, if someone wants to take over, be my guest :)
roy
---
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
Tlf: 98013356
---
Why is it drug addicts and computer afficionados are both called
users? -- Clifford Stol
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and playing guitars at people
- Terry Pratchett
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Roy Sigurd Karlsbakk
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and playing guitars at people
- Terry Pratchett
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jeans and playing guitars at people
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guitars at people
- Terry Pratchett
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, like dressing up in jackboots and shooting people, or
dressing up in white sheets and lynching people, or dressing up in
tie-dye jeans and playing guitars at people
- Terry Pratchett
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Roy Sigurd Karlsbakk
[EMAIL PROTECTED
hi
reading http://www.digium.com/en/docs/misc/compatibility_notes.php, I
see that some rather cool servers from IBM are listed unsupported.
can someone please explain what's so bad about these?
roy
--
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
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In space, loud sounds, like
I am trying to do, yes, as to do all DID administration
myself without contacting the switch monkey.
It's quite possible, it seems, by sending a cause 34, lying about no
bchans being available to handle the call.
roy
--
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
(+47) 98013356
---
In space
accepting them to
the dialplan).
Thanks you
roy
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[EMAIL PROTECTED]
(+47) 98013356
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In space, loud sounds, like explosions, are even louder because there
is no air to get in the way.
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them to
the dialplan).
Thats how it should be, it takes all of some 2 ms (give or take some,
I might be off) to reject a call in the Dialplan which shuldn't be a
problem.
I just wondered what PRI_CAUSE to use, really.
roy
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!DSPAM:44e8a400319061757298282!
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In space, loud sounds, like explosions, are even louder because there
is no air to get in the way
regards
roy
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In space, loud sounds, like explosions, are even louder because there
is no air to get in the way.
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hi
this site now also has a wiki, and the whole site will be migrated to
a wiki-only after some time
so just add your stuff :)
roy
On Jun 20, 2006, at 6:15 PM, Roy Sigurd Karlsbakk wrote:
hi all
I just setup a new site, perhaps soon a wiki, to collect what's out
there of useful
hi all
I just setup a new site, perhaps soon a wiki, to collect what's out
there of useful backports from Trunk/1.4 beta back to 1.2. Take a
look at http://http://www.asterisk-backports.org/ and judge for
yourself ;)
roy
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Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
(+47) 98013356
hi
does anyone have an idea how it could be possible to do email - fax
gatewaying with asterisk + app_txfax, but still keep track of who
sent the fax? i've thought a little about smtp auth, but it doesn't
look too easy to integrate smoothly with asterisk
roy
On Feb 8, 2006, at 2:41 AM, Jeremy McNamara wrote:
hi
I recently spoke to mr McNamara on IRC, and he mentioned there was
a far better way to do realtime-stuff than the usual realtime
in asterisk, and that this was GPL. He failed, however, to ever
mention how this could be done, so I
Sigurd Karlsbakk
[EMAIL PROTECTED]
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with two sangoma A104 cards, and
that could just handle the load. I beleive the digium cards will
require about the same load, since it is mainly zaptel eating cpu...
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[EMAIL PROTECTED]
---
In space, loud sounds, like explosions, are even louder because there
is no air
colleagues beat you to death :)
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Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
---
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is no air to get in the way.
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unplugging the other NICs cable?
Best regards
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
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hi
i'm setting up a rig to handle quite a few SIP clients, so i need a
way to simulate, say, 20k SIP ATAs. Does anyone know how? This should
of course be as close as possible to 'reality', meaning n% calls per
client and the usual REGISTER/OPTION traffic.
thanks
Best regards
Roy Sigurd
sure, but I need to simulate the SIP REGISTER and OPTION traffic sent
by ATAs as well.
Best regards
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
---
In space, loud sounds, like explosions, are even louder because there
is no air to get in the way.
On Jan 29, 2006, at 7:26 PM, Wai Wu wrote
?
Short answer: Yes
Long answer: They use the zaptel drivers and are recognized as a
Zaptel device. You do have to load and configure the Sangoma wanpipe
drivers first, but in the end it'll function as a timing source just
like a Digium card
MATT---
On 1/26/06, Roy Sigurd Karlsbakk [EMAIL
Hi
Does anyone know a good, scalable switchboard solution for asterisk?
I've been looking around and I've found a couple but I'm not sure yet...
Have anyone here used one in large environments? We need usable GUI
with the usual stuff like queues, transfer, meetme etc
roy
hi
building a new setup, we want to try using sangoma cards. can these
be used as time sources the same way as TE410Ps?
thanks
roy
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Hi
The latest new about the Open Source Asterisk PBX is now that mister
Spencer himself has forbidden Allison to work with any other open
source project than Asterisk. I don't know about the openness in such
an action. This sounds more like the ways Microsoft and other large
monopolists
What part of Mark left the decision up to me did you not grasp from
the original thread? Stop spreading FUD.
Brian was very precise in telling where Allison had got her orders
roy
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What part of Mark left the decision up to me did you not grasp
from
the original thread? Stop spreading FUD.
Brian was very precise in telling where Allison had got her orders
To which Allison responded Just so we're clear, Mark left the
decision up to me. Again, what part of that didn't
Brian was very precise in telling where Allison had got her orders
Yes, Brian was very precise about it but let's be perfectly clear
here: Where
is Brian's UNEQUIVOCAL EVIDENCE that Mark forced Alison's hand?
Brian has
none, that's where it is. He is speculating, pure and simple.
It's
i heard some talk about something in zaptel is currently
incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
I was thinking more about the CVS HEAD version
My machines are totally CVS HEAD and have been
I read roadmap on www.openpbx.org.
Does chan_exosip2 will be able to provide a real sip
proxy ?
What about asterisk solutions ?
I guess you can use chan_exosip2 with asterisk if you hack it in
yourself. Also, as soon as asterisk is released in one single GPL
license, it may as well be
hi
i heard some talk about something in zaptel is currently incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
thanks
roy
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i heard some talk about something in zaptel is currently incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
I was thinking more about the CVS HEAD version
roy
yes
On 28. sep. 2005, at 15.54, Tom Hayden wrote:
You're going to need to explain a little more. When you say central
are you talking about an SMSC?
--
Tom
On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
hi
is it possible to use asterisk as an sms central to send SMSes
directly
for the community, I think it is important to have at least t.38
passthrough first then the other devolpments.
In this way t.38 can be easly spreaded and catch up more supporters.
What do you mean more supporters. t.38 is only *reliable* way for
transporting
fax over ip. Fax over g711 is
What do you mean more supporters. t.38 is only *reliable* way for
transporting
fax over ip. Fax over g711 is pure luck...
Hi,
it is rather a question of IP quality than good luck.
I think, 99.9% of all faxes are transported via G.711.
Is there any telecom network operator left using
hi
is it possible to use asterisk as an sms central to send SMSes
directly to clients on PSTN instead of just communicating with a
central? the telco to which we're currently connected doesn't have a
central
roy
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It seems that HFC-S cards can be connected with asterisk in a few
different
ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried
isdn4linux (severe
echo, reproducable on every inbound call) and zaphfc (intermittent
echo,
disappears within about 30 secs of the call starting).
What's
hi
i get these messages every now and then
-- PROGRESS with cause code 34 received
wtf is this?
roy
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hi
i've heard it should be possible, but i can't find out how...
I want to configure a bunch of asterisk boxes to do SIP/PSTN
connectivity, and I need SER or something to do some balancing in
front of them. The requirements are listed below.
* SER MUST accept and load balance incoming
hi
anyone here that knows a good howto of setting up rate-engine?
i've made some changes to make it work on cvs head, but the
documentation is rather poor, so i just wonder..
roy
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hi
is this stuff still available?
roy
On 14. okt. 2004, at 16.10, Ben Merrills wrote:
I've been doing some work on a queue log analyser for a while now,
getting the basics in place, an example of which you can find at
the URL
below. However, just wondering what information people think
Hello!I'm new to asterisk and linux, so please don't blame me if I write silly things :-)I'd like to setup a system with IVR only.I'll use a SIP gateway to receive calls from the outside world, and I'll install asterisk on a dedicated linux server placed in another location that will be
hi
is it possible to do all the user authentication on SER and somehow
allow calls proxied by SER through asterisk without any direct user/
peer-to-asterisk authentication?
roy
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I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
of memory. This is serving about 75 sip clients, Polycom500's and
600's. We are running into problems with the memory. Asterisk, right
now, is using about 1.8GB of system memory. I am using Asterisk
1.0.7,
Zaptel 1.0.7
hi
can someone please tell me how much memory asterisk requires? I'm
running 1.0.7 and after just a couple of days uptime, this is the
process as reported from ps axfv
13766 ?S 0:15 0 605 1365974 1265468 61.0 \_
asterisk -vvvg -c
roy
can someone please tell me how much memory asterisk requires? I'm
running 1.0.7 and after just a couple of days uptime, this is the
process as reported from ps axfv
13766 ? S 0:15 0 605 1365974 1265468 61.0 \_ asterisk -vvvg -c
It would appear to require about 1.4GB, based on all the
hi
if receiving a call, I lookup wheather or not that call should be
diverted at user's request. if it should be diverted, I want to use
ForkCDR to keep this entry
orig srcorig dst
and add one more
orig dstdivert dst
but with forkcdr, I only get
orig dst
can someone tell me more about this?
On Apr 14, 2005, at 17:55, Roy Sigurd Karlsbakk wrote:
hi
for what I can see on digium's site, there is an x86-64 optimised
g.729a codec. is this particularly optimised for intel or amd? I
wonder most about sse/3dnow/whatever, as AFAICR this is quite
hi
for what I can see on digium's site, there is an x86-64 optimised
g.729a codec. is this particularly optimised for intel or amd? I wonder
most about sse/3dnow/whatever, as AFAICR this is quite different
between the two.
roy
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Technically speaking not. But Sangoma's support seems to be pretty
much
better.
My understanding is that to an extent when we buy Sangoma we're
putting the dagger to Digium. They're glad to use Asterisk as a
selling point for their hardware, but unwilling to donate anything
back to the
What about pricing of the Sangoma compared to Digium, is it comparable?
about the same. last i checked the digium te410 was $1599 and the
4-port e1/t1 card from sangom was $1699.
Can Sangoma card handle modem data incoming calls at all?
iirc, modem data is just voice/noise :P.
roy
cpu load on te4xxp cards is very low, and now that they have echo
cancellers as add-ons cards, it will be even lower.
I can't speak on hardware compatibility as i never tried a sangoma
card.
(But i can say that in the last year i've never had an issue with
digium
cards and we have 8 in use.) The
Hi all
Does anyone have any details on the actual differences of using Sangoma
PRI cards as compared to the TE410? How are CPU usage, interrupt load?
Are there other diffferences?
PS: Don't tell me 'buy digium since they gave us asterisk'. This is a
technical post, not a political one.
roy
hi
is there a way to keep dynamic queue members over a restart? I don't
think I can use agents, since some members are cell phones dialed
through zap
roy
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x-tad-bigger1 RAID5 server with 2 SATA HDs/x-tad-bigger
Last I heard, RAID-5 requires three or more hard drives :P
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1.Does Asterisk support SS7 and ISDN?
ISDN is supported out of the box. SS7 support is (or will soon be?)
supported by a commercial version of Asterisk. Search the list
archives or
post to asterisk-biz.
Steve Underwood (here on the list) has a commercial ss7 solution for
asterisk.
roy
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729
as
I'm running out of DSPs using a high complexity codec on the Ciscos.
I
would think it would work just as g729 does, which has been working
fine
for me, but
what is the best way to do that (also with accounting and
authentication).
which one of those options
1) sipphone - SER - ASTERISK - SER - PSTN
2) sipphone - SER -ASTERISK -PSTN
on the first option i am trying to return the call to the ser after
it's pass the asterisk for some routing
google
On Mar 22, 2005, at 12:59, Ronald Wiplinger wrote:
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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any chance of seeing some code soon?
On Jan 20, 2005, at 12:17, Ben Merrills wrote:
I've not released the source yet, I asked last week on the mailing
list for people to send me over some example queue_logs, because so
far I've only been able to test the software against my own.
I have however
Hello Everyone,
I am trying to find a single port isdn pci card in the usa for
asterisk,
but it seems everything is abroad.
Does anyone know a good place to find a BRI S/T and U card for north
america?
Perhaps it could be possible if you get an NT1 box giving you an S0 bus
and then using a
Maybe I've missed it but I'm wondering if there has been any movement
towards getting t.38 support into asterisk.. has there been any news?
Where is t.38 support at? will it even happen?
Steve Underwood is working on it. There's a bounty at
I tried wiki, but I got too many pages (I think all of them), ...as
answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
see the perl agi package from http://asterisk.gnuinter.net/, the
agi/agi-test from the asterisk source and
You mean that if on a certain queue, your agents are using SIP or IAX
phones, and you want to do a check so that when a cllers tryies to
get into
the queue, if no agent is logged in, do something else with the caller
instead of hanging up?
Actually, I think he wants to go one step deeper, and if
hi
having a queue with some SIP members, is there a way to check how many
of them are connected to asterisk, and if none are, go to a different
context?
thanks
roy
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I've not released the source yet, I asked last week on the mailing
list for people to send me over some example queue_logs, because so
far I've only been able to test the software against my own.
I have however made a lot of changes to it since last I posted about
it.
How is the progress on
On Tue, 2005-03-01 at 08:03, Kristian Kielhofner wrote:
Tony Mountifield wrote:
I've just set up a new box with FC1+updates and the latest Stable
Asterisk from CVS.
Why are you using FC1 when FC3 is out? Better yet, why are you using
FCx at all?
Why not? What are you, some sort of Debian
hi
I'm trying to make video work over SIP between two softphones
I can get audio, but video fails
sip debug is here
http://karlsbakk.net/videotest.log.gz
can someone take a look at it, please?
roy
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test.
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HELP NEEDED TURNING OFF THE cAPS lOCK KEY
:)
On Feb 25, 2005, at 20:07, Edward Banfa wrote:
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog phones that i would like to connect
to my Mediatrix box and then connect the Mediatrix
sure, but what about using asterisk?
On Feb 22, 2005, at 12:39, googleplex wrote:
google for inalp isdn sip gateway
On Tue, 22 Feb 2005 12:23:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP
videophoning with asterisk
Guys.. Im doing a simple IVR system with some menus but I was
wondering,
maybe it already does but does asterisk keep track of themenu hoices
that
each call did? for example, is a caller calls in and then hits 1,3,2,6
does
that stay on some log file?
I doubt it, but you could always run off an
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP
videophoning with asterisk using libpri/zaptel etc?
roy
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To
Disable the call waiting feature in the phone, so it will signal 486
- Busy here to additionally incoming calls.
Is it possible to test if a call to SIP/xxx is in place before dialling
out? This could help a lot to centralize administation of whether or
not to use call waiting instead of
hi
is the wiki down again?
roy
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* Should I mail something to digium? ;)
fax them the agreement from http://www.digium.com/disclaimer.txt
roy
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[...] In the meantime, get a Sipura 2100, supports 2 729 calls and
has both WAN/LAN ports.
I was told that the Uniden DTA200 also supports 2 g729 calls. I'm
buying one to test. Street price around US$ 90.
Another one with dual g729 channels is MTA V102. Street price US$ 100.
Also will test this
I wouldn't recommend the grandstreams, I had very bad experience using
the grandstream 102, It kep locking up on me. The buttons are very bad
buttons. The sound quality is just as bad.
grandstream barbie^H^H^H^H^Hudgettone phones really sucks. they're
cheap, and that's it
roy
I wonder what makes the difference between inserting 4 HFC-S cards
(cca. 120
EUR) and using 1 QuadBRI card (approx. 700 EUR) ?
What makes such difference ? Is it possible to do first configuration
?
With what drivers ? Is it stable ?
1 HFC-S card - lots of interrupts
4 cards - interrupt havoc
Adding in experimental patches willy-nilly, especially ones that have
the potential to cause huge problems, confounds attempts to isolate
bugs and test functionality.
Mark does a pretty good job of keeping the HEAD version solid enough
to use in production, as most of us running it on a daily
hi
the norwegian company nextgentel uses custom ATAs. does anyone know
these by view?
http://www.nextgentel.no/ressurser/brukerveiledninger/NextPhone.pdf
thanks
roy
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Does anyone know of a speech recognition module (like say yes or no,
or numbers) I guess due to the complexity of speech recognition it
might just be found in commercial applications or am I wrong like
always?
What's wrong with the old and non-fancy IVR?
Voice recognition menus only piss people
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532
There isn't even any code for SIP yet. However the iax integration
works
wonders for a link with just a bit of packet loss and jitter. Voice
conversations are nice and crisp and without the pops associated with
lost
packets or
It's untested and unfinished and touches the core of asterisk. (maybe
causing massive amounts of deadlocks).
It should go into CVS soon. Wasn't there a feature freeze around the
end of february? Does this mean we'll have to wait till 1.4 or
something to get decent sound on SIP?
roy
So? That's what CVS-HEAD is there for.
Adding in experimental patches willy-nilly, especially ones that have
the potential to cause huge problems, confounds attempts to isolate
bugs and test functionality.
Mark does a pretty good job of keeping the HEAD version solid enough
to use in
hi
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
roy
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hi
how can I tune SIP jitter? is it possible today in asterisk?
ryo
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hi
are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
roy
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are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
I'm not sure what studio lite means to you. Maybe hard figures would
be more precise.
G.722 might be interesting : 64 kbps, 7 kHz. It's not free.
Otherwise,
hi
I've been trying to fax digium this agreement for a month or so now
Any chance they can fix their fax?
roy
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how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on
the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams. There are pland for the next
generation jitter buffer
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
v1-0 is the tag used for the latest changes to the stable branch.
Releases are still your best bet, but if you are monitoring the CVS
mailing list for commits to v1-0 stable, then you may see a patch go
in that
Wouldn't it be much simpler and effective to just boot them off the
list? I think they would get the picture pretty quick when they got
back...
They can always check the archives to read up on missed posts, and it
would save us all the trouble in the mean time ;-)
I support this.
Get them off
hi
there are some comments here,
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty,
that people that have earlier offered high bounties for T.38 in
asterisk. Please add up, so the one that one day manages to add good
T.38 support may get something back for it :)
roy
I beleive what you're looking for is a scalable SIP proxy, like SER :)
That way, all clients registers to SER and SER redirects the caller to
one of the asterisk boxes. Search the wiki at voip-info.org for
asterisk at large :)
Yes, that is one of the many pages I've read. But we still have a
I'm trying to stay away from a software based load balancer cause what
happens if that server fails?
Its far less likely for a piece of dedicated hardware to fail than an
actual
computer.
A piece of dedicated hardware runs an OS as well.
I've been running software solutions for virtually
For growth, all you do is add more SER and more Asterisk boxes.
Are you sure one SER box won't be sufficient?
Makes sense to me to have these TWO - you can take one of those
off-line
without interrupting service, and that's the entire idea of this
discussion, isn't it? ;-
Yeah
Get two cisco load
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