Hi,
Asterisk 1.4
Working (jitter buffers created as expected):
ZAP - SIP
SIP - ZAP
Not working (no jitter buffers created):
SIP - chan_local (with /nj) - ZAP
SIP - chan_local (with /j) - ZAP
SIP - chan_local (with no flags) - ZAP
I have this in zapata.conf:
jbenable=yes
jbforce=no
jbimpl=fixed
That's from asterisk-addons, you can ignore that error.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mark morreny
Sent: Tuesday, March 25, 2008 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Have problem
Asterisk: 1.4.17 with sip realtime
Openser 1.3.x
Hi,
I had this setup working fine until I try putting OpenSER in the picture as
a proxy.
Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip
users get send to them etc. Now with a proxy in the picture asterisk asks
the
Hi,
I currently have these two lines in my dialplan to extract different parts
out of a variable and I'd like to do it in one line instead. Does anyone
know how to use regular expression subexpressions in the dialplan?
Outputting a comma separated list that can be sent to ARRAY() would be nice
Hi,
Asterisk 1.4.17
Sangoma a102DE
I'm having some issues sending CallerID Name to a Dialogic based phone app.
According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see
that it is sending the CallerID Name Mike - Budgetone - reachme.com to the
Dialogic card, but it isn't
Agreed, Callweaver and Freeswitch are both better for conferencing
(especially if you don't have zaptel hardware).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 12, 2008 1:28 PM
To: Asterisk Users Mailing List -
I believe that is/was one of the goals of the phonecall project.
-Original Message-
Does it implement the ability to run more than 1 PBX in asterisk ? (Virtual
PBX)
To be clear:
more then 1 company using the same physical asterisk
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You'll need to post more info. Version and a scenario of what was happening
at the time would be a good start...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango
Sent: Wednesday, March 12, 2008 6:32 PM
To: Asterisk Users Mailing List -
Rich Adamson wrote:
I've tested a large number of other external adapters and have not
found a single one that had a reasonable echo canceller built in. Many
of them work fine on short pstn lines (where echo is much less of a
problem), but provided even reasonable service on longer pstn lines
When will Digium include the octasic on the TDM2400P? And maybe the
TDM400P?
Also how does the TE415P and TE420P differ from the TE412P card?
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other. Also each review should have a date so the reader can see
how fresh the data is to current.
http://www.voip-info.org/wiki/view/VOIP+Phones+Reviews
An example would be:
June 28th, 2006
Mike Fedyk
I have used these phones and I rank them in this order:
Linksys 941
Polycom 301
Sipura 841
Is anyone else getting messages from the lists.digium.com mail server
with errors about a mail loop?
I've been getting this for the last few weeks, but I don't have any list
software on my server. Any ideas?
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Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
English, please, folks.
I don't know Portuguese
Kristian Kielhofner wrote:
Mike Fedyk wrote:
I happen to have asterisk running as a router, so I use it doing QoS
with tc (traffic control) and wondershaper set to prioritize based on
port ranges. I sent a patch to the debian bug tracking system a
while back with a few improvements -- I
arp in the shell
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any network tools.
Thanks-John
Kevin P. Fleming wrote:
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to offer any suggestions.
The only requirement
this does not make any sense.
How do you dial to a service provider from your * box? Does it use PPP
and IP? And then you connect to another * box that is on a cable
connection that receives the call over IP and then dials out to a voip
provider? How do any fxo devices come into this
I'm in southern California, are you close or can you ship?
Bob Knight wrote:
I have 4 sparc based sun boxes I am about to pay money so I can
get rid of them. They are running older versions of Solaris.
You should be able to load Solaris 10 and play around with *
on them.
Time to clean the
Hi,
I've just been going through the various modules that are autoloaded to
see what I need and what I don't and came across chan_phone.so which
loads /etc/asterisk/phone.conf. I did a lookup on voip-info and google
and came across this article in Linux Journal from 2001.
Anyone know why
Michael Graves wrote:
I have the IP600 and like it a lot. However, I really LOVE the Aastra
480i CT. It supports more lines than the ip600, has a backlit LCD, and
the cordless handset is GREAT!
How is the range, and in what environment did you test?
Can you a call on the cordless and the
Warren wrote:
So the next question becomes... Is hardware EC necessary or can *
handle it in software? I am looking at some pretty beefy hardware for
my platform, a Dell PE2850 with dual Xeon 3Ghz processors and plenty
of RAM to spare.
Can your processors handle the load, yes. Do you want
Andrew Kohlsmith wrote:
Again, good to know. Thank you for your detailed post! The XML config for
these phones gives them a leg up over the ip501 as well, that is for sure.
I believe the IP501 phones do have a XML config file. At least the
IP301 does.
I take it that you mean the XML is
Andrew Kohlsmith wrote:
On Friday 16 June 2006 14:50, Mike Fedyk wrote:
I take it that you mean the XML is better on the 480i CT. If so, can
you be more specific?
No, I mean the XML config file for controlling the screen on the Aastra 480i.
There is no such thing on the ip301/501
Steve Davies wrote:
On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Steve Davies wrote:
We have even experienced problems within Europe where providers route
national calls via international routes to save money. This adds
significant latency and makes any echo so heavily delayed
Steve Davies wrote:
On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
There are 3 PRI's connected to the card each from different operators.
Especially echo occured on span 3 is really annoying. Configuration
files
are as follows. Is there something wrong in conf ?
Have you verified
Matthias Fechner wrote:
Hi Gareth,
Gareth Blades wrote:
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done
Eric ManxPower Wieling wrote:
Warren wrote:
I was just told that for my forthcoming system I will be getting a data
T-1 instead of a voice T-1. Given that all of the handsets will be voip
phones, no analog at all, do I need echo cancellation? I looked at the
voip-info wiki and it seems to me
Tzafrir Cohen wrote:
On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote:
FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly.
Just sitting there doing nothing on my test system it is using 170MB.
How exactly do you meassure memory usage?
E.g: on my laptop:
Comfort noise is the sound you hear from the phone to assure the user
that there is still a connection to the other end. It is there to keep
you from hearing no sound through the speaker and thinking you have been
disconnected.
Check your phone's config for comfort noise or silence
Try reducing the gain on the microphone. These phones pick up room
sounds *very* well.
Andres wrote:
Has anybody else experienced bad echo issues with this SPA941 phone
when calling SIP-SIP to another SPA ATA? When I call remote office
phones that are attached to SPA ATAs, I get very
Andres wrote:
Mike Fedyk wrote:
Try reducing the gain on the microphone. These phones pick up room
sounds *very* well.
WellI'm not using the speakerphone. Plus there is no gain setting
at all that I am aware off. Just Handset Volume or Speaker Volume.
I'm not talking about
Get some hardware, a TDM410b is only $125. Or upgrade to 2.6.13 or
later. Don't compile the kernel unless you know what you are doing.
You might try, Ubuntu 6.06, FC4 with updates or FC5 to see if that makes
a difference.
Also there are patches on mantis for delays in meetme conferences
Asterisk guy wrote:
are there any open source sip softphone (Window OS version )?
http://www.voip-info.org/wiki-Open+Source+VOIP+Software
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Tzafrir Cohen wrote:
On Wed, Jun 14, 2006 at 11:51:06AM -0400, Mike wrote:
Hi,
I'm stuck writing a Web GUI because nothing out there is exactly what I
need. I'm not writing something as extensive as what _is_ out there, but
just something that allows users to change where their calls are
Erick Perez wrote:
I just don't want to install it and then after a 5th user going to
call someone the asterisk begin to crash due to lack of resuources.
Check the wiki for SIP load generation tools you can use to test your
setup on any number of calls you like.
Martin Joseph wrote:
Ultimately you need to set up a server that does what you need and see
how it performs. Usually hardware overkill is a good bet, but you
don't need to go crazy.
So, one cpu per call is too much?
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Erick Perez wrote:
I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.
24 SIP users and one E1 card in an Intel 945board (533 Front side bus)
with 1GB DDR 533mhz of ram, one Pentium Dual Core
Steve Glaus wrote:
Mike Hammett wrote:
(ICMP) pings were under 1 ms. No amount of different Asterisk
versions or phone firmware revisions seems to solve this. All was
well, then (as far as we know) without changes, it crapped out.
Any ideas?
I'm having much the same issues only I'm
Carl Youngblood wrote:
Our asterisk system gains access to the PSTN through a voip provider.
We have no digium or other telephony hardware in our system. Do the
zttest results still matter to us? Our results were as follows:
--- Results after 1007 passes ---
Best: 100.00 -- Worst:
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
http://www.voip-info.org/wiki-Open+Source+VOIP+Software
Erick Perez wrote:
I appreciate all your help and posting.
I will then load (with test calls) using SIPP and astertest
will post back the result of this machine in question
any
to it?
On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote:
IAX trunking and meetme conferences are some of the heaviest users of
zaptel timing. I'd suggest if you don't have hardware timing (or at
least a 2.6.13 based kernel), then use SIP all the way or at least turn
off IAX trunking
Tom wrote:
Most of the latest generation POE switches including the Linksys
SRW224P provide their power on the data pairs, not the unused pairs.
So if both the data and the power are on the same pairs, how do you
make a cable adapter to work with the 7960G?
Maybe bridge the unused pairs with
First, remove telnet from your vocabulary. It should only be used over
serial connections these days. All other times, you should be using ssh.
Second, do you want the computer to be installed and running without any
major software changes for a year or more? Then use Centos or Ubuntu
Patrick wrote:
On Tue, 2006-06-13 at 23:47 +0800, Dinesh Nair wrote:
On 06/13/06 22:49 Colin Anderson said the following:
Although this may have changed in the newer 1.2.X series of Asterisk, I
believe that Asterisk does not support SMP from the perspective of
isnt asterisk
Steve Davies wrote:
On 6/12/06, Doug Crompton [EMAIL PROTECTED] wrote:
It seems that any firmware is usable on any hardware as my hardware is
2.x. I wonder if 3102 firmware could be used on the 3000. Is the size
the
same? I guess you would have to be willing to make a brick to find out!
I
shadowym wrote:
Any other recommendations/links for increasing the reliability of Asterisk
servers?
Separate the various use cases of the filesystem into different volumes
with LVM. The parts that are not written to except during upgrades like
/usr should be mounted read-only, and the various
Chris Mason (Lists) wrote:
Cory Andrews wrote:
IP430, will sit between the IP301 and IP501, IP430 will have dual
Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239
street price should fall likely between IP301 and IP501.
That looks great, the 301 is almost useless due
Peter Doyle wrote:
I figured asterisk was looking for SIP user 06, so I added it, but I
still got 404's. Turns out I just needed an EXTENSION, 06. I can now
make calls and receive them, too. Of course, if you have multiple
incoming lines, you'd need extension 06, 07, 08 ... etc, since each
Steve Davies wrote:
On 6/9/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
Consider getting a Sangoma A200D
(http://www.sangoma.com/datasheets/p_a200-specs) with the optional
hardware echo canceller module. It just works for echo cancellation;
no tweaks required. It takes a while to
Kevin P. Fleming wrote:
- Matt Riddell (IT) [EMAIL PROTECTED] wrote:
What does the onboard DSP do when used with Asterisk? Did Digium or
someone put code inside Asterisk to hand off the
processing/transcoding
to a Sangoma card?
According the Sangoma data sheet, the Octasic part
Kevin P. Fleming wrote:
- Mike Fedyk [EMAIL PROTECTED] wrote:
Will it have a 1024 tap echo can on all 96 channels? What about 8 T1
support like sangoma?
Those are completely unrelated questions; there is no need for an 8-span echo
can module when there is no 8-span T1 card
http://www.asterisk.org/download
http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS
amna saleem wrote:
Hi All,
I need a suggestion.
I want to run only IAX on two windows based PCs and asterisk
Can you suggest which asterisk , libpri and zaptel versions should i use?
do i need
I have a client who has about six of these phones. Luckily (for me, not
for them) they were purchased before I came into the picture.
Daniel Salama wrote:
I have heard complaints from my client about the speakerphone and they
are now
You don't notice any problems when using the speaker-phone,
and the GXP-2000. He hasn't
complained about the problems I mentioned on the GXP-2000 - yet :)
Thanks,
Daniel
On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:
Do you have multiple phones going down at the same time? If so,
monitor them with qualify=500 in sip.conf to see if they hit that
limit
party well or the remote party cannot hear them well.
Sometimes it works and sometimes the volume is very low and that's why
they cannot hear.
- Daniel
On Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:
What specifically were the voice quality complaints about the spa-841
phones? The only thing I
First of all, I'm not knocking Sipura/Linksys. I have heard very good
things about their products.
I'm just wondering if they are the only quality shop on the market. I
know about the zoom 5801 where you can't dial out the FXO from SIP, only
from the FXS port. And I have heard similar
John Novack wrote:
Is the 94x any better? seems without backlighting, any are next to
useless.
Yes, I like the 941 better than the Polycom 301 and the display is much
improved (no backlight, but one of the guys at voipsupply told me that
the 942 has a backlight which sounds very promising).
Kerry Garrison wrote:
I would never ever ever sell a client on a SPA-841 or heaven forbid the
GXP-2000. All the clients who bought those originally sold them off and went
for better phones very quickly.
Let me say that when suggesting the spa-841 it is only in the context of
sub-$100 phones.
behind a DSL circuit. I don't know if it's
because their DSL line is going up/down. They don't necessarily
claim their Internet goes down, however, they are not constantly
check it.
What would you (or anyone else) suggest?
Thanks,
Daniel
On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:
Do you have
Kevin P. Fleming wrote:
- Jon Lewis [EMAIL PROTECTED] wrote:
IMO, locking the licensing to a piece of system thats often built-in,
has
been very annoying. I think I'd be happier if it was locked to some
sort
of dongle (parallel, or more likely today, USB). At least that way,
we
In other words, please post your message to asterisk-biz instead.
Martin Joseph wrote:
What part of NON-COMMERCIAL do you not understand?
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Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound
call. Before integrating Asterisk, all calls route through their
current non-VoIP based phone provider. After integrating 1 trunk
from a VoIP
How do you setup asterisk so that the assistant sees the lights but
doesn't hear the rings?
picciuX wrote:
damon, i think many guys here missed your point and went away from it.
What you want to do is possible: i managed to do that using a GXP-2000
with beta firmare and asterisk 1.2.0.
GXP
Damon Estep wrote:
I understand your technology agnostic position, and it makes sense,
however my vote (for the little that it is worth) would be to implement
a SIP rfc complaint shared line appearance capability (and/or bridged
line appearance), and then, if possible, extend it to support
There are too many changes happening in trunk to constantly update
-addons to work with it. Once things settle down a bit, they will bring
-addons up to date.
This has been repeated a few times in asterisk-dev recently. Did you
google for asterisk trunk addons compile?
Damon Estep wrote:
Rich Adamson wrote:
Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding
from scratch. I installed FreePBX (CentOs) from scratch and asterisk
was running, but had not yet been configured. It too crashed with a
kernel panic. Ran memtest for 24 hours; no errors or issues
Juan Salas wrote:
Hello.
Has anybody knows how run two asterisk process
in one hardware? (each one with its own configuration?)
What end outcome do you want? Maybe there is another way to do it...
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I have a client with an installation with 3 TDM400P cards. 6 FXO, 6FXS
ports.
I followed the txgain/rxgain instructions and now have no echo
problems. The only problem I have now is the flaky network the SIP
phones are accessing asterisk with. (you should see the wiring there, ugh).
It's
No, you replied to a message from Vladimir Montealegre with the subject
Re: [Asterisk-Users] RJ21-RJ11.
That is called thread hijacking.
You may sort your mail by date, but others use a feature called
threading. It keeps track of who replied to what message to be able to
see a conversation
Matt wrote:
On 1/12/06, Tomislav Parcina [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
First,
Something seems to be wrong with the list. I'm not the only person
who has expressed seeing their messages either arrive late, or not at
all.
I'm
Connection pooling doesn't require threading.
You can also use a pool of processes which are quite cheap on Linux.
Douglas Garstang wrote:
Do you have a link to where it says this? The DBI docs that I looked at
(perldoc dbi) said that it isn't thread-safe.
-Original Message-
From:
Simone Cittadini wrote:
Douglas Garstang ha scritto:
So I really wish there was some way to measure how well the worst
case scenario would perform. This would be 120 simultaneous calls
(don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850
with 2GB RAM. Asterisk would call an
Andreas Sikkema wrote:
Is it possible to have nested MySQL queries in extensions.conf?
Ie, perform a query, grab a value, and then jump to another
location in the dialplan and do another query based on that
original value. I'm having problems with the result and
fetchid's and I'm not sure
Chris Albertson wrote:
Under Linux (and other OSes) It's not as bad as that. Even with
128 Perl processes running there is only one copy of the Perl
interpeter in memory. Each of the 128 running processes would
have it's own copy of only it's data segments. With Perl
already in memory the
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Peter,
Too slow! We're going to potentially be doing several MySQL lookups for routing
even the
most basic of calls, and if every one of those queries has to make a call out
to an AGI
script,
all calls. The script would have to
perform multiple database queries in order to route a call.
-Original Message-
From: Mike Fedyk [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject
. Asterisk just
opens a connection to a TCP port instead of executing a binary.
-Original Message-
From: Mike Fedyk [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Nested MySQL
Forwarded to OCLUG, LUGIE UUASC which have members that have expressed
interest in asterisk.
Mike
Kerry Garrison wrote:
The SoCal Asterisk Users Group will be meeting at the Heritage Park Public
Library on the corner of Walnut and Yale in Irvine on the 3rd Thursday every
month. The
It might be easier to dig instead.
[EMAIL PROTECTED] wrote:
I would love to be there, but it's just too far to drive.
regards,
PaulH
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com; asterisk-biz@lists.digium.com
Sent: Wednesday, January 11, 2006
Jean-Michel Hiver wrote:
Alexander Lopez a écrit :
TDMoE is stable and stale, There is no more development planed or
needed as it only opens up a pipe between two ethernet points using
Layer 2.
OK... What would be in the advantage in running TDMoE rather than
using IAX or SIP?
TDMoE
Tzafrir Cohen wrote:
On Sun, Jan 08, 2006 at 06:16:16PM -0800, Mike Fedyk wrote:
Tzafrir Cohen wrote:
Experimental: Asterisk 1.2:
At the moment they are not that experimental anymore and should be ready
for use, but are not well-tested yet.
To use it, define both sources:
deb
Small, medium and large are relative. What do you want it to do, and
why do you want to change your phone system? With the right talent,
(consultant or in-house) Asterisk can be used in most situations. With
that no more details, then a simple answer will have to suffice.
Most likely yes.
From: http://www.voip-info.org/wiki-Asterisk+Hardware
Dialogic
D/41JCT-LS
Note: The D/41JCT-LS is a full duplex card and is the first of the D/41
family that will work with Asterisk. Older D/41 cards like D/41(E)PCI
are half duplex cards designed for IVR type applications so they won't
Tzafrir Cohen wrote:
Experimental: Asterisk 1.2:
At the moment they are not that experimental anymore and should be ready
for use, but are not well-tested yet.
To use it, define both sources:
deb http://rapid.dotsrc.org/ experimental/
How does this compare with Asterisk 1.2.1.dfsg-1
Eric ManxPower Wieling wrote:
JCC wrote:
I don't get it. What is the advantage of using a GSM gateway? VOIP
calls are
pretty inexpensive as they are now. Is the use of a gateway intended
as a
backup incase a wired network connection goes down? I have being looking
around the net for
Mike Hammett wrote:
I've been Googling around for some time now (a few hours on dial-up).
I find all kinds of questions similar to mine, but either there is no
answer or the answer has nothing to do with the question. Hopefully
this post isn't another one of those.
Does Asterisk favor FPU
Is there anything like FastCGI for Asterisk so that AGIs can have
persistent processes?
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David Sampson wrote:
Hello –
I have a non-PRI T1
[...]
How do I take incoming calls on these same channels?
You should get a PRI T1.
The minute you get close to capacity on this line you will run into
timing issues with incoming and outgoing lines competing with each
other. This
Douglas Garstang wrote:
I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses.
Kevin P. Fleming wrote:
If the two servers service distinctly separate groups of endpoints,
they can share the same table since they won't care about the other
server's entries. If the two servers service the same endpoints but in
an active/passive arrangement, that would also work.
Can the
Kevin P. Fleming wrote:
Mike Fedyk wrote:
Can the various *RT servers be configured to use different tables so
there won't be any conflicts even if there is any client overlap
between the servers?
Yes, but I'm not sure how you'd manage failover in that situation then.
I was thinking
Brett, Gary wrote:
From what ive read on this list and the wiki, centos 4.x has issues with the
TE110P card ( a lot of people having issues after first reboot).Would 3.5 be
better (I know [EMAIL PROTECTED] uses this)
Am I right in saying that OS's with the 1.6 kernel still require a lot
Matt Riddell wrote:
Alistair Cunningham wrote:
We've been asked to quote for a large cluster running Asterisk and our
ITSP in a box product. The system will be SIP throughout, with mixed
codecs.
We're considering using Dell blade servers, 1855 or similar, on the
grounds that we normally
If you are not big enough to have your own domain, then you don't need
a disclaimer.
This e-mail transmission may contain information that is non-proprietary,
unprivileged and/or non-confidential and is intended exclusively to provide a clue
to Jason D. Wolfe. Any use, copying, retention or
Michael Graves wrote:
On Wed, 04 Jan 2006 19:04:18 +0100, Matt Riddell wrote:
I don't use m0n0wall, but wouldn't it be better just to shape based on a Type
Of Service and then set the TOS flags in iax.conf and sip.conf accordingly?
--
Cheers,
Matt Riddell
In a more general sense
Kevin P. Fleming wrote:
Rich Adamson wrote:
If you take the word dynamic out of that, then can he effectively
have primary/secondary/backup systems that allows the user to
re-register and/or redial his call on a different * server?
I don't understand the question.
I don't know if it was
Kevin P. Fleming wrote:
Mike Fedyk wrote:
Think of this scenario: You have two * RT servers running heartbeat
and one goes down. If the SIP registration information was kept in
the DB tables, the backup server could take over the ethernet and IP
addresses and continue without forcing
Brett, Gary wrote:
My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2
Use Debian or Centos (Free RHEL).
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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
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
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