On Mon, Jan 02, 2006 at 11:25:02AM +0800, Craig Guy wrote:
Are you using raid for performance or redundancy? Software raid is
nice except when the drive that fails is the one with your boot
partition on it. I guess you could always tftp boot the kernel or
something.
On our raid1 machines
Just Test
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On Saturday 31 December 2005 01:57, Ross C wrote:
... and 2 Snom 320's (now discontinued I think).
No, they are not discontinued !!!
Regards,
Sven
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Hello Friends,
I was trying to dial agents from a normal extension. My extensions.conf is
configured as
exten = 11,1,AgentCallbackLogin
exten = 12,1,Dial(Agent/12) ;; configured in agents.conf as agent =
12,12, vivek
exten = 13,1,Dial(SIP/13) ,, is configured in sip.conf
I am not sure the test modules work
yet.
PaulH
- Original Message -
From:
RdBSD
To: asterisk-users@lists.digium.com
Sent: Monday, January 02, 2006 7:41
PM
Subject: [Asterisk-Users] test
Just Test
It's the welltech wellgate 3804 4FXO gateway.
More info:
http://www.welltech.com/product_e_03.htm
I have another model 3702 (2FXO+2FXS).
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta
Sent: Monday, January 02, 2006 2:09 PM
To:
Hi,I'm trying to set up asterisk 1,2 with Festival and everything works fine until I install additional languages. When I dial appropriate extension I get something like this:Jan 2 10:43:06 WARNING[836]: app_festival.c:484 festival_exec: Festival returned LP : cstr_pl_em_diphone
Does anyone know
http://www.alibaba.com/catalog/10886425/Fxs_fxo_Port_Converter.html
I have one and bad voice quality.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, January 02, 2006 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial
Friends in the Asterisk community!
HAPPY NEW YEAR!
(You have to emulate Allison saying that yourself, or try to copy me
saying it with my Swedish accent!)
2005 was a great year for Asterisk. After more than a year's work, we
released Asterisk 1.2 with lots of new functionality. We had two
On 1/1/06, Nir Simionovich [EMAIL PROTECTED] wrote:
Well, the documentation states that Video Conferencing is possible. I'vetried working with EyeBeam, which yielded nice Results, but anything beyondthat - I can't comment.Nir Scan you share your experience with us
i.e. what asterisk version what
hi all,
i am trying to execute a call file in asterisk by placing it in the outgoing directory.In my system i m running rtpproxy and openser also.Asterisk is communicating with openser because i am able to make incoming calls to asterisk.But when i try to put call file in the outgoing
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is common.
We have users, what like to have more then one Phone, - Homeoffice and Desk
Rigth now if a other phone registers whith the data,
Hello,
as You are running two processes handling SIP (asterisk and openser), I
think the Call-File addresses the wrong instance.
If Your callfile contains a line like
Channel: SIP/accountname
try something like
Channel: SIP/[EMAIL PROTECTED]:port
where ipaddress and port
I'd be very interested in hearing more about this
as I am in need of a similar installation. Anyone
have a hint?
Kristian
On Thu, Dec 29, 2005 at 02:14:29PM -0500, Asterisk wrote:
I am working an a multiple box asterisk solution. I need phones to be able
to login to multiple asterisk
I have an Avaya IP Office PBX connected to an
Asterisk system via a Sangoma ISDN PRI card.
Dialing from the as
terisk system into the avaya works just fine but
when trying to call from a phone connected to the
avaya syste
m something goes wrong. After punching the first
four digits the Avaya calls
Hello,
first of all, I say Happy New Year to this list!
While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
chan_capi 0.4.0-PRE1), I ran into the following problem.
I want to signal busy to an incoming call, but that doesn't work.
The dialplan looks like this:
exten =
Hi list and happy New Year.
I working on an application based on Asterisk Manager
and I have to recover caller number and called number.
Are there manager functions able to do that?
If no function already exists, does someone know an
issue to resolve my problem?
Thanks
Amaury
Thanks for the new firmware, finally some of the features are becoming
available that make the phone more usable with Asterisk.
One question though, ringer tone #2 on the Snom 360 firmware has been
replaced?
How can I get the old ringtone back? I was using the ringtone on phones in
locations
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
Hello,
first of all, I say Happy New Year to this list!
While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
chan_capi 0.4.0-PRE1), I ran into the following problem.
I want to signal busy to an incoming call, but that doesn't
On Mon, 2 Jan 2006, Kristian Larsson wrote:
I have an Avaya IP Office PBX connected to an
Asterisk system via a Sangoma ISDN PRI card.
Dialing from the as
terisk system into the avaya works just fine but
when trying to call from a phone connected to the
avaya syste
m something goes
On Mon, Jan 02, 2006 at 03:36:57PM +0200, [EMAIL PROTECTED] wrote:
On Mon, 2 Jan 2006, Kristian Larsson wrote:
I have an Avaya IP Office PBX connected to an
Asterisk system via a Sangoma ISDN PRI card.
Dialing from the as
terisk system into the avaya works just fine but
when
Original Message
From: Andreas Koch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Monday, January 02, 2006 1:03 PM
Subject: [Asterisk-Users] connect more the one phone to ONE sip Acoount
Hello,
how is it possible to
Hello Armin,
On Mo, 02.01.2006 Armin Schindler wrote:
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
Hello,
first of all, I say Happy New Year to this list!
While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
chan_capi 0.4.0-PRE1), I ran into the following problem.
Original Message
From: Ross C [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 7:18
AM Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away
Thanks, but I'm looking for information on
I don't think this is the same problem I am experiencing. As you can see below
the two BT's are almost identical and I have others the same too. so the fault
is fairly consistent, unfortunately I have been unable to determine the exact
reason for it yet. It is not the whole box crashing it is
Mike Fedyk ha scritto:
Hiu Yen Onn wrote:
How big of RAM for Asterisk server? My production environment will be
about 400 users in the office.
In one server? 4GB. And more if you can.
I'd suggest you use several servers for 400 users unless the
percentage of active phones is ~10%.
i have the same problem. but when i remove all hints from my dialplan
in extensions.conf.
on more crash will occur.
Paradise Dove
On 1/2/06, Andrew Gough [EMAIL PROTECTED] wrote:
I don't think this is the same problem I am experiencing. As you can see
below the two BT's are almost identical
An interesting wrinkle I'm running against is that you cannot port
numbers from a cellular carrier to a landline. i.e. I can't port my
cell # to a DID on my PRI. I am not sure if this is just a line of
bullshit fed to me from Bell Mobility (Canadian CDMA carrier) but
I've not had the time to
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
Hello Armin,
On Mo, 02.01.2006 Armin Schindler wrote:
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
Hello,
first of all, I say Happy New Year to this list!
While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
Craig Guy a écrit :
Are you using raid for performance or redundancy? Software raid is
nice except when the drive that fails is the one with your boot
partition on it. I guess you could always tftp boot the kernel or
something.
If you're using GRUB, fallback option allow you to boot on
Greetings,
Here's my issue. My local free VSP isn't transfering proper DTMF
(inband or converting to RFC2833) so I'm stuck with making a php
interface so my roommates whom are not using softphone/ata devices to
call out via * (and thusly get the better deals in Long Distance).
I've tried using
Hi,
I am running the latest version of asterisk. (1.2.1) When the
machine is first started, asterisk runs great and there is no echo.
As asterisk is running for a few days (2 weeks or so) echo begins to
become more and more noticeable.
The echo is noticeable to both parties (both sides of
Hi,
I would like to know if asterisk is able to translate between two
differnet codecs. For example:
I have this config in sip.conf file:
[phone]
disallow=all
allow=ulaw
dtmfmode=rfc2833
dtmf=rfc2833
username=phone
type=friend
host=dynamic
secret=
mailbox=3001
context = sip
Yes, we do development under Linux for this. Was there some particular
support you were after?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
Your problem is related to not reading the documentation in voip-info.org :)
You can originate a call to anyplace doing:
- First a LoginAction.
- Then an Originate action with the proper arguments.
In the example you put, you are doing neither of them. You can test manually
how the protocol
Hi!
In sip.conf you can use
setvar=hair=brown
How do you do this when using sip realtime? I have seached the wiki and the mailinglist without results.-- Morten Isaksenhttp://www.misak.dk/blog/
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be sure you allow the g729 codec in [general] context in sip.conf for
the sjphone.
On 1/2/06, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote:
Hi,
I would like to know if asterisk is able to translate between two
differnet codecs. For example:
I have this config in sip.conf file:
Looks like I did not check what codes are supported on SJLABS.
It does not support g726.Thats why it is not working.
I checked it with gsm and is working.
My fault.
Sorry
Hi,
I would like to know if asterisk is able to translate between two
differnet codecs. For example:
I have this config
Hi,
Can anyone give me their thoughts and experiences with companies like
DASH-911, etc that will let me forward 911 calls OVER THE PSTN and
have them routed via the company? The company should also be able to
update the PS/ALI records.
I've heard good stuff about DASH-911, but called them today
For this and another issues we reboot our phone servers every week, Saturday
02:00 am. You can do it with croon.weekly. That stopped all the issues
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, January 02, 2006 9:47
From what I can see
The 2 legs of the call are: 'phone' in alaw and 'laptop' in g726, why
should he need G.729 anywhere ?
Bartosz, not exactly that familiar, but I guess you could try to debug
the call establishmment.
(one thing that puzzles me, you mention IAXy, but you show 2 sip.conf
Also, make sure the Asterisk application is running. The span will be clean
without it, but the application itself generates the d-channel messages.
-Original Message-
From: David Yat Sin [mailto:[EMAIL PROTECTED]
Sent: Friday, December 30, 2005 8:39 AM
To: 'Asterisk Users Mailing
On Mon, 2006-01-02 at 10:35 -0500, Matt wrote:
Hi,
Can anyone give me their thoughts and experiences with companies like
DASH-911, etc that will let me forward 911 calls OVER THE PSTN and
have them routed via the company? The company should also be able to
update the PS/ALI records.
I've
They probably have today off for the new year. We do.
On 1/2/06, Matt [EMAIL PROTECTED] wrote:
Hi,
Can anyone give me their thoughts and experiences with companies like
DASH-911, etc that will let me forward 911 calls OVER THE PSTN and
have them routed via the company? The company should
I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk.
Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk.
On 1/1/06, Brian
It is a national holiday, So many companies and Gooverment services are
on a Sunday schedule. Many phone companies BellSouth included (local
ILEC) will not make changes on a day like today.
If you have a real emergency hang-up and call 911, WAIT you did that
already :-)
-Original
HI all,
I am wondering if asterisk supports USB phones.
Thanks.
David
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On Mon, 2006-01-02 at 10:55 -0500, Tom Vile wrote:
They probably have today off for the new year. We do.
But you have voicemail dont you? I agree with the original poster that
no voicemail or anything else that way makes it somewhat concerning ...
I however will give new companies a little
Hi all,
anyone known if is there any SIP client to install on
an I-Mate SP5m with Windows Mobile ?
Thanks
Giordano
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Hi Remco,
Old Ringer 2 is not there on the phone anymore, perhaps you can use another
ring melody or a suitable custom melody:
The wav file itself should be a PCM encoded 8 KHz file at 16bit mono.
The time for loading the file should not be longer then 3 seconds ! And the
size should be below
Hi,
not good if you have a 911 emergency
with your routing, and can't get ahold of your 911 routing company!
I saw somewhere on the site that the 24*7 call center will not open
until sometime in Jan.
--
Leonard Burton, N9URK
[EMAIL PROTECTED]
You must understand, if you are on a plane you
Hello Giordano,
You may try SJPhone on your device, i'm using it with my Qtek 9090 and
it's pretty good working for more than 6 month now.
You can found it here : http://www.sjlabs.com/sjp.html
Cem
Giordano Grandis a crit:
Hi all,
anyone known
if is there any SIP client to
hi all, I have implemented VoIP gateway using X100P card. i have downloaded 3 X-Lite phones on 3 different PCs. Because of X100P, i can make call to analog phone also. Now i want to simulate my VoIP gateway. how much bandwidth it can consume, jitter, delay in network etc can
Sorry!!
Just discontinued @ voipsupply.com I guess.
Thx for the correction.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
(support)
Sent: Monday, January 02, 2006 2:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Tnx Cem,
but I need installation for
Windows Mobile 5.0, not on PPC. On PPC I have just tryied it and properly
worked.
Thanks
Giordano
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: lunedì 2 gennaio 2006
17.52
A: Asterisk Users
No, because USB phones are on the client side. What you
need to find is a softphone for your operating system that supports a USB
handset, there are a number of them available.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP
NewbieSent: Monday,
Ugo Bellavance wrote:
Kerry Garrison wrote:
You REALLY don't want to have call waiting on a line going into any
PBX. You
are only asking for problems. My basic home setup is an SPA-3000 but the
PSTN line only has call forward on busy, when busy, the number is
forwarded
to a DID at iax.cc.
Morel Mosolff wrote:
Dear friends and business associates,
I will be out of office until January the 12th, 2006.
With kind regards,
Morel Mosolff
H1 more of these and I will start a loop on a spare high bandwidth
server :)
--
Cheers,
Matt Riddell
This doesn't seem to be correct, too...
Sven
On Monday 02 January 2006 17:43, Ross C wrote:
Sorry!!
Just discontinued @ voipsupply.com I guess.
Thx for the correction.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
(support)
Sent:
We do this with AMP all the time:
http://voipspeak.net/index.php?/content/view/49/28/
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP
NewbieSent: Monday, January 02, 2006 9:03 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
Yes. I use X-Ten (now CounterPath) X-Pro for Windows Mobile devices. It
costs about $30 and works relatively well on my Windows Mobile PDA. Note that
you won't be able to readily use bluetooth headsets etc. but it works well
enough using the internal speaker/mic on the device.
That said,
Hi Usman,
Thanks for the explanation.
Could you make the old Ringer 2 available in some form, preferable
already in the format the phone understands?
That would solve the problem too :)
Thanks!!
Remco
On Mon, 2 Jan 2006, Usman Tahir wrote:
Hi Remco,
Old Ringer 2 is not there on the
http://www.voipsupply.com/product_info.php?cPath=95_114products_id=883
am I misreading something?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
(support)
Sent: Monday, January 02, 2006 11:11 AM
To: Asterisk Users Mailing List -
On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648
The seller refuses to tell me who the vendor is. Anyone know?
It is a Wellgate 3804 unit from Welltech. http://www.welltech.com/product_e_03.htm
--
Carlos
VoIP Newbie wrote:
HI all,
I am wondering if asterisk supports USB phones.
USB Phones require a softphone to work with usually. Even if it is an
invisible one.
So if you have a softphone with the USB phone, then yes.
--
Cheers,
Matt Riddell
___
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: Monday, January 02, 2006 7:26 AM
Your problem is related to not reading the documentation in
voip-info.org :)
Umm.. Yeah I have. Otherwise I wouldn't be a pain in the ass right now.
I'd just be clueless. :)
You
Michael Collins wrote:
-- Executing Dial(SIP/Mikey-3b78, Zap/g2/5551212) in new stack
Dec 30 13:09:03 NOTICE[5657]: app_dial.c:1010 dial_exec_full: Unable
to create c
hannel of type 'Zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
There
Andreas Koch wrote:
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is common.
We have users, what like to have more then one Phone, - Homeoffice and
Desk
Rigth now if a other phone
C F wrote:
What is the ASTUM module? If I may ask.
ASTUM - The Asterisk User Management module.
Cheers,
Madhawa
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The SPA3000 also answers the phone as soon as it goes off hook (you
can even hear it dialing).
On 1/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Perhaps a Sipura-3000 could be of use here? Anyone have any ideas about
that?
Greg
-Original Message-
From: [EMAIL PROTECTED]
C F wrote:
What is the ASTUM module? If I may ask.
If you read my New Year's mail earlier today, you will get more
information :-)
/O
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I believe that the Mediatrix 1204 also bridges the call as soon as it
is done dialing (you hear ringing from the POTS not from the SIP
channel).
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Well, it would have to be 4 of them for each of the available PSTN lines. I
have also considered a
Yes thanks to the Happy New Asterisk Year post I figured that out.
Thank You
On 1/2/06, Dualcall.com [EMAIL PROTECTED] wrote:
C F wrote:
What is the ASTUM module? If I may ask.
ASTUM - The Asterisk User Management module.
Cheers,
Madhawa
You are not gonna be able to modify this behaviour from the asterisk since in your case asterisk is only receiving the digits from someone else (an Avaya in your case but could be PSTN for instance)Just asked an Avaya support guy and told me you should take a look at the ARS Digit Analysis Table,
On 15:10, Mon 02 Jan 06, [EMAIL PROTECTED] wrote:
Hello Friends,
I was trying to dial agents from a normal extension. My extensions.conf is
configured as
exten = 11,1,AgentCallbackLogin
exten = 12,1,Dial(Agent/12) ;; configured in agents.conf as agent =
12,12, vivek
exten =
Hi everyone My problem is the following: I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk. My soft phone registers without problems
Um, I'm not too sure. :-) I guess just some relevant information.
Something pretty to show clients. I'll ask my SysAdmin if he has any ideas
of what he'd like to see.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
My understanding is that there is currently a shortage of phones at voipsupply
(and also in other places). The 320 is selling pretty good :-) and we are
making the biggest production run *ever* this month!
snom does not discontinue the 320!
Christian
-Original Message-
From: [EMAIL
Hi Al-
It appears that my config files are being overwritten on restart and/or
reboot wit my [EMAIL PROTECTED] distribution. Anyone familiar with this
behavior? I'd like to be able to set some of the reboot defaults, but can't
quite figure out what is going on here.
Any help would be
Yeah I'm well familiar with Linux and know how to cron a reboot.. but
WHY?! Even if reboot is what I need to do... does anyone have an
explination for what is happening?
I had read somewhere (but now can't find) that instead of a reboot I
can just unload the zap module (after stopping asterisk)
you need to add your custom mods to the _custom.conf files and then
they wont get over written
On 1/2/06, Schochet, Wes [EMAIL PROTECTED] wrote:
Hi Al-
It appears that my config files are being overwritten on restart and/or
reboot wit my [EMAIL PROTECTED] distribution. Anyone familiar with
What configs are being overwritten?
Also, you will get much more [EMAIL PROTECTED] support from the forums at
sourceforge
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Schochet, Wes
Sent: Monday, January 02, 2006 12:24 PM
To: 'Asterisk
I think if you use config files like _custom.conf, [EMAIL PROTECTED] won't
overwrite
them. For example, put your custom configs for sip.conf in sip_custom.conf,
not sip_additional.conf.
Someone wanna back me up on this?? I'm not doing so hot today (see 'GXP-2000
any good with *?' thread) ;)
I am getting these errors at times with my Asterisk 1.2 server. The
call will end and not make it through but if you call back its fine.
Its happening a few times a day.
Jan 2 12:55:24 DEBUG[2751] chan_sip.c: Ignoring too old SIP packet
packet 103 (expecting = 104)
Jan 2 12:55:24 DEBUG[2751]
Oooh!
So it's on backorder @ voipsupply...
If that's the case,
To the voipsupply folks:
The red text on the VoipSupply site is worded to kind of imply that the 320
isn't available anymore. It should mention something about order it now,
and we'll notify you when we have them or
hi list
need your opinion. thanks in advance
when calls come in from zap channel (te110p as E1/PRI) and go out to a SIP
peer, no ringtone heard at zap channel.
sip.conf:
[from_sipproxy]
progressinband=no
type=peer
context=from-proxy
host=xxx.yyy.zzz.www
port=5060
disallow=all
allow=alaw
OK - makes sense. Thanks. My problem is that I need to remove some defaults
- specifically the 82xx meetme extensions which conflict with my existing
dial plan. Any ideas on this.
-Original Message-
From: Tom Vile [mailto:[EMAIL PROTECTED]
Sent: Monday, January 02, 2006 2:26 PM
To:
When I dial a festival extension (1222), all I hear is a series of fast
clicks and then it hangups. I do not have a sound card installed but I
would think I don't need one. Is a sound card necessary? Should I use
a script instead of the scheme code? Can someone who has this working
send me
Administrator TOOTAI wrote:
Craig Guy a écrit :
Are you using raid for performance or redundancy? Software raid is
nice except when the drive that fails is the one with your boot
partition on it. I guess you could always tftp boot the kernel or
something.
If you're using GRUB, fallback
On Mon, Jan 02, 2006 at 01:01:44PM -0800, Mike Fedyk wrote:
Administrator TOOTAI wrote:
Craig Guy a écrit :
Are you using raid for performance or redundancy? Software raid is nice
except when the drive that fails is the one with your boot partition on it.
I
guess you could always
Simone Cittadini wrote:
Mike Fedyk ha scritto:
Hiu Yen Onn wrote:
How big of RAM for Asterisk server? My production environment will
be about 400 users in the office.
In one server? 4GB. And more if you can.
I'd suggest you use several servers for 400 users unless the
percentage of
Something is using up way too much memory, are you sure asterisk is
using 800mb of ram ? it should be ten times less.
Zoa
Mike Fedyk wrote:
Simone Cittadini wrote:
Mike Fedyk ha scritto:
Hiu Yen Onn wrote:
How big of RAM for Asterisk server? My production environment will
be about 400
Louis-David Mitterrand wrote:
On Mon, Jan 02, 2006 at 01:01:44PM -0800, Mike Fedyk wrote:
Administrator TOOTAI wrote:
Craig Guy a écrit :
Are you using raid for performance or redundancy? Software raid is nice
except when the drive that fails is the one with your boot
Hi!
I figured it out myself. Just add a setvar column in the sip table.
The variables is stored in this format: variable1=value1;variable2=value2;...
I have also updated the wiki.
On 1/2/06, Morten Isaksen [EMAIL PROTECTED] wrote:
Hi!
In sip.conf you can use
setvar=hair=brown
How do you do
Wes, you need to save your modifications in the custom files so they
don't get over written, that's what they are there for.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Schochet,
Wes
Sent: Monday, 2 January 2006 3:24 PM
To: 'Asterisk
You need to modify the conference section of the amp config so it
doesn't allocate the conference rooms to these extensions.
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Schochet,
Wes
Sent: Monday, 2 January 2006 3:38 PM
To: 'Asterisk Users
Hello,
I have some Pb with my asterisk .
The box runs out of memory
We have 1 Gb of Memory , try on different box, we different version of
asterisk ( 1.0.7 = 1.2.1) unable to stabilise memory.
the mmlogs :
1136224256 - WARNING: Freeing unused memory at 0x679600, in dial_exec_full of
That mem line didn't show the amount of memory cached, which is
subtracted from used memory to get an approximate count of application
memory.
It is more complicated to get an exact count since part of that cached
memory is mmapped executable code and data. Not to mention shared
memory
umm - you usually grab it from the cdr...and it
works very nicely if you are pushing your cdr into mysql.
PaulH
- Original Message -
From:
amaury BOSSE
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 03, 2006 12:13
AM
Subject: [Asterisk-Users] Is it
Hello all,
After upgrading to asterisk 1.2 or the current CVS HEAD, asterisk doesnt
receive all of the DID number that is calling in. I have a T100P and a freshly
installed version of Fedora.
zapata.conf and zaptel are default except for enableing em wink.
Any ideas? If I install
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