insisted it went
into the server room
ASAP.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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a map that finds the nearest phone to a given MAC.
Use that to update the realtime database.
Sounds like a fun project, shame I don't need it !
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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rate a clock that is independent of your CPU
clock.
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() or something more deterministic like a counter)
Assuming the head box takes all the calls you could just use setgroup
and getgroupcount on the pri box and use them to count the calls.
Using groups has the advantage of dealing with hangup right.
The only tricky bit would be implementing min(group) i
heard rumors that some networks (e.g. 3g mobile carriers) put a
QoS on VPN packets to improve
the response time for their business customers. So in some
exceptional cases the IAX over VPN could
show lower latency than straight IAX, but it is pretty unlikely.
Tim Panton
www.mexua
te options visit:
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do with a streaming
solution,
which might matter with something as quick as a basketball game.
Tim Panton
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hat maps to the called number
in the relevant context then asterisk will reject the call without
answering. I don't think this helps the OP's
situation, but for the sake of the archives I think its worth
clarifying
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
_
n the Savaje OS.
Do you think that there is a market for this? If so contact me off-list.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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27;t got
performance issues.
Imho roaming support on 802.11 wifi networks is far from being
usable...
The WAP54's have a 'repeater' mode which I've used on occasion.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
__
On 14 Feb 2007, at 17:35, Stephen Bosch wrote:
Tim Panton wrote:
We've used www.Simplewire.com , they have a x86 linux executable
which
we wrap in a
shell script and call from the dialplan with a System() call.
We've been happy with them for years.
Wow! Are these guys in Ca
omeone can convince me otherwise.
No, I don't think they are commercial grade. I have 2 dead Via ITX
motherboards
out of 6 I've bought in the last couple of years.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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lan with a System() call.
We've been happy with them for years.
Tim.
Tim Panton
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tting"
of data seems to take longer and longer. If I kill astmanproxy and
restart it, the system is lighting fast for another few days.
What settings would I need for astmanproxy in order simply to
access the AMI API (I don't need to receive or process events from
asterisk
u should evaluate it at the
very least.
With those sorts of numbers you will have to mute the vast majority
of users
and have them press a button before speaking otherwise the audio will
vanish under the weight of 150 people breathing :-(
T.
Tim Panton
www.mexuar.ne
iax channel is
available,
if it is, the call is routed home, if it isn't the call is routed to
the sip desk phone.
If you have a consistent naming convention you can get this as a macro.
Tim.
Tim Panton
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www.westhawk.co.uk/
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I believe jitter is
off by default.
Thanks,
That's really weird - everything you say makes it look like it would
be a great setup,
except that it isn't :-(
The only thing I can advise now is to get a packet capture
(etherreal) of the IAX stream for a
bad call and see if the pa
On 8 Feb 2007, at 12:33, Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java
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round-trip time between you and the teliax server ?
Do you have the jitterbuffer on or off ? (if you only have 6ms of
jitter, I'd
switch it off....)
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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HTML and
javascript,
so you can get it _exactly_ the way you want it.
Any recomendations?
Clearly I'm biased :-)
Tim Panton
www.mexuar.com
www.westhawk.co.uk/
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playGSM/PlayGSM.html
for a demo.
Gordon
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On 1 Feb 2007, at 16:34, Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two
boxes
that are setup back to back with 4 PRI connections? We're having
problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm
tired of "testing" them in a produc
get out of the modem
with any reliability was 9600 baud - any of the higher speed modulations
failed miserably (and slowly).
If your analog modems are 1200 baud you should be ok :-)
Better get a device to sit in front of the asterisk and have it
split off a couple of analog channels before the
dress and a port map
for udp 4569 in the router. The other can simply register to it with
zero
router config.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 25 Jan 2007, at 06:57, Brad Templeton wrote:
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can
te-machine together I didn't expect the ringing
message _ever_
come after a call is answered.
But it can, if you have
exten => s,1,Answer()
exten => s,2,Playback(your-call-may-be-recorded-blah-blah)
exten => s,3,Dial(Zap/g1/004416128824245) ; this line can generate a
ringing m
etc all of which
play into the choice of phone.
Perhaps you should be looking at the SNOM too ?
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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sions with canreinvite
set to true. Your internal network however does not permit rtp
traffic between
the handsets.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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ernally, save you bandwidth by trunking the IAX traffic
to the central asterisk and avoid all the NAT hassle by using
a single port (outgoing) and refreshing it often enough for the
router to hold it open.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
and however it oftentimes makes the issue worse.
What other traffic do you have on the IAX trunk link ? Even if it isn't
'full' you may be hearing your IAX packets being delayed behind 'bigger'
packets, or sitting in a low priority q
tone could be generated, either in
asterisk - then sent to the softphone as audio media, or alternatively
in the softphone itself - in response to asterisk sending a 'ringing'
message.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
___
/to a specific
extension?
try creating a separate (duplicate) entry in iax.conf for the teliax
connection disallow 729 on that
trunk and use it for fax/alarm calls.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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.168.1.102:
> requested format = alaw,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (ulaw|alaw|gsm),
> priority = mine
Looks like a codec negotiation issue. The softphones are saying alaw
(only), but your
pri trunk is ulaw. Try enabling ulaw
lower case 'w's in the default
nokia
font!
I was pleasantly surprised by it, but it is still a first generation
solution, to be
given to early adopters and technophiles.
I lived with it for a week, and the only thing I can't cope with is
that it isn't
a clamshell.
Ti
results to asterisk.
Then again it would be cheaper to do the re-write at insert time with
a trigger.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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that the caps were clearly sub-standard.
When I replace these systems I won't be buying VIA EPIA again.
If you do buy them, make sure you spec a CD drive in the package.
I didn't and OS rebuilds are a pain to do via a USB DVD drive.
Tim Panto
eave it open to anyone
who has the time/skills to maintain the unicall<-> asterisk
channel driver - or perhaps write a new one.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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en
you are introducing yet more lag.
So, I simply don't think that TCP is suitable for telephony media
streams over any
network where the roundtrip time is of the same order as the packet
interval.
Now there are 'reliable datagram protocols' ( IL for example) but
they aren&
On 30 Dec 2006, at 23:14, Lee Jenkins wrote:
Matthew Mackes wrote:
Check them out: www.neobits.c_m/zultys_wip_2__wi-
fi_ip_voip_sip_telephone_p9656.html
Got 404 error on that link.
try :
http://www.neobits.com/zultys_wip_2__wi-
fi_ip_voip_sip_telephone_p9656.html
Tim Panton
hardware.
Could you tell me which asterisk version you tested this on ? Also
how many of these calls
were over trunked IAX and how many 'normal' ?
Thanks _very_ much.
Tim.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 14 Dec 2006, at 13:32, Jordan Novak wrote:
My need to do this through asterisk is simply the ability to
provide me access with no additional cost to my customer. It seems
like a nice thing to include as long as authentication is done
well. I have worked on a dozen or more types of switc
(in memory?) database, then use some standard
JDBC/servlets (or whatever) to query those events using the
channel current user's as a key.
Now, depending on what you are trying to do, there may be other ways
to get there....
Tim Panton
www.mexuar.net
www
ually goes out via a Public ISDN line, you have to get
the provider to agree to let
you set the outgoing number. Normally they will only let you set it
to one of the inbound numbers
that you have bought from them :-)
If that doesn't help,
please re-phrase the question...
Tim Panton
www.
cat /proc/zaptel/1
And enable pri debugging in asterisk and send the output when you
make a call.
Don't be afraid to ask the provider what they are seeing...
First time I put in a PRI I spent 2 days messing with it before I
rang them, to
be told that they hadn't enabled outbound c
cat /proc/zaptel/1
And enable pri debugging in asterisk and send the output when you
make a call.
Don't be afraid to ask the provider what they are seeing...
First time I put in a PRI I spent 2 days messing with it before I
rang them, to
be told that they hadn't enabled outbound c
On 11 Dec 2006, at 04:25, cb wrote:
I recently purchased a Mediatrix 1124 from an auction of a company
that went out of business. It came with nothing other than the unit
itself.
In digging thru the Mediatrix web site, and various google
searches, it looks like it only supports SNMP setu
listen to old and new messages and change your outgoing
message too.
Regards
You might want to look at integrating my (free opensource) gsmPlay
applet into the web front end of that,
it would let your users play their gsm voicemails without installing
quicktime...
Tim Panton
www.mexu
in asterisk 1.4 and we must wait for 1.6
to be officially supported feature :'(
PJ
On 7 Dec 2006, at 21:29, Vicky wrote:
I am still on asterisk 1.2 branch svn ( afraid of word beta on
server :( ) . I will try out that patch.
Alternatively try setting
${SIP_CODEC}
before yo
.
If you add a USB disk you can even build asterisk on it :-)
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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xten config, I think you need:
Domain/Realm: Asterisk
Tim Panton
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www.westhawk.co.uk/
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email and get things done faster.
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Tim Panton
On 1 Dec 2006, at 03:49, Doug Crompton wrote:
no - make menuselect - does the same thing.
Have you got a (non asterisk) binary or shell script called
menuselect in your path?
try
which menuselect
Tim Panton
www.mexuar.net
www.westhawk.co.uk
rm into the 'h'
extension, that way it gets run whenever the user hangs up.
exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten => h,1,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm")
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
__
aleta - it does both
classic "click to call" as you described and also
"click to call 2.0" where the call originates in a browser based
softphone.
See the url below.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 28 Nov 2006, at 08:47, Olivier Saulnier wrote:
Hello,
I want to use AGI for give some information for a softphone, as:
exten => 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010)
We use Ruby langage.
The line doesn't worksin as this, but works with shell command.
Also, if i modify my ruby sc
On 28 Nov 2006, at 03:01, Eric Bishop wrote:
I am trying to do it with FOP and Calling Circles. Both have closed
code. Anyway to do it from Asterisk?
You could use the 'Local' channel as the argument to the originate
command
and then set it in the dialplan.
Tim Panton
www.
a different way"
That's not at the end... though close to it.
I don't know, a weird interaction with a jitter buffer perhaps.
The only way to be sure would be to get an ethereal trace
and see what is happening on the wire.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
_
s and dump them.
You can fix it by padding the end of any files to a whole number of 20ms
or using a fixed length codec - like gsm.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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tries
it is the default and it won't be right if one of your
firewalls does port mapping...
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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e are others
see:
http://www.voip-info.org/wiki/view/Asterisk+hardware
Tim Panton
www.mexuar.net
www.westhawk.co.uk/industries/voip.html
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On 21 Nov 2006, at 14:34, Jay Moore wrote:
Tim Panton wrote:
On 20 Nov 2006, at 21:46, Jay Moore wrote:
Doug wrote:
Hmmm. I think this may work for WinAmp and
incidently for Windows Media Player:
http://www.mlkj.net/gsm/
No luck with WMP. Anyone else have any suggestions on
playing
termination (120ohms I
think).
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 22 Nov 2006, at 22:21, Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension
On 23 Nov 2006, at 11:36, Zeeshan Zakaria wrote:
iax2 debug is giving following messages repeatedly.
Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: REGREQ
Timestamp: 1ms SCall: 00010 DCall: 0 [xxx.xxx.
157.230:4569]
USERNAME: XXX9072835
RE
nored on reload,
you have to stop and start
asterisk for it to take effect.
If that doesn't do the trick, you will have to write some tricksy
iptables port mapping rules
for 4569
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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ting points
probably make them sign contracts to that effect.
Of course if you can prove to Gradwell (or whoever) that the number is
yours, then it isn't spoofing - even if the call didn't really
originate on that
line.
T.
Tim Panton
www
load of channel local variables, then use the variables
via the
dialplan to guide the user. If the user actually _does_ something that
requires a database write, we have a second AGI at the end.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
___
to all and sundry.
I've written a java program (using asterisk-java) that invokes the
originate method on the manager API.
which can be put in an CGI. If anyone is interested I can clean it up
and post it somewhere.
Tim.
Tim Panton
www.mexuar.ne
On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote:
Same here - wrote an email to support. They claim that their
servers are fine and will get back to me in a day or two...
Now there is a definitive case of a 'lagged' communication channel!
:-)
Tim Panton
www.mexuar.net
www.west
thinking about getting a replacement....
Tim.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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ut swap the machines, put a small device on the DMZ and
the real asterisk
deep in your LAN.
4 is my preferred option, but your situation may vary.
Others will tell you how to get the XXYZ phone to do NAT just right
with the ABCD routers :-
nager port is non-responsive.
Tim Panton
www.mexuar.com
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On 2 Nov 2006, at 14:54, Al Bochter wrote:
VOIP is NOT telephone so the FCC don't have anything to say about
VOIP.
Well not right now.
But in CAN there are cable co. that block the SIP ports and there
is an up charge for them to unblock SIP.
Ask Vonage..
Yet another advantage of us
d only one phone is being used.
Please help before I start buying new stuff to replace some of the
existing stuff, just to find out that it didn't help anyways.
Thanks
Sounds like a clock slip problem on your PRI interface - what is in your
zaptel.conf ?
Tim Panton
years snoms under license but
are quite a bit cheaper
- I'm not sure if they have a PoE version.
Tim Panton
www.mexuar.com
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gistered with them,
but when
they send you a 'new' your box fails to authenticate them as it can't
find a matching
user/friend entry in iax.conf.
Tim Panton
www.mexuar.com
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sk
MIB in the free-ware
from snmp.westhawk.co.uk
(Disclaimer - I wrote large chunks of it so I'm biased :-) )
Tim Panton
www.mexuar.com
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27;ll take a look.
At a guess your softphone has a bug, and asterisk is just issuing a
warning,
but I don't have enough evidence yet.
Tim Panton
www.mexuar.com
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(Plus of course the other IAX advantages everyone else mentioned...)
Tim Panton
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o
the link dropping completely.
IAX will detect total packet loss (in either direction) and hang up
the call
after about 30 seconds. SIP can often not notice the RTP stream has
died (esp if the control channel is still up) leaving you with half a
call
hanging around.
Tim Panton
www.mexua
oving twice is a bit unusual, but I successfully moved
some BT basic rate numbers from BT to NTL (who now own telewest).
It took a couple of weeks and a few faxes.
T.
Tim Panton
www.mexuar.com
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cause your platform does not support dynamic loading :-)
2) because you are debugging multiple versions and you want
certainty about which version you are running.
I'm sure there are other reasons folks can come up with.
Tim.
Tim Panton
works ok, then you might be able to use our softphone.
(What OS is there on your thin clients ? WinCE ?)
Tim.
Tim Panton
www.mexuar.com
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orks? Tab completion is not working here
either. But the fact the console doesn't except any fully typed
commands is more worry some.
I get different results if I run :
asterisk -c; # (which works ok - except for the high CPU) and
asterisk ; asterisk -r # (where the con
On 21 Oct 2006, at 19:50, Joshua Colp wrote:
Tim Panton wrote:
On 21 Oct 2006, at 09:58, Martin Joseph wrote:
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command asterisk -v to start
On 21 Oct 2006, at 17:24, Martin Joseph wrote:
On 2006-10-21 05:09:33 -0700, Tim Panton <[EMAIL PROTECTED]> said:
On 21 Oct 2006, at 09:58, Martin Joseph wrote:
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory
guess that isn't true any more. I'm taking a
couple of mac's
with me.
Can you tell me which version of xcode you have installed? I'm wondering
if the problem is related to the update in the tool chain.
Tim Panton
www.mexuar.com
the byte swapped slug
didn't like my reply (which was wrong) in a way that failed, rather
than the i386 arch systems
which just silently ignored the my error.
My point (if I had one) was that testing against 'odd' architectures
is _very_ informative.
Tim Panton
a couple of bugs in Corraleta that only
showed up when testing against the slug - byte order things if
I remember.)
Tim Panton
www.mexuar.com
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It depends on exactly what you want..
You may be able to do this with the Local channel and manager
Oringinate.
Tim.
Tim Panton
www.mexuar.com
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To
the cdr
may not be in the database (yet) when your softphone posts it's data
(you will probably need to put an index on it to get decent performance)
When it comes time to implement your softphone, please give our
Corraleta Technology SDK
a look it is designed for this sort of thing - ful
tween
3) the remote asterisk has been configured to use 1207 (unlikely)
Tim Panton
www.mexuar.com
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On 16 Oct 2006, at 09:09, Martin Joseph wrote:
On 2006-10-15 23:50:34 -0700, Tim Panton <[EMAIL PROTECTED]> said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton <[EMAIL PROTECTED]> said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton <[EMAIL PROTECTED]> said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(
sk them to pay 3x as much for the
service 'cos of all the
redundancy in your offering.
These are people who grew up with windows 95 - they expect
technology to fail!
Tim.
Tim Panton
www.mexuar.com
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ty shows, we have just (reluctantly) rebooted a
decently constructed
database server with an uptime of 850 days to add a disk, it's
previous reboot
was to add a new tape drive, to my knowledge it has been in service
for > 5 years.
How long do companies keep their phone systems ? 5 yea
vity on the port 80
transfers. There is no data "coming in out of the blue" to you
browser.
This makes it MUCH simpler for you NAT to send the right data to
the right machine.
Which is exactly why IAX is so much better than SIP at this stuff.
Tim P
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