Quoting C F [EMAIL PROTECTED]:
Version? What is the CLI output? What phone are you using?
It appears that it is hanging up cleanly, and the reorder tone is from
the phone.
This is asterisk v1.4.13. The CLI say:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/2401-081e4440, Unconditional
Andrew Kohlsmith wrote:
On Tuesday 16 October 2007 15:25:13 Philipp Kempgen wrote:
Michael Collins wrote:
I don't know if it's relevant or not, but I do know that at least one
legacy PBX vendor (NEC) has a 'solution' that helps with some of the
sillier CDR's that could get generated. They
On Tue, 16 Oct 2007, Lees, James (UK) wrote:
I am slowly getting up to speed with asterisk. This is a very basic
problem but I would appreciate any help.
I am using a small network of clients and an asterisk server. Each
client has a headset to communicate. Is there a simple way of playing
On Tue, Oct 16, 2007 at 09:20:32PM -0500, Brian West wrote:
You'll need to compile with debug symbols and have ulimited -c
unlimited set. Then you can examine the core and find out what
exactly caused the crash... Segfaults either are easy to find or very
hard to find, depending on what
Well, most of the configuration (but the dialplan) can be kicked up pretty
fast by preparing a block (eg for a SIP extension) and then pasting it
over with minor modifications. I usually keep the original demo config
files around, that already include most options, and use them as a
Robert McNaught wrote:
Alan,
What do you mean by the udev rules?
I previously had asterisk compiled and running as user and group 'asterisk'
zaptel and libpri were compiled and installed using user 'root'
so the zaptel service was root. I had a dependency issue with asterisk
trying
I for one would like to thank you for looking at the CDR processing in
Asterisk. In the scheme of things I guess it's not the most compelling of tasks
but it's definitely somehting that needs some work.
At the moment I actually block SIP responses from my users with a status code
300 and
Giedrius Augys wrote:
Hi,
I have problems with asterisk and hylafax+ iaxmodem. I can successfully
send faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I
have problems: No carrier. This is hylafax log, maybe you can suggest
me where to find ...
Oct 17 07:38:48.22:
CountryCode:1
AreaCode: 800
FAXNumber: +3705203230
LongDistancePrefix: 1
InternationalPrefix:011
DialStringRules:etc/dialrules
ServerTracing: 0xFFF
SessionTracing: 0xFFF
RecvFileMode: 0600
LogFileMode:
On 10/17/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 16 October 2007 15:25:13 Philipp Kempgen wrote:
Michael Collins wrote:
I don't know if it's relevant or not, but I do know that at least one
legacy PBX vendor (NEC) has a 'solution' that helps with some of the
sillier
Hello everybody,
I know that I can define variables in the [global] context of extensions.conf.
How can I do the same thing in other conf files, like features.conf?
Thanks in advance,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95
On Wed, Oct 17, 2007 at 08:43:59AM +0100, Alan Lord wrote:
Robert McNaught wrote:
Alan,
What do you mean by the udev rules?
I previously had asterisk compiled and running as user and group 'asterisk'
zaptel and libpri were compiled and installed using user 'root'
so the
Witch interface are you using to send faxes, SIP, IAX, ZAP, MISDN,...,etc
VoipCrazy
2007/10/17, Giedrius Augys [EMAIL PROTECTED]:
CountryCode:1
AreaCode: 800
FAXNumber: +3705203230
LongDistancePrefix: 1
InternationalPrefix:011
SIP, but I think I know whereis a problem. Fax is connected to d-link, and
is turn on t.38
2007/10/17, voip crazy [EMAIL PROTECTED]:
Witch interface are you using to send faxes, SIP, IAX, ZAP, MISDN,...,etc
VoipCrazy
2007/10/17, Giedrius Augys [EMAIL PROTECTED]:
CountryCode:
Behalf Of Anselm Martin Hoffmeister wrote:
Sent: 16 octombrie 2007 09:29
Subject: Re: [asterisk-users] About .call files when the congestionis
on myside
*IF* an unanswered call stops the retry cycle then it's true, I can
simply
ask for lots of retries. I assumed an unanswered call
Hello everyone.
I recently boght a Nokia BH900 headset and USB bluetooth dongle and I'd
like to use them to make calls from a sofphone. I managed to this with
boxe XTen-Lite and the Zoiper - but they both see the device as a simple
sound card through the BlueSoleil drivers. While this is
So still without the .conf files, the GUI will not be
enough to do the work.
And the .conf files come with GUI are all the .conf
files that existed in the CLI commands or it is a
matter of some files that usually will be used?
Regards
Bilal
bilal ghayyad wrote:
Can I download and install
I'm very happy to announce the first stable version of my little
creation: Miruna Asterisk System.
This version run only on Soekris 4801 board, but I think if we change
kernel run on other boards:)
You can download it at:
http://sourceforge.net/projects/miruna-asterisk/
In next days I'll setup a
That was exactly it. Default 1.4 install include unavail.ulaw, which was
matching over all other recordings. When I deleted the useless files it went
fine.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch
Sent: Tuesday, October 16, 2007
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt
Thanks!
David
___
--Bandwidth and Colocation
Anyone know when Asterisk Business Edition 1.4 will be released? We are
looking to purchase, but with all the changes between 1.2 and 1.4 think it may
be best to wait if the new version is just around the corner.
Thanks,
John Beaman
Telecom Specialist II
Voice Telecommunications Services
I used DEC's EDT for almost 20 years on PDP-11 and find jed with the EDT
interface useful!
You can't teach an old dog new tricks!
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: 16 October 2007 17:10
To: 'Asterisk Users Mailing List
David Thomas wrote:
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say
fmt
Your wav files are broken.
(or maybe in a format not supported
Alan Lord wrote:
Hi,
Is it mandatory that all voicemail entries have a numerical mailbox
name? Or is it possible (like SIP extension for example) for them to
have alphanumeric names?
All references I have read use numerical definitions and examples, but
do not explicitly state this is
On Wednesday 17 October 2007 08:18:01 David Thomas wrote:
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not
say fmt
It means that your .wav files
Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
extension.conf. Every Dial() command in my diaplan has the appropriate
On Sunday 14 October 2007 15:02:43 Volker Sauer wrote:
Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
Dear all
I want to configure Huntgroup for my company like i call on 1100
extention i will transfer to avalible group extention i got some document on
voip-info website but this is not working for me
Giedrius Augys wrote:
I have problems with asterisk and hylafax+ iaxmodem. I can
successfully send faxes to Panasonic KX-FT932 fax, but with Xerox
WorkCentre M20i I have problems: No carrier. This is hylafax log,
maybe you can suggest me where to find ...
Oct 17 07:38:48.22: [22428]:
On 10/17/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I want to configure Huntgroup for my company like i call on
1100 extention i will transfer to avalible group extention i got some
document on voip-info website but this is not working for me
Hi,
Does anybody have some ideas - how to play a sound file on channel, after that
bridged channel got hanged up?
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
Thanks ... I forgot to say I tried it with
priorityjumping=yes
in the [globals] section of extensions.conf
still no go...
Gerald, I'll try your suggestion,
and try to figure out the result code tests :-)
Thanks,
Rich
-Original Message-
From: Gerald A [mailto:[EMAIL PROTECTED]
- Original Message -
From: Atis Lezdins [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 17, 2007 5:18 PM
Subject: [asterisk-users] Play sound on hangup
Hi,
Does anybody have some ideas - how to play a sound file on channel, after
that
bridged
Atis Lezdins wrote:
Does anybody have some ideas - how to play a sound file on channel, after
that
bridged channel got hanged up?
---cut---
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
---cut---
Maybe something like
In article [EMAIL PROTECTED],
Atis Lezdins [EMAIL PROTECTED] wrote:
Hi,
Does anybody have some ideas - how to play a sound file on channel, after
that
bridged channel got hanged up?
So A calls B, who answers; A talks to B and B then hangs up - you want a sound
file played to A?
Assuming
On Wed, 2007-10-17 at 02:56 +0200, Philipp Kempgen wrote:
btw:
Who's responsible if someone (A) calls you (B) via an operator
and you agree to pay reverse charges for the call? ;)
It was A who dialed but it is you who will be sent the bill.
Let's carry this a bit further:
After talking for a
Ok so you use templates. I understand that. The problem is some people on
here seem to be claiming they type it all in from scratch in like 3 minutes.
-Original Message-
From: Lenz [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 17, 2007 12:27 AM
To: Asterisk Users Mailing List -
On Wednesday 17 October 2007 19:03:55 Tony Mountifield wrote:
Does anybody have some ideas - how to play a sound file on channel, after
that bridged channel got hanged up?
So A calls B, who answers; A talks to B and B then hangs up - you want a
sound file played to A?
Assuming A is
On Wednesday 17 October 2007 18:54:58 Philipp Kempgen wrote:
Atis Lezdins wrote:
Does anybody have some ideas - how to play a sound file on channel, after
that bridged channel got hanged up?
---cut---
g- Proceed with dialplan execution at the current extension if the
On 10/17/07, shadowym [EMAIL PROTECTED] wrote:
Ok so you use templates. I understand that. The problem is some people on
here seem to be claiming they type it all in from scratch in like 3 minutes.
Just call me out if you feel the need to. Please don't try and hide
behind the some people on
Hello List,
For those of you using Cisco phones, did you have to purchase a 'SIP
license' for each phone?
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-723-1690
[EMAIL PROTECTED]
We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1,
Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual
Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two
TE120P cards and everything was working fine. Since they needed to add
a third E1 line we
On 13:01, Wed 17 Oct 07, Anciso, Roy wrote:
Hello List,
For those of you using Cisco phones, did you have to purchase a 'SIP
license' for each phone?
Thanks
Yes.
Or use chan_skinny.so
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
Carlos Chavez wrote:
crashes about three times a week. I can see the Asterisk process go
from around 10% usage to 100%, then 200% and the server hangs after
that.
Why would inserting a multiport card affect Asterisk and the server?
How can I debug this situation? I do not have enough
On Wednesday 17 October 2007 19:09:23 Carlos Chavez wrote:
Why would inserting a multiport card affect Asterisk and the
server? How can I debug this situation? I do not have enough slots to
insert three single cards of the same type so I need the multiport card to
work. When Asterisk
I don't see whats wrong with that. If show channels doesn't show any
active channel then you are ok. What phone are you using? What exactly
is the problem?
On 10/17/07, Turbo Fredriksson [EMAIL PROTECTED] wrote:
Quoting C F [EMAIL PROTECTED]:
Version? What is the CLI output? What phone are
At 11:08 AM 10/17/2007, you wrote:
It's developed a nasty habit. At random times, it likes to dial my cell
phone voicemail number and play my messages to anybody who happens to be
within earshot.
Any clues where to look at what's going on? My voice mail number
(extension 220 in my dialplan) is
On Wed, 2007-10-17 at 21:03 +0300, Atis Lezdins wrote:
On Wednesday 17 October 2007 19:09:23 Carlos Chavez wrote:
Why would inserting a multiport card affect Asterisk and the
server? How can I debug this situation? I do not have enough slots to
insert three single cards of the same
I have a Sipura SPA-841.
It's developed a nasty habit. At random times, it likes to dial my cell
phone voicemail number and play my messages to anybody who happens to be
within earshot.
Any clues where to look at what's going on? My voice mail number
(extension 220 in my dialplan) is the only
I know I can get free DID's with SIP, is anyone giving out free DID's with
IAX?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline
Carlos Chavez wrote:
We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1,
Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual
Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two
TE120P cards and everything was working fine. Since they needed to
Steve Edwards wrote:
I have a Sipura SPA-841.
It's developed a nasty habit. At random times, it likes to dial my cell
phone voicemail number and play my messages to anybody who happens to be
within earshot.
Any clues where to look at what's going on? My voice mail number
(extension
Quoting Dave Fullerton [EMAIL PROTECTED]:
its a form of revenge for all those times you shuffled on the carpet
and zapped it.
Steve Edwards wrote:
I have a Sipura SPA-841.
It's developed a nasty habit. At random times, it likes to dial my cell
phone voicemail number and play my
On Wednesday 17 October 2007 13:22:00 Carlos Chavez wrote:
The server is running CentOS 5 with all upgrades applied. I have
glibc-2.5-12 installed. If I could only test the individual cards for
two days before having to install the Sangoma A104x again and in those
two days I did not
Does anyone know how I can send only calls that have been forwarded or
redirected out on a specific channel?
My PRI provider (BellSouth/ATT) does not let me set the outgoing caller
ID to anything other than our DID numbers, so when a user FWD's his
Polycom phone to his mobile, it looks like
He's worried that the Hangup application returns non-zero.
C F wrote:
I don't see whats wrong with that. If show channels doesn't show any
active channel then you are ok. What phone are you using? What exactly
is the problem?
On 10/17/07, Turbo Fredriksson [EMAIL PROTECTED] wrote:
==
Hi,
Am writing scripts to manage configuration management and Asterisk.
I would like to be able to point the asterisk binary at config
directory with an asterisk.conf in it, and for asterisk to run a pre-
flight check. A bit like a pint check in php, 'apachectl configtest'
and lots of
john beaman wrote:
Anyone know when Asterisk Business Edition 1.4 will be released? We are
looking to purchase, but with all the changes between 1.2 and 1.4 think it
may be best to wait if the new version is just around the corner.
I can confirm that is just around the corner. However,
On Wednesday 17 October 2007 22:23:23 Andy Davidson wrote:
Hi,
Am writing scripts to manage configuration management and Asterisk.
I would like to be able to point the asterisk binary at config
directory with an asterisk.conf in it, and for asterisk to run a pre-
flight check. A bit like a
On Wed, Oct 17, 2007 at 08:23:23PM +0100, Andy Davidson wrote:
Hi,
Am writing scripts to manage configuration management and Asterisk.
I would like to be able to point the asterisk binary at config
directory with an asterisk.conf in it, and for asterisk to run a pre-
flight check. A
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really frustrating and
On Wednesday 17 October 2007 22:57:41 Michael Iedema wrote:
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error,
Michael Iedema wrote:
Greetings everyone,
today I spent the last part of my day trying to find a parse error
You have:
GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
Mine uses Double Quotes as such:
GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
Doug
--
Ben Franklin quote:
Those
Am Mittwoch, den 17.10.2007, 21:57 +0200 schrieb Michael Iedema:
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse
Atis,
Try removing spaces around =
Thanks for the tip! I'd tried that before but I tried again with
another fail. I've posted all of the relevant bits from my dialplan as
well as the log output.
http://pastebin.ca/740270
It seems now that everything is working except the macro
email-hungup
At 12:57 PM 10/17/2007, you wrote:
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really frustrating and is probably a stupid mistake.
exten = h,n,GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
Should
Michael Iedema wrote:
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight.
On Mi, 17 Okt 2007, Atis Lezdins [EMAIL PROTECTED] wrote:
If it's a global variable, and it works one way, but not another - then you
should post a bug.
Yes, maybe it's a bug. But I'm not sure. I still think I'm missing a
detail
You can also try using in your call file:
Set:
On 10/17/07, Ira [EMAIL PROTECTED] wrote:
At 12:57 PM 10/17/2007, you wrote:
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really frustrating and is probably a stupid mistake.
exten =
Hi,
I want asterisk to call a person on the phone for monitoring the
refrigerator storing vaccines.
I am clueless where to look. Can someone clue me in ?
Thanks,
balu raman
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On Wednesday 17 October 2007 23:41:28 Michael Iedema wrote:
On 10/17/07, Ira [EMAIL PROTECTED] wrote:
At 12:57 PM 10/17/2007, you wrote:
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really
On Wednesday 17 October 2007 14:57:41 Michael Iedema wrote:
exten = h,1,Set(MISSEDCAUSE=hungup)
exten = h,n,GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
exten = h,n,Hangup()
exten = notify,1,Macro(email-${MISSEDCAUSE})
exten = notify,n,Return()
exten = h,1,Set(MISSEDCAUSE=hungup)
exten =
Anything to do about portscans? Is there any way (should I) to see
if the connection is a legit (only SIP currently) connection BEFORE
my * answers?
[2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts:
Bad request protocol 01@ASTERISK_IP SIP/2.0
-- Executing
UDP being stateless and connection-unaware as it is (unless you're using
TCP transport), there's not really a meaningful sense in which your
Asterisk answers as there is no initial dialogue or handshake. It
simply replies to messages on an atomic basis.
Thus, it is even more difficult than
Quoting Balu Raman [EMAIL PROTECTED]:
Hi,
I want asterisk to call a person on the phone for monitoring the
refrigerator storing vaccines.
I am clueless where to look. Can someone clue me in ?
Thanks,
balu raman
I used this way back for something similar but forget the details
there is a
The refrigerators will have external outputs to trip relays (even if
your customer doesn't know that), ask for the number of their
refrigeration mechanic he will tell you how to get electrical/relay
outputs for the alarms.
These are then connected to asterisk via an interface board so when
Il Neofita wrote:
Hi,
I update from asterisk 1.2 to 1.4 and I have some problems.
In the extensions I used DIAL(SIP/100SIP/101,30,tTr) if I receive a
call from an external providers
now in 1.4 I recieve only one ring
What can I do to solve this problem?
You can start by removing the Fake
Balu,
Do you want events passed to Asterisk from the refrigerator? Or does a
reminder type phone call need to be placed on an interval? Please be
more specific, since this sounds like a special purpose refrigerator,
does it have any way of passing events to an external device?
Omar A. Sabek
On
Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]:
He's worried that the Hangup application returns non-zero.
And that the phone (Polycom SoundPoint IP430 SIP) 'indicates' an
error even though there wasn't.
It does this because of the forced Hangup() as I see/understand it.
My phone isn't registering with my Asterisk appliance,
but I'm not sure where to find any logs to see what is
going on? Does the appliance not support log viewing?
Thanks,
Scott
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Tilghman,
exten = h,1,Set(MISSEDCAUSE=hungup)
exten = h,n,GosubIf($[${SENDNOTIFICATIONS} = 1]?notify)
exten = h,n,Hangup()
exten = h,n(notify),Macro(email-${MISSEDCAUSE})
exten = h,n,Return()
Don't leave the h extension; it's special. :-)
Fantastic! That solved it. I also needed one more
Omar,
I am hoping that there may be some temp sensor interface that can be
routed to a pc and if the temp falls out of a range, I can have this
event call someone. I know what to do in asterisk to make a call. I
have to do some research. may be, someone has already done a similar
thing. Has to be
Omar A. Sabek wrote:
since this sounds like a special purpose refrigerator,
does it have any way of passing events to an external device?
SNMP? Syslog? ;)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and
Greetings to all list members!
My name is Aldo Sudak and I am an Asterisk newbie. I am writing now because I
have not
been able to find any mention to the issues described below, neither in this
list nor in the wiki.
I am performing preliminary tests with Asterisk 1.4.11 and Zaptel 1.4.5.1 on
Ok Thanks,
I guess I'll have to give it a shot. I just assumed it would be more work
than 30minutes (after the initial learning curve) for a moderately complex
dialplan..
-Original Message-
From: Erik Anderson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 17, 2007 9:49 AM
To:
Don't laugh, some of the commercial units come with Ethernet built in and
external query capabilities for interfacing into external applications.
Hotel automation has come a long way since a clipboard on a string at the front
door.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
We use similar things here for issues like our generator battery voltage
monitoring. We just have a relay going into our alarm system and as
asterisk monitors our alarms it initiates emails or calls out. The alarm
system is also linked into a seperate SMS unit for emergency backup so
we also get
I have read all the wiki's and blogs and how to links about Bluetooth but so
far no luck. I can confirm that CentOS5 sees my Bluetooth adapter and my
cell phone. No Joy on Asterisk 1.4. The information out there is kind of
confusing as there is a lot of outdated info sometimes referring to
That a refrigerator is getting power is not the same as it operating
nominally.
Doors get left open... compressors fail... refrigerant eventually leaks
out of seals and coils...
Best to query it for temperature... and at a point faraway from the
coils, such as the top of the door... which
On Wed, Oct 17, 2007 at 06:37:21PM -0300, Aldo D. Sudak wrote:
Greetings to all list members!
My name is Aldo Sudak and I am an Asterisk newbie. I am writing now because I
have not
been able to find any mention to the issues described below, neither in this
list nor in the wiki.
I am
Size/Speed/write cycles have gone way up, price has gone way down. More
common than CompactFlash and no need for an adapter. So is it feasible to
run an Asterisk server on something like this? With a MTBF of 1million
write cycles coupled with dynamic wear management on a 4Gig USB drive,
On 10/17/07, Mike Hammett [EMAIL PROTECTED] wrote:
I wasn't looking to involve Asterisk until I got it working solidly without
it. ;-)
I have yet to find on the entire internet a working example of T.38
pass-thru. I need to file a bugreport of T.38 being broken :(
view http.conf
bindaddr=0.0.0.0
On 10/10/07, Sanjoy Rath [EMAIL PROTECTED] wrote:
Hello,
When I click on User menu, I get loading screen status. It runs
indefinitely without showing me
the user list and the user admin menu.
Any thoughts ?
Thanks,
Sanjoy.
Russell:
Do you think there might be a working t.38 implementation in ABE 1.4?
Even passthru Between endpoints and gateways with Asterisk in the
middle?
Best regards,
Andrew
On 10/17/07, Russell Bryant [EMAIL PROTECTED] wrote:
john beaman wrote:
Anyone know when Asterisk Business Edition
Hi list,
Last Friday, an Asterisk server became unresponsive after ~8,5 months of
smooth operation (~32 calls). Server did reply to pings, but no ssh,
no more console login. Also Asterisk no longer took calls, but ISDNguard
watchdog was still alive. Looking at the logs after reboot, I
On Wed, 2007-10-17 at 14:53 -0700, shadowym wrote:
Ok Thanks,
I guess I'll have to give it a shot. I just assumed it would be more work
than 30minutes (after the initial learning curve) for a moderately complex
dialplan..
The other issue that arrives is that a complex dialplan can't be
On 10/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On 10/17/07, Mike Hammett [EMAIL PROTECTED] wrote:
I wasn't looking to involve Asterisk until I got it working solidly
without
it. ;-)
I have yet to find on the entire internet a working example of T.38
pass-thru. I need to
I am new in asterisk world can u shortly explian how to create queue and how to
work this ?
David Gomillion [EMAIL PROTECTED] wrote:
On 10/17/07, satish patel [EMAIL PROTECTED] wrote: Dear all
I want to configure Huntgroup for my company like i call on 1100
extention i will
shadowym wrote:
Size/Speed/write cycles have gone way up, price has gone way down. More
common than CompactFlash and no need for an adapter. So is it feasible to
run an Asterisk server on something like this? With a MTBF of 1million
write cycles coupled with dynamic wear management on a
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