EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 38 -v
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8191 sample intervals 99.987793%
8192
If you do php, try this:
?
$format = '%d/%m/%Y %H:%M:%S';
$strf = strftime($format,$argv[1]);
echo $strf\n;
?
Copy the code into program.php
So run it as: php -q progra.php 232313123.2
23232323 been the timestmap you want translated into real date
Hope this helps.
|-Original
PeruSent: Monday, April 10, 2006 5:47 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] te110p and interrupts
use lspci -vb for detecting interrupt conflicts..
On 4/10/06, Anton
Krall [EMAIL PROTECTED]
wrote:
Guys.
I have an issue
: [Asterisk-Users] te110p and interrupts
|
|Try booting with apic off, I think it's noapic kernel option.
|Notice this is APIC and not ACPI, which you referred to.
|Then get your boards on different REAL irqs.
|
|Moj
|
|Anton Krall wrote:
| This system has acpi enabled. This is how the cards looks like
This is a Xeon with HT... I remember I disabled HT on supermicros bios but
then again, maybe somebody enabled it again.. Thats why its showing 2 cpus?
Ill disable it too..
Thx for the hint.. BTW why did you commented the timer line?
|-Original Message-
|From: [EMAIL PROTECTED]
|
|Anton Krall wrote:
| I will try that and see what happens...
| This server is a supermicro one.. Anybody else had issues
|like this on
| supermicro? Any hints on how to resolv them?
|
| If I remember correctly, supermicro bios does let you assign irq to
| certain pci ports right? Will that help
] queueue recording and what to do next
|
|Anton Krall wrote:
|
|Guys, if you define recording on queues.conf and also define a
|monitor_filename var on your dialplna, you can record a queue
|call but,
|isthere a way to do something with the file after the call
|ends? I need
|to move the file
Im running hylafax, iaxmodem and spandsp with asterisk on unicall with mfcr2
e1 in Mexico and Im having a very hard time getiing faxes to work.. I hear
click on the lines and they are E1 so.. I have no clue whats happening but
could it be frame slips?
|-Original Message-
|From: [EMAIL
Guys, if you define recording on queues.conf and also define a
monitor_filename var on your dialplna, you can record a queue call but,
isthere a way to do something with the file after the call ends? I need to
move the file to some other place but I cant find where to define a command
to run after
It really makes that much diff. using slinear vs. ulaw?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Colin Anderson
|Sent: Tuesday, April 04, 2006 11:26 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE:
next
|
|On 14:36, Tue 04 Apr 06, Anton Krall wrote:
| Guys, if you define recording on queues.conf and also define a
| monitor_filename var on your dialplna, you can record a queue call
| but, isthere a way to do something with the file after the
|call ends?
| I need to move the file to some other
What do you mean by overlap dialing?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of C F
|Sent: Monday, April 03, 2006 3:26 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] polycom overlap dialing?
|
|Use
Guys.
Ive been plyaing around with applicationmap in features.conf
Im sing it to playback a sound to the caller but here is a problem, how does
the callee know when the sound has finished playing if he cant hear it at
all so I was wondering, has anybody played around with this?
Is there a
This may sound complicated but how about adding another
extenion in extensions.conf where you define meetme with the admin flags and
make another extension for normal users without the flags.
Plus you can use apps like authenticate or maybe even mysql
statements to auth the admin in the
, sometimes
|this means adding t or T to the dial command.
|Please note, there is no security whatsoever to disallow the other
|party from activating an application map.
|
|On 4/3/06, Anton Krall [EMAIL PROTECTED] wrote:
| Guys.
|
| Ive been plyaing around with applicationmap in features.conf
|
| Im
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