Questions:
[1] Can I use oslec for echo cancellation? I'll have beefy hardware.
Is echo cancellation necessary?
Yes you can use oslec provided that either your distribution has a zaptel
package with the oslec patch (or you build the zaptel drivers + oslec
yourself)
Well without echo cancelation y
hi, all
is there any way in queue app. to execute asterisk app. after Queue() app. i.e
[myplan]
exten => _X.,1,Answer
exten => _X.,n,Queue(myqueue)
exten => _X.,n,Background(file-to-play)
exten => 1,1,Playback(thnks)
exten => 2,2,Playback(by)
Is these possible above situation , how
thnks, B
Hello,
Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.
I'm setting up a new office and a home office and i'm shopping for hardware.
Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Link: http://www.voipsupply.com
Hi,
A makes call to B. B has connection problem with the server (say, the
lan cable is unplugged).
1: A ---> server
2: A <--- server
3: server > B
In 2, server will send the ring to A and it will hear ringing tone.
In 3, server will try to connect B until timeout.
My question is:
A will stil
Does anyone have a better ENUM lookup handler than the built-in
ENUMLOOKUP() function? The built-in function does not properly handle
multiple return values such as:
8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP"
"!^\\+1866(.*)$!sip:[EMAIL PROTECTED]" .
8.9.9.3.2.8.8.6.6.8.
Tomorrow night is the monthly Asterisk night...in melbourne
(australia)...
The usual stuff - get together, eat, show off tech toys.
At the Pint on Punt, from 7pm.
later,
PaulH
___
-- Bandwidth and Colocation Provided by http://ww
On Tue, May 6, 2008 at 11:42 AM, Anthony Francis <[EMAIL PROTECTED]> wrote:
>
>
>
> Tilghman Lesher wrote:
> > On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
> >
> >> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
> >>
> >>> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
> >>>
>
On Tue, May 6, 2008 at 5:19 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
>
> On Tue, May 06, 2008 at 03:33:31PM -0400, OCG Technical Support wrote:
> >A client has asked that our asterisk installation leverage their large
> >investment in their existing data center infrastructure. We'r
Hi to all,
I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls)
and 15 SIP extensions.
The receptionist has a SNOM-360.
How many SIP accounts would you configure on that phone?
Only one would be enough?
One SIP account, has a limit on concurrent calls?
I saw that the SNOM-360 can han
Are you saying the * server does NOT TRY to re-establish the BD connection ?
Does your whole * SERVER freeze ?
If NOT, what happens to you CDR records ?
Anthony Francis wrote:
> Tilghman Lesher wrote:
>
>> On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>>
>>
>>> 5 maj 2008 k
On Tue, May 06, 2008 at 03:33:31PM -0400, OCG Technical Support wrote:
>A client has asked that our asterisk installation leverage their large
>investment in their existing data center infrastructure. Weâre thinking
>about putting the voicemail messages onto a Samba share (on their f
Tilghman Lesher wrote:
> On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
>> Using 1.6-rc8.
>>
>> In iax.conf on the calling box, I have:
>>
>> [iax-out]
>> .
>> callerid = "sean" <447>
>>
>> I even also put the same on called box.
>>
>> But I can't seem to set the callerid:
>>
>> exten =>
On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
> We are actually running an AsteriskNow appliance with asterisk 1.4.18.1
> and it's quite unstable.
> We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy
> deadlock"
> and now that we have added a Queue
It would probably be wiser to run an IMAP server and do imap storage instead of
writing to a cifs-mounted directory... or use ODBC storage... assuming they are
running a database server somewhere.
I don't have any experience with having * write voicemail files to CIFS/SMBFS,
but I also think it
Tzafrir Cohen a écrit :
> On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
>
>
>> Here it is, but since the AsteriskNow release has stripped the binary
>> i fear it won't be of much use:
>>
>
> Is there any "-debug" package for asterisknow's asterisk package?
>
> On RedHat th
A client has asked that our asterisk installation leverage their large
investment in their existing data center infrastructure. We're thinking
about putting the voicemail messages onto a Samba share (on their file
servers). Any pros/cons to this? Does network/samba latency create
choppiness?
I setup two asterisk servers with identical settings
(same extensions, same queues, etc). Each one is
connected to the same amount of incoming/outgoing
links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box).
Most extensions are sip and they register via DNS SRV
and other methods so that the two serv
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
> Here it is, but since the AsteriskNow release has stripped the binary
> i fear it won't be of much use:
Is there any "-debug" package for asterisknow's asterisk package?
On RedHat they are generated automatically. On Debian they r
I had a similar problem when the calls were not recorded when there was a
transfer. Is that your case? If so, the solution is to start recording on the
inbound leg of the call for all channels. Just like that:
[default]
exten => _00[2-6]XXX,1,Dial(Local/[EMAIL PROTECTED])
[recording]
ex
On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
> Using 1.6-rc8.
>
> In iax.conf on the calling box, I have:
>
> [iax-out]
> .
> callerid = "sean" <447>
>
> I even also put the same on called box.
>
> But I can't seem to set the callerid:
>
> exten =>_NXX,1,Answer()
> exten =>_NXX,n,NoOp(
*1.4
Sorry for a dumb question, but I'm working with my SIP
provider on a problem and I can't answer this question
for them. They don't know Asterisk.
When I do a "sip show channels"
What is the "User/ANR" field?
Bill
___
-- Bandwidth and Colocati
That's fine... honestly I hate the message myself, however corporate policy is
corporate policy so there isn't much of a point in discussing it.
That being said, the message does clearly say that the message is for the named
recipients, in this particular case, the named recipient is a public ma
I had a similar problem. In my case Asterisk was crashing due to MixMonitor()
and then automatically restarting.
I have never found a alternative solution to record the calls.
Regards,
Sanjay Rajdev
- Original Message -
From: "Rahul Yadav" <[EMAIL PROTECTED]>
To: asterisk-users@li
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = "sean" <447>
I even also put the same on called box.
But I can't seem to set the callerid:
exten =>_NXX,1,Answer()
exten =>_NXX,n,NoOp(${CALLERID(num)})
Answer("IAX2/iax-in-7", "") in new stack
NoOp("IAX2
/etc/init.d/asterisk start
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
-- Bandwidth and Colocation Provided by http://www.a
tp://lists.digium.com/mailman/listinfo/asterisk-users
__ NOD32 3078 (20080506) Information __
This message was checked by NOD32 antivirus system.
http://www.eset.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.c
Hi All
I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not recording
complete recording.
Suppose i got a call connected and talking for 3 minutes then mixmonitor
records only 2 minutes of call.
This problem happens randomly.Please help as i am suffering very much due to
this problem
I'm not an expert, but FastAGI you use in a live channel over TCP connection
... Most people agree it is better because doesn't spawn a new process every
time it is called ... but they also suggest to do not use in the same server ...
DeadAGI you cannot use it in a live channel ...
By the way
Matt Watson schrieb:
> Disclaimer Statement: This e-mail is confidential and is intended for the
> above-named recipient(s) only. If you are not the intended recipient and/or
> have received this e-mail in error, please notify us by telephone and delete
> this e-mail from your system without re
Hi all,
Are there Asterisk user groups and organized Asterisk enthusiasts in
the following cities: Salt Lake City Utah, Chicago IL, Boston MA,
Tampa FL? Looking to meet up with folks in these cities during the
month of May when I am in those cities.
Drop me a mail if you are looking to shoot the b
I`m using collect call blocking with astunicall from moises and it`s working
properly.
UC_CATEGORY=INTERNATIONAL_DATA (for brazilian users) indicates a collect
call.
I spent a long time searching a way to do it, but it was only possible with
moises code.
Thank you.
Luis A P Barbosa.
2008/5/6
My bad, I also should of mentioned...
That was on Asterisk 1.4.18 and Zaptel 1.4.10
Using a TE220B
--
Matt
From: Matt Watson
Sent: Tuesday, May 06, 2008 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: PRI D-Channel reconfiguration = crash asterisk?
Hello,
I just
Hello,
I just had to have MTS Allstream fix a new T1 install that we have that we
aren't running in production yet, but it is attached to a production machine.
Apparently they setup the T1 with only a 1 B-channel (how useful!) even though
we had ordered it fully loaded with 23. Anyways... th
Steve Totaro a écrit :
> On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>
>> lordfuknowsyou a écrit :
>>
>>
>>> Vinícius Fontes wrote:
>>>
>> >
>> > I use 1.4.18 with no problems. We have quite a few users(125 total
>> > between branches), but the call v
Tilghman Lesher wrote:
> On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>
>> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>>
>>> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>>>
5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> On Monday 05 May 20
Tilghman Lesher a écrit :
> On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:
>
>> Tilghman Lesher a écrit :
>>
>>> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
>>>
lordfuknowsyou a écrit :
> Vinícius Fontes wrote:
>
> I use 1.4.18 with no
On May 6, 2008, at 10:20 AM, [EMAIL PROTECTED]
wrote:
I'm wondering what version of asterisk people use in production
environnement ?
on which distribution ?
And what is your setup like ?
We are actually running an AsteriskNow appliance with asterisk
1.4.18.1
and it's quite unstable.
I
On Tuesday 06 May 2008 09:02:47 Steve Totaro wrote:
> On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher
>
> <[EMAIL PROTECTED]> wrote:
> > On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
> > > While these may not be popular opinions, I still ask, what does
> > > SwitchVox use? What do some of t
On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:
> Tilghman Lesher a écrit :
> > On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
> >> lordfuknowsyou a écrit :
> >>> Vinícius Fontes wrote:
> >>>
> >>> I use 1.4.18 with no problems. We have quite a few users(125 total
> >>> between branche
These are the instructions that I followed. I did managed to get the
fast busy to go away, but the RDNIS simply does not seem to work. These
are the instructions that I followed on this project. I have run out of
time trying to get Call Manager 4.x to talk to Asterisk 1.4.
http://www.voip-info.or
Google is awesome
http://www.voip-info.org/wiki-Asterisk+AGI
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles
Sent: Tuesday, May 06, 2008 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Performance issues
Hel
I'm using 1.4.18 in production on 2 boxes... one of which being a custom built
desktop basically, the other being a Dell 1950 III
We are in a migration phase to the Dell box, right now the 1st box is doing
nothing more than being a PSTN gateway to some FXO lines... basically waiting
for numbers
Hello,
Our company did 200+ installations around the globe and had no issues with
stability with correct Asterisk version.
We used most of 1.4. As far as I remember 1.4.16 has some nasty bugs along
with 1.4.19.x (SIP + realtime).
So current stable is 1.4.18.1 (for us).
For load check: http://wi
Tilghman Lesher a écrit :
> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
>
>> lordfuknowsyou a écrit :
>>
>>> Vinícius Fontes wrote:
>>>
>>> I use 1.4.18 with no problems. We have quite a few users(125 total
>>> between branches), but the call volume at the most has been around 15
We are using Asterisk 1.4.13 in production, were we have almost 30 SIP users on
a Asterisk box, we are also using IAX to communicate between main Asterisk
server and the other. we use Queues, Conference too.
Regards,
Sanjay Rajdev
- Original Message -
From: "Benoit Plessis" <[EMAIL
Hello,
We are thinking in use asterisk-java to an billing solution, wich is the
better choice, and if someone could give us a understandable description
about the difference between DeadAGI and FastAGI, i found a very interesting
project called asterisk2billing and they use DeadAGI, anyway wich on
Steve Totaro a écrit :
> On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>
>> Any IAX2 phone or mostly SIP ?
>> Do you use Call Queues ?
>>
>> We have less user than that, less concurrent call but quite a few
>> crash/deadlock
>
> Try SIP only if you can and report
Steve Totaro wrote:
>> I use 1.4.18 with no problems. We have quite a few users(125 total
>> between branches), but the call volume at the most has been around 15
>> active calls at a time.
>>
>>
>>
>
> I would classify that as "Light to Medium Call Volume" or "SMB".
>
> Let me clarify what
Benoit Plessis wrote:
> lordfuknowsyou a écrit :
>
>> Vinícius Fontes wrote:
>>
>> I use 1.4.18 with no problems. We have quite a few users(125 total
>> between branches), but the call volume at the most has been around 15
>> active calls at a time.
>>
>>
> Any IAX2 phone or mostly
On Tue, May 06, 2008 at 10:01:54AM -0400, Jay R. Ashworth wrote:
> On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote:
> > I'm wondering what version of asterisk people use in production
> > environnement ? on which distribution ?
> >
> > And what is your setup like ?
>
> Well, we're
Tilghman Lesher wrote:
> On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
>
>> All I see in the ABE release
>> notes is 1.2 although I have heard that ABE should be running 1.4
>> "Very Soon" many many moons ago
>> http://www.digium.com/en/docs/ABE/README . So either Digium doesn't
>> trust
On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
> > While these may not be popular opinions, I still ask, what does
> > SwitchVox use? What do some of the guys around here that setup large
> > systems use? Is ABE e
On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote:
> I'm wondering what version of asterisk people use in production
> environnement ? on which distribution ?
>
> And what is your setup like ?
Well, we're running a cluster of about 15 boxes or so with Slack 10 or
12 and 1.2.17(?, ei
I just want to Run Asterisk with the basic required modules, What can I do to
achieve so?
My only requirement is to run SIP clients and the Dictate Module.
Regards,
Sanjay Rajdev
___
-- Bandwidth and Colocation Provided by http://www.api-digital.co
On Tue, 2008-05-06 at 08:42 -0500, Tilghman Lesher wrote:
>
> It's not actually a fix to the security fix.
No, indeed.
> The security fix simply
> highlighted an issue which was already present in Asterisk.
That may be true, but the security fix now depends on that new fix, so
it's tangentially
On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
> lordfuknowsyou a écrit :
>
> > Vinícius Fontes wrote:
> >
> > I use 1.4.18 with no problems. We have quite a few users(125 total
> > between branches), but the call volume at the most has been around 15
> > active calls
Hello everybody,
I've set up DUNDi between two asterisk boxes, and sometimes happens that calls
from machine A can't reach peers in machine B, but calls from B to A work
correctly.
The strange thing is that the CLI command 'dundi show peers' shows correctly
the registered peer in both servers, and
I still have not had time to dig and find what I have but there are
several "worksheets" ranging from sizing or initial customer
questionnaires. This will give you an idea of what kind of hardware
you will need to purchase to put together a (hardware) quote.
Another "worksheet" goes over features
Mostly SIP, some of my clients have queues and everything is working fine by
now.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- "Benoit Plessis" <[EMAIL PROTECTED]> escreveu:
> lordfuknowsyou a écrit :
> > Vinícius Fontes wrote:
> >
> > I use 1.4.18 with no
On Tuesday 06 May 2008 07:35:14 Brian J. Murrell wrote:
> On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote:
> > Yes.
>
> Hrm. For those of us that are following along the AST-* train, patching
> as per the AST-* release notices, as a matter of process, wouldn't it
> have been good to r
On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:
> While these may not be popular opinions, I still ask, what does
> SwitchVox use? What do some of the guys around here that setup large
> systems use? Is ABE even using 1.4 yet?
Yes, ABE version C (in release for several months) is using the 1
On Mon, May 5, 2008 at 9:29 PM, gmail <[EMAIL PROTECTED]> wrote:
>
>
> Hi all,
> I want to use a cell phone as my FXO line to Asterisk Box ,did anyone
> try this and configured it and how to physically connect it to Asterisk
> server?
Check out chan_mobile. Super cool.
Thanks,
Steve Totaro
Hi all,
I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try
this and configured it and how to physically connect it to Asterisk server?___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mai
On Tue, 2008-05-06 at 07:58 -0400, Steve Totaro wrote:
[snip]
> While these may not be popular opinions, I still ask, what does
> SwitchVox use?
Not sure what Asterisk version they use but I saw (iirc) a presentation
on their website that they run switchvox on top of Fedora Core 6. FC6
has been
On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
> lordfuknowsyou a écrit :
> > Vinícius Fontes wrote:
> >
> > I use 1.4.18 with no problems. We have quite a few users(125 total
> > between branches), but the call volume at the most has been around 15
> > active calls at a time.
>
> Any IAX2 p
On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
> > On Monday 05 May 2008 11:24, Johansson Olle E wrote:
> >> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> >>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
> Another issue that we nee
> I use 1.4.18 with no problems. We have quite a few users(125 total
> between branches), but the call volume at the most has been around 15
> active calls at a time.
>
>
I would classify that as "Light to Medium Call Volume" or "SMB".
Let me clarify what I consider "High Call Volume". ~400 s
lordfuknowsyou a écrit :
> Vinícius Fontes wrote:
>
> I use 1.4.18 with no problems. We have quite a few users(125 total
> between branches), but the call volume at the most has been around 15
> active calls at a time.
>
Any IAX2 phone or mostly SIP ?
Do you use Call Queues ?
We have less
Vinícius Fontes wrote:
> There were some really unstable Asterisk releases in the 1.4 branch. I
> personally use 1.4.13 or 1.4.15 in production. Every single time I tried
> 1.4.16 or higher I had problems.
>
>
>
> Att
> Vinícius Fontes
> Desenvolvimento
> Canall Tecnologia em Comunicações Ltda.
>
On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote:
> Yes.
Hrm. For those of us that are following along the AST-* train, patching
as per the AST-* release notices, as a matter of process, wouldn't it
have been good to republish AST-2008-006 and include this fix along with
the original
Yes.
Julian
Brian J. Murrell wrote:
> On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
>> That was a bug in the release.
>>
>> From the 1.4.20-rc1 Changelog:
>> 2008-04-30 16:30 + [r114891] Russell Bryant <[EMAIL PROTECTED]>
>
> So basically, r114891 was a fix to AST-2008-006? So if yo
On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
> That was a bug in the release.
>
> From the 1.4.20-rc1 Changelog:
> 2008-04-30 16:30 + [r114891] Russell Bryant <[EMAIL PROTECTED]>
So basically, r114891 was a fix to AST-2008-006? So if you applied the
patch for AST-2008-006 you now re
On 2008-05-06 at 03:46 Tzafrir Cohen wrote:
>On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez
>wrote:
>> Move to root:
>>
>> sudo -s
>>
>> type your passwd
>>
>> and as root:
>>
>>
>> Edit the file /etc/init.d/asterisk
>>
>> And uncommet the two lines than sasys so
There were some really unstable Asterisk releases in the 1.4 branch. I
personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16
or higher I had problems.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- "Steve Totaro" <[EMAIL PROTECTED]>
On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I'm wondering what version of asterisk people use in production
> environnement ?
> on which distribution ?
>
> And what is your setup like ?
>
> We are actually running an AsteriskNow appliance with asteris
On 5/6/08, Asterisk <[EMAIL PROTECTED]> wrote:
> Hi guys,
>
> I would like to ask you, if any of you has any experiences with the
> predictive dialers available for Asterisk? Are open source predictive dialers
> such as VICIDIAL Dialer any good?
>
> Which one would you recommend for a ca. 45 se
Hi,
I'm wondering what version of asterisk people use in production
environnement ?
on which distribution ?
And what is your setup like ?
We are actually running an AsteriskNow appliance with asterisk 1.4.18.1
and it's quite unstable.
We have ~30 IAX2 SoftPhones and encounter some "Avoiding I
Hi guys,
I would like to ask you, if any of you has any experiences with the predictive
dialers available for Asterisk? Are open source predictive dialers such as
VICIDIAL Dialer any good?
Which one would you recommend for a ca. 45 seat call center where most of the
agents work on both inbound
Hi!
Thank you for your answer.
However I would like to know, if there is any other possibility of
making this work, with older unicall versions.
The reason I ask, is because it's not easy for me to upgrade the
asterisk and unicall version on my machine, since it is a production
machine, and I
Christian wrote:
> Hi all,
> I have seen discussions on this earlier on, but just want to hear some quick
> thoughts.
> I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to
> make it run at boot. Since I've got a firewall and don't have any other
> servers running I am n
I totally agree. Someone filed a bugreport for this? Also asterisk
init script should be installed by default too.
I am going to give Cesar's instructions a try (sans removing /bin/sh)
and hope it works!
On Tue, May 6, 2008 at 3:24 AM, Stelios Koroneos
<[EMAIL PROTECTED]> wrote:
> In general, if
In general, if your asterisk is accesible from the internet its much better
to have it run as a non-root process.
(My opinion is that this should be the default out-of-the-makefile ;)
asterisk behaviour)
This is the "norm" for more of the servers/services running on a linux
system, and can act as a
5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
>>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
Another issue that we need to fix with the MYSQL driver is that
we're
lacking
83 matches
Mail list logo
