I don't imagine this would be too complicated - don't have any experience
with AsteriskNOW - but on a 'vanilla' linux distro it would just be a matter
of making sure dahdi is loading the correct drivers and doing a couple of
minor config file updates.
On Tue, Dec 28, 2010 at 3:01 PM, Tyler Davis
I'd suggest you take the opportunity to patch that box while its going to go
offline anyways!
*nix is not immune to security problems despite what some might like to
think! Also, I believe Asterisk has had a couple relatively
serious vulnerabilities that have been patched in 1.4 and 1.6 over the
Just out of curiosity, what country are you in?
I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections. I would ask them for
specifics about the "legal" issues with what you are doing - it sounds to me
like they are just trying
On Wed, Sep 22, 2010 at 10:46 AM, Adam Moffett wrote:
> That's probably what I'm going to have to do. Thanks.
>
> > I suppose that merely removing ATA and asterisk from the middle, and
> > plugging a pots line into a fax machine is out of the question.
> >
> >
>
>
>
If you are running 1.6, you c
On Tue, Jun 29, 2010 at 10:39 AM, William Stillwell (Lists) <
william.stillwell-li...@ablebody.net> wrote:
> I use SecureCRT+FX , and use ansi graphics.
>
> Putty is nice w/WinSCP as well.
>
>
>
I'll +1 this - SecureCRT+FX is the first thing I got my employer to buy a
license of for me when I had
Dell server by any chance?
I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues with digium cards in dell servers as
well.
I've actually done something similar to what you have done - isolated the
TE220B onto its own IRQ and set processor
You should be safe to upgrade to the latest Aastra firmware - i've never had
problems with any of the Aastra upgrades - that being said, thats probably
better asked to Aastra support instead of here.
As for the 169.254.0.0/16 address...
theres nothing "weird" about these addresses... 169.254.0.0/
On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion wrote:
>
>>
> Attached is an untested (I did not had the time yet) port to Asterisk
> 1.4.29.1 (DAHDI). Maybe the modules need some adaptions too.
>
> Maybe someone wants to give it a try.
>
> regards
> klaus
>
>
Just as an FYI, your 1.4.29.1 patch a
On Mon, Mar 15, 2010 at 1:44 PM, Ott Rose wrote:
> I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1
> second flash on the screen then the phone hangs up. the FOP says it is on
> DND but some ext are still getting calls. once i do a *76 FOP still says I
> am on dnd. I am run
Tue, Mar 9, 2010 at 11:34 PM, Matt Watson wrote:
> Hi Dave,
>
> Sure enough my astdb does contain references to VM files as shown with
> strings - doing the database dump however does not show the references.
>
> I'm not sure about the internals of how Berk DB works, howeve
Hi Dave,
Sure enough my astdb does contain references to VM files as shown with
strings - doing the database dump however does not show the references.
I'm not sure about the internals of how Berk DB works, however I;m also
seeing references to lots of other data that really shouldn't be part of
I just downloaded a copy of this, by any chances does Zoiper by any chance
have diff files available for a more recent 1.4.x release? (I know 1.6 is
probably out of the question)
Thanks,
--
Matt
On Mon, Mar 8, 2010 at 12:11 PM, Matt Watson wrote:
> Awesome!
>
> I was an Attrafax cus
Awesome!
I was an Attrafax customer and was very disappointed when it vanished and
couldn;t get new modules for newer versions Asterisk with our paid license.
If anybody is working on t38 gatewaying code for 1.6, it would be worth a
look at this, as I can attest that Attrafax worked quite well at
I'm having this EXACT same problem, I haven;t been able to narrow down the
cause of it yet, but it seems to me that users are receiving notifications
for voicemails in mailboxes that belong to other people, as sometimes their
mail count magically disappears, which I have been suspecting is when
som
There already is a special character to tell asterisk not to parse a line...
its: ";" that is why the default configuration is completely filled with
lines that start with ; its considered a comment character to asterisk and
will make it ignore the line... you'd just want to add some extra charac
I'd be interested in this as well... I;m coming up to an upgrade cycle and
trying to decide if I should upgrade to the latest 1.4 or 1.6.1
When others that have commented on this say they have had problems with PSTN
connections, are you referring to T1 or POTS? I;m in a T1 scenerio, so if
problem
Not that I;m exactly a big fan of NFS but... why would you choose to
implement a filesystem that was designed to emulate Windows shares for your
UNIX-type environment? You have to kind of expect odd problems like this
when you choose to use things for other than their intended purpose. Samba
I wo
The 1.x firmware for Aastra's (for the 9112i / 9133i / 480i) do support some
of the XML functionality that you see in the newer 2.x firmware (for the
more recent models).
I;m not sure if controlling LED status of the keys is supported by 1.x - but
you should be able to find that out by taking a lo
On Mon, Feb 23, 2009 at 9:38 PM, Steve Edwards wrote:
> On Tue, 24 Feb 2009, David @ULC wrote:
>
> > When I am trying to delete voice logs,
> > [r...@vicidialnow monitor]# rm * -r -f
> > -bash: /bin/rm: Argument list too long
>
> In the past 30 days, you've asked questions about
>
> configuring Ap
I find it a little strange that for some reason your box is using includes
located in /usr/local... while there could be reason for this, that seems
like a sign that something might be a little broken on your box.
Also, if you don;t mind me asking...
why would you want to install * directly in /u
Your boss is going to change their mind when they see how awful and
unreliable this setup is going to be lol.
On Fri, Feb 20, 2009 at 1:42 PM, Ignacio wrote:
> yep, it is mainly due to cabling issues. My boss doesn't want to
> recabling the office.
>
>
> On Fri, Feb 20, 2009 at 7:19 PM, Jeff L
going to
stay that way or not... But I haven;t seen anything that would indicate
otherwise.
--
Matt Watson
On 30/01/09 11:11 AM, "Jeff LaCoursiere" wrote:
>
> Following up my own thread, I am kicking myself for quickly posting
> without doing a bit of research. Apparently (n
On Sat, Jan 17, 2009 at 11:51 AM, Steve Gladden <
aster...@michiganbroadband.com> wrote:
> The scenario we have is fax send/recieve software that ONLY talks T38
> and an asterisk box.
>
> We have ITSP providers that do NOT talk T38 but G711 only.
>
> Does asterisk have the capability to take the T
On Thu, Jan 22, 2009 at 6:05 PM, Wilton Helm wrote:
> >making sure to patch any holes through which the hacker might have come
>
> In my case, I had been getting regular attacks through SSH for months,
> probably 100 a day (bots). Apparently after nine months of this, someone
> stumbled on to m
On Thu, Jan 22, 2009 at 5:30 PM, Wilton Helm wrote:
>
> Tripwire would be fine, if it had a baseline, but I don't think its any
> good after the fact.
>
Correct - tripwire does need to be setup beforehand, and its not the most
simple thing to setup *properly*. After the fact... you are basicall
On Thu, Jan 22, 2009 at 1:52 PM, David @ULC wrote:
>
> I tried :
>
> 1. Shut down the machine. (Ctrl+Alt+Del)
>
> 2. When it reboot and reach the CentOS boot up screen, then press any key
> to go into a select menu. Then press "e" and navigate to the second line
> "grub.conf" line (kernel) and pr
On Thu, Jan 22, 2009 at 1:48 PM, Wilton Helm wrote:
> There have been a number of answers provided. The one that was given to
> me when I encountered this same problem was to boot a live CD, mount the
> root file system and delete the password file which would force your normal
> distro boot to
Yes, it is available on the SPA2102 - you just login to the web interface,
goto the advanced section, then lan setup... its the very first option.
--
Matt
On Thu, Jan 22, 2009 at 3:29 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:
> On Thu, Jan 22, 2009 at 3:11 PM, Jeff LaCoursi
work almost all UNIXs)
(b) is the more generic and preferred method IMO - it should work just about
everywhere... unless you have total disk encryption or encrypted filesystems
and are unable to mount the partitions... in which case... best of luck to
you.
--
Matt Watson
/dahdi.makeopts - but I have not verified
that.
--
Matt Watson
On Thu, Dec 18, 2008 at 11:49 AM, Jerry Geis wrote:
> >
> > Jerry Geis schrieb:
> > >/ Is there a way to install DAHDI and NOT download the echo canceler
> files?
> > />/ I dont have firewall acces
g... but i would expect very poor results.
3. You are right, you can';t really just make one yourself from scratch, you
need a source that has already been tuned properly to use as a reference for
creating your own.
--
Matt Watson
On Thu, Dec 11, 2008 at 11:01 AM, Olivier <[EMAIL PROTEC
I;m using Aastra 480i's 9133i's, 9112i's, and 57i's and none of them have
experienced problems with qualify=yes.
I;m currently on Asterisk 1.4.17, but I've also tested them with 1.4.14 up
to 1.4.19.
--
Matt
http://www.mattgwatson.ca
On Fri, Aug 15, 2008 at 10:59 AM, Drew Gibson <[EMAIL PROTECTED
I've seen it before infact there is a website setup where people can
post stuff made with it... kind of super nerdy!
http://www.ratemydialplan.com
--
Matt
http://www.mattgwatson.ca
On 7/27/08, Peter Lindquist <[EMAIL PROTECTED]> wrote:
>
> Dean Collins wrote:
>
> I just stumbled across thi
c:5839 set_destination:
> Can't find address for host '"72.16.1.20'
>
Might want to post a sip debug of one of the sessions from the Mitel phone.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided b
I believe HPEC actually is the same algorithm (G.168) that the HW echo
cancel modules use.. the difference being that HPEC uses up CPU cycles and
its performance will be impacted on a system with higher CPU load, whereas
the HW modules have a dedicated DSP for it.
http://blogs.digium.com/2007/09/0
12:52:55 nelson Completed startup!
Thats on my TE220B. the VPN450 is the hw echo can daughter board.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 P
#x27;t need to
install libpri if you are just using a TDM400P (since its not a PRI / BRI
[1.6 libpri does BRI as well] card).
Might save you a little bit of time in the future, and its one less thing to
consider as a problem.
--
Matt Watson
http://www.
e some of that money to further develop spandsp.
That being said... i;m also quite pleased to see T.38 support being worked
on for Asterisk... its a pretty important area to further develop IMHO.
--
Matt Watson
http://www.mattgwatson.ca
On Thu, Jul 10, 2008 at 11:54 AM, Steve Totaro <
[
is it only cell phone calls that don't work? or is it any external call
coming in over your lines?
What type of inbound lines do you have? I;m guessing analog lines... if
thats the case what type of signalling are you using?
if its only cell calls and not all external calls then I have no idea
nce TFTP is authentication-less, but its still possible...
and if security is in mind you could limit your TFTP server to specific
source IPs
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com -
the last year or so... so i;m really just making some logical
assumptions here.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, A
Let me be the first to say: "Pick whatever distro you are comfortable with"
Distro is more of a personal choice than anything... ultimatly they all have
the same software available to them (for the most part), they all just do it
a little bit differently.
--
Matt W
p I believe is
the Xorcom AstriBank... if you don;t actually have an AstriBank then there is
no sense in even compiling/installing the drivers for it.
I;m guessing you haven't run a make menuselect to select only the drivers you
need?
--
Matt Watson
http://www.mattgwatson.ca
_
like you have installed asterisk-addons and not asterisk itself... if
you compiled from source maybe you just forgot to 'make install' for
asterisk?
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http:/
..does anyone want to create an app to send notification
> to a cell phone to set/clear these bits?
could you provide a link to where you got the info from? I'd be interested in
seeing if i can get this to do anything useful.
--
Matt Watson
http://www.mattgwatson.ca
ge me on my cell via dialup TAP (so my
monitoring server can still page me in the event of an internet outage).
Anyhow, sorry that message got a bit lengthy pretty fast!
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provide
tle like buying stuff from Sony... you pay a
bit of a brand tax just to have the APC logo printed on it. However I think
that the APC logo on something means alot more than the Sony logo :)
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth
o turning those ports on/off however is irrelavent to its device
classification it could be a web interface like this device, a telnet/ssh
interface, it might not even have any remote capability and just have
physically switches for each port.
--
PC model I mentioned in my last post will do load metering...
it'll cost you about twice as much as the one Steve posted however. The sale
price that is, couple hundred more than the regular price.
--
Matt Watson
http://www.mattgwatson.ca
___
--
ce with either however.
All in all it looks like a decent product... i'd be interested in hearing from
anybody that might of been using them for a long period of time (1-2yrs+).
I'm pretty picky about power distribution, i've seen bad power cause too many
problems in my com
nd out of
Asterisk was suitable for their organization... and it was seemingly like
they were literally trying to offload their job to this list
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-dig
uld matter, but you don;t need the duplicate group=,
signalling=, switchtype= in zapata.conf
4. you can ditch rxwink= that setting is for non-PRI T1s
try that and see if that helps... I suspect the span not being used as primary
timing source is whats causing your greif.
good luck!
--
M
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax on FXS
On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote:
> On June 9, 2008 01:34:31 pm Eric "ManxPower" Wieling wrote:
> > You should not expect FaxOverVoiceOverIPOverInternet to work well. If
> > you sti
spect that may be
your problem. Might want to check the archives for other issues that people
have talked about DNS as a possible cause and see if there are any
similarities.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation P
end a fax through a
VoIP system using a voice-codec like G.711
However, there is the T.38 protocol which is designed to solve this exact
problem, Asterisk support for it is just rather limited currently (pass-thru
only).
T.38 often gets referred to as FoIP (Fax ove
the SIP ATAs obviously have to have T.38 support - for example a Linksys
SPA2102 should do it for you. I've never tried faxing between ATA's so I
don;t know if they can actually negotiate T.38 support between each other,
but I don't really see a reason why they couldn't.
--
ure
for an FXS port to attach a fax machine to it...
keep in mind that faxing over VoIP is extremely tricky at best, but if your
entire call path is TDM then you shouldn;t have much of a problem.
--
Matt Watson
http://www.mattgwatson.ca
___
-
In short, fxotune adjusts line impedance, where as adjusting gains I believe
is essentially adjusting the amplification / deamplification of the signal.
http://www.voip-info.org/wiki/view/Asterisk+fxotune
--
Matt Watson
http://www.mattgwatson.ca
On June 6, 2008 12:43:51 am Noah Miller wrote
ably isn't going to give you any benefit.
--
Matt Watson
http://www.mattgwatson.ca
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: Wednesday, June 04, 2008 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussio
On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote:
> I used to do lots of Asterisk, but got "an offer I couldn't refuse," and
> went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
> to set up a test system. One thing I'd really like to get my hands on is
> recent firmware, etc
On May 23, 2008 11:25:55 am Dennis P. Clark wrote:
> Will fax and dial-up internet work through the gateway?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Joe
> Carroll
> Sent: Friday, May 23, 2008 8:51 AM
> To: Asterisk Users Mailing List - Non-C
On May 22, 2008 04:42:27 pm Steve Totaro wrote:
> PS. Figured I would start with DHADI now.
>
psst. its DAHDI ;)
--
Matt
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users maili
On May 22, 2008 02:06:06 pm Jared Smith wrote:
> On Thu, 2008-05-22 at 10:48 -0700, Douglas Garstang wrote:
> > We didn't want to be generating actual network traffic for this, so I
> > tried originating a call to [EMAIL PROTECTED]
>
> Why not try [EMAIL PROTECTED] and see if that solves the proble
Does your extensions.conf have any more configuration than what you've shown?
If not, then you are lacking dialplan for anything but internal calls.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users@lists
You might want to see if you can change the IRQ assignments in your servers
bios (might have to turn off the "PNP OS Installed" option if you have one)
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Cavanna, Richard [EMAIL
PROTECTED]
Sent:
k Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Machine Options
Matt Watson wrote:
> I believe Asterisk 1.6 with app_fax supports T.38 origination and
> termination, that is not gatewaying, however if origination and termination
> are already there, gat
The first part of this is kind of off topic as it doesn't answer OP's original
question, but instead is a reply to one of the replies.
Cisco is certainly not the only option for doing T38 gatewaying with Asterisk.
I believe Asterisk 1.6 with app_fax supports T.38 origination and termination,
th
On May 19, 2008 06:49:23 pm Kevin Smith wrote:
> I almost hate to admit this...but I'm still running Asterisk 1.2 on
> Fedora 4 :D
IMO theres nothing wrong with running an old version of * or an old version of
the OS... as long as the box doesn't have a public IP bound to it.
If its an internal-
On May 19, 2008 03:21:34 pm Aaron Stranberg wrote:
> Folks,
> We are a small office with remote users less than 20 total phone
> extensions, and I am looking for some guidance on choosing between
> asterisknow and a centos/ubuntu or any other os with an asterisk +
> asteriskgui build out? Lookin
On May 19, 2008 12:51:09 pm Grygoriy Dobrovolskyy wrote:
> Hmm, i dont like aastra really much, their transfer management is not human
> friendly ;)
What do you mean by that? I've run my Aastra's with BLF using both
Aastra's 'blf' mode and 'blfxfer' mode... the former is basically attended
tran
On May 17, 2008 06:59:43 am Gordon Henderson wrote:
> On Sat, 17 May 2008, bilal ghayyad wrote:
> > Well, why Digium is still using this kind of power
> > connector while all new machines does not come with
> > these types?
>
> The new machines that I buy come with "legacy" power connectors. The fl
You can NOT use bogomips as any kind of measurement for system performance.
First of all, Bogomips is a linux-specific thing and not available on other OS
that Asterisk runs on.
The second, and far more important point. "Bogo" is taken from the word
"Bogus". Bogomips are not a measurement of
Is there any reason you don't want to use Wait()?
However, I would use WaitForRing() myself - its also a great solution on dirty
analog lines where you receive "phantom calls".
That being said, I don't know how to do it without using some form of Wait.. as
far as I know zapata.conf doesn't prov
You'd probably want to run something else to handle your registrations like
OpenSER with that many phones.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk Users Mailing List - Non-Com
I'm using Aastra 57i + 560M sidecars for receptionists... the only downside is
that they support a max of 50 BLF subscriptions... you can setup up to 180 blf
keys with 3 560Ms but it will still only subscribe to a max of 50... from what
I understand it's a firmware limitation.
For 4-6 phones yo
Do you mean
"What do I need to configure on my * installation so that only registered sip
users can make calls?" ?
If so, you are going to need to give a lot more details regarding your current
configuration for you to get any answers.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PRO
I'm not sure if a full-height card would fit (vertically) in a 3U chassis...
but I would probably also assume that if it would not, that the chassis/mobo
would have a PCI/PCI-Express riser card that would mount the cards horizontally.
Might want to check that out with the manufacturer of the cha
Poking around the zaptel SVN earlier today i see support was added for an
AEX410 card recently...
I'm going to go out on a limb and assume this is the PCI-Express version of the
TDM410?
Any hints on a general availability date?
--
Matt
___
-- Bandwi
I just took a quick look at the dialplan that freepbx uses for doing call
confirmation... the dialplan part of it is actually quite simple... its just a
matter of setting the USE_CONFIRMATION varialbe =TRUE.
However, the actual magic looks like it happenes through its dialparties.agi...
which i
FreePBX has this functionality... they call it "Confirm Calls"
I;m not sure if you can set it on actual extensions, but I know you can set it
on ring groups.
I don't imagine the dialplan for doing it is very complicated if you wanted to
do it by hand.
--
Matt
__
I'm a Gentoo guy myself too... but the best advice I can give is just
re-hashing what others have already set... pick whatever you are the most
comfortable with... and if support contracts are important to you, then that
will be a factor as well. I've used most of bigger distros out there over
Are you using IAX2 as your transport between the 2 servers or SIP?
If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either
machine? If so, you may be encountering the IAX2 bug that some have been
discussing on the list recently you can read it here:
http://lists.digium
I don;t have any answers for you...
But I would love to hear about the results after you get this working and what
road blocks you hit and how you overcame them.
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED]
Sent
Why are you trying to change the ToS from 46 (0x2e) Expedited for the RTP/RTCP
packets to 16 (0x10)?
I mean... these values really only need to be meaningful to yourself, your
switches, your routers etc however
ToS 46 (0x2e) is the "standard" value for RTP / RTCP as it is basically the
hig
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX
in a flash, etc. etc.
If you have trouble finding it let me know and I can send you it.
I can;t really vouch for its quality, but I do use it and it does work... but
i;m not sure how well it handles multiple results.
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
Behalf Of Philipp Kempgen
Sent: Tuesday, May 06, 2008 12:27 PM
To: Asterisk Users
Subject: [asterisk-users] "This e-mail is confidential" ... (was: Re: PRI
D-Channel reconfiguration = crash asterisk?)
Matt Watson schrieb:
> Disclaimer Statement: This e-mail is confidential and is in
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
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
What Godaddy.com has told you is more or less correct.
Its not their fault that Chinese visitors cannot hit your pages... the internet
is China is highly censored, and quite often they firewall even very large big
name sites like BBC news. Typically they block sites that have any type
discuss
There is really no reason why you cannot.
Personally... I'd avoid using Java for AGI's that you think are going to
receive heavy use... simply because the JVM adds a lot of overhead, and
possibly a very real performance impact from having the load the JVM everytime.
Of course there is overhead
I don't know about the Cisco phones... I;m using Aastra phones which I can
send a SIP NOTIFY to have them check for updated config... when they detect a
new config they reboot themselves and download the new config.
But your switch might also have an option to disable PoE on a per-port basis...
err, that should of read TE220B
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Matt Watson [EMAIL
PROTECTED]
Sent: Thursday, May 01, 2008 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel
Does anybody know if this version fixes the soft lockup during ztcfg using a
TE200B?
http://bugs.digium.com/print_bug_page.php?bug_id=12468
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development
Team [EMAIL PROTECTED]
Sent:
This is my understanding of hyper threading, which I believe to be accurate.
Basically, as some have mentioned previously, the OS 'sees' your single
physical core processor as 2 logical processors, in generally, logical
processors are treated exactly as if they were real processors, and in the c
You might want to begin with tuning your rxgain and txgain settings... there
are a few methods for doing this on the internet, unfortunatly nobody can give
you exactly values to use for tx/rxgain as they will be not only specific to
your install, but specific to every single analog line you have
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be
using Zaptel (even your sangoma... the script might just be configuring it for
you)...
and btw, software echo cancel happens in the zaptel kernel driver... it has
nothing to do with the hardware (hence why its a softwa
I can;t imagine what headaches you'd have going from 1.4.11 to 1.4.19.1... that
is a minor version upgrade... no real change in functionality thats
basically 8 versions of bug fixes... if you just apply the IAX2 patch, you'll
be fixing 1 out of probably a hundreds of bugs. Going from 1.4.x
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