* SIP isn't a standard. It could be made into an official standard,
if there was a standards document. Someone should write one, and
start an IETF working-group. If the IETF adopted it, there would be
wider acceptance.
True it isn't. However it has a written spec (I have not seen such a
Forgive my lack of depth in this area, but aren't SSL based VPNs
fundamentally IP centric? Whereas RTP IAX2 streams are UDP? I had
read some time ago about a company that was planning to revolutionize
voip through SSL based VPNs, they met with much scorn from those who I
thought were
Actually, not a perfect example:
In this case there was exactly one poster who answered on-list to the
original post. The answer was wrong list. The original poster insisted
then on re-asking the question, and then the flames really started.
Which is why I think the way of doing it is
(obviously if you do other magic in your dialplan this needs to be adjusted.
The important part is the 'g' flag to Dial (go on after hangup), and the NoOp
which echos the dialstatus and hangupcause variables to the console.
How would you do this in an AGI script? Basically what I have at
Roy Sigurd Karlsbakk wrote:
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
Hi List,
I am writing some basic LCR (Least Cost Routing) AGI script for Asterisk
written in Perl which I intend to GPL when it's finished and properly
documented.
In order to have a way to compare rates, I'd like to have some 'rule of
thumb' rule - which will be overridable - and which
Matt wrote:
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
I think that's what you'd have to do.
Also, is it possible to put a phone
Other than Broadvoice, are there any VoIP providers (Vonage, Packet8,
etc) that can be hooked into Asterisk directly? I read about a scheme
for Packet8 that involved routing it in through an analog connection
on a FXO port...I'd rather have something I can connect in directly.
Save yourself
However, I don't know the specific requirements for the T1 line or how to split the data and voice.
The point of VoIP is to consolidate data and voice onto one network.
Combining both allows for economies of scale:
* you don't have to use sangoma or digium card, this is the VoIP
provider's
Ummm, in an office of x people, where x is some arbitrary integer 1,
is 1 PSTN line for emergency services sufficient ?? Personally, I would
think it isn't, but haven't quite determined what number is sufficient.
Consider, an office of 50 people, and a small fire breaks out, how many
people will
Really?? What if you happen to be the 4th person calling, and need to
inform them that you are trapped in the stationary cupboard, and luckily
you were carrying the cordless handset (but not your mobile phone)??
At the end of the day, I wouldn't expect any office to have a 1:1 ratio
between
Sys Admin wrote:
After 20 posts, in 2005 the ideal setup for a new installtion of a 50
user asterisk is:
Option1: IAX2 with softphone firefly
Option2: SIP with softphone
Option3: IAX2 with hardphones (which brand?)
Option4: SIP with hardphones.
Seems like we cannot come to a definite conclusion,
If the IAXy had a bit more work done on it, it could be a good option,
but it's not at the current time.
Yep! Things like:
- more codecs (just ulaw? come on...)
- proper DHCP and possibility of static IP
- a 'reset' button
To start with would be nice to have.
And my IAXy doesn't work with my
Ed Greenberg wrote:
Anybody using a Sipura 3000 for FXO with Asterisk?
Mine is working except for one small nit...
When a call comes in from the PSTN, the Sipura answers it and then
passes it on to Asterisk, which plays extension ring tone.
I'd prefer for the POTS line to stay on-hook while the
Richard Reina wrote:
I installed the AGI perl library then put the
following script in a file called
/var/lib/asterisk/agi-bin/send_clid.agi,
updated my [incoming] context with exten =
s,1,AGI(send_clid.agi) and did a restart now.
use Asterisk::AGI;
my $agi = Asterisk::AGI-new();
my %input =
My question is: is it possible to have repeated users on sip.conf
being identified by their different passwords? I tried to do that but
got an authentication failure. Is there a way to do this? Or I should
always have different usernames?
I think it would be less crazy best if you had a naming
Chuck Bunn wrote:
Hi,
I am new to Asterisk and the first thing I have noticed about Asterisk
and Pingtels open PBX's is that they are using this dinosaur method of
running forums. It is a real pain getting every message in the forum
and essentially
snipped nonsense
The one thing I like with
[EMAIL PROTECTED] wrote:
Any decent on-line forum would be much better than these digium email lists.
The lists are
poorly formatted,
They look fine in my mail client.
there is no easy way to post code
Yes there is. Cut, paste, indent. Or if it's too big, put it up
somewhere and paste in a
I've noticed that nufone returns 'circuit busy' messages FAST (when it
does) while this tends to take a while with VoIPJet.
I've also seen 'circuit fast busy' message - what is the difference
between the two?
Thanks,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5
Or, better yet, Digium should shut this list down and move it to a
commercial vBulletin style forum and get some good moderators to delete
posts that do not follow a basic set of social rules of behavior. Here are
the rules from UNIX.COM, and they work very well:
Because *you* think forums
Irakli Natsvlishvili wrote:
I don't know following has debated here or not, but is there in this
world following stuff:
I think you want a Soekris.
Cheers,
--
Ykoz Un Max - La VoIP en pr-pay!
Essayez gratuitement - 5 crdits offerts.
--- http://ykoz.net/voip/max ---
YE HAA.. We 'be out har' in rural internet space, mamma
Stepping in cow pats . Way out past the fancy, hi falutin',
city-slicker, collaborative tools of 2005 ..
You babble all this nonsense, and *YOU ARE* the guy advocating moderation?
BWAHAHAHAHAHAHAHAHAAA!
I think I can use Asterisk for this issue. I have installed it and I
have run it, but I don't know how I have to configure it.
You should go and read some docs:
http://www.digium.com/handbook-draft.pdf
I have read the documentation, but It's so much big and I don't know
what I have to do.
You are the bully. So far the majority wish the email list to
continue and yet you still continue to demand that Digium
convert to a forum.
Looks like M. Bass likes to troll about mailing lists, see this post:
http://info.ccone.at/INFO/Mail-Archives/procmail/Feb-2003/msg00230.html
As usual,
Kerry Garrison wrote:
The book bills itself as a beginner's guide to Asterisk and Voice over
IP (VoIP). Even with over 270 pages, it isn't possible to go through
every single feature that Asterisk has to offer but the book does give
enough information to get you started and even apply a few
1. Support for message threads - replies to messages are shown right
below the original message.
Any decent mailer has that functionality.
2. Support for subject matter sub forums - different message
categories can be established.
You just split the list into differents mailing lists and
Lee Lee wrote:
Hi everyone
Presently all our calls are channel to one provider and we would like
to change that based on LCR.
the following is what we have presently;
# Dial the requested number, if we got something from the caller.
if ($dialto != )
{
$AGI-exec('SetAccount',
Rod Bacon wrote:
Sorry if this is off-topic, but I know there's a quite a few smart
people who frequent these groups, and I was thinking that it'd be a
good place to ask.
We have an opening for an experienced PERL programmer. If you (or
anyone you know) is interested, please feel free to email
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users mailing
list? This breaks the reply to sender only / reply to all / list reply
functionality of my mailer. It's really broken :(
Cheers,
--
Ykoz Un Max - La VoIP en pré-payé
Oops, sorry for the list reply :/
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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at the docs on this page:
http://ykoz.net/intl/lcr/
There's no mailing list for this yet. If anybody is interested in the
project, setting one up for me would be nice :)
Best Regards,
Jean-Michel Hiver.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip
It seems Petal.pm (maybe a part of FreezeThaw???) is missing on my
system. Where can I find it?
It's a 'cut and paste' from another script, and it's a mistake. I'll
remove the dependency.
Meanwhile, you can 'fix' the issue by installing the module:
perl -MCPAN -e 'install Petal'
Alternatively,
I put the Who? in Mishehu wrote:
It is alright to sell hardware, and it is alright to sell labor when
dealing with open source software. But selling licensing on something
that does not exist (extension licensing???) is wrong.
No, it's not wrong as long as the code remains GPL'ed and you don't
Ronald Wiplinger wrote:
I am not sure how many licenses of G729 I need to purchase from Digium.
I have a TDM22 card.
Do I need for each FXS (2) or each FXO (2) or for both the license?
Other SIP phones do have the license already, am I right here?
You need a license whenever Asterisk is
Seth Ueland Chancy wrote:
What are the known distributions of Linux with which Asterisk is known
not to work?
To answer the real question which is on the back of your head, unless
you're lucky you'll probably have to do a lot of fiddling around no
matter which distro you choose to get * to
I was wondering what could be pros and cons of ztdummy vs proper timer
device (i.e. X100P).
I am going to set up an asterisk server in europe (to do trunking, to
save bandwith) and I was wondering if it'll be OK to get it going with
ztdummy.
Furthermore, I have only a 1024/256kbps PPPOE DSL
I would seriously doubt that you can actually squeeze 12 channels through
that dsl and obtain anything reasonable for quality, regardless of which
asterisk codec you choose. But, it certainly would not be that hard to
test it and validate assumptions.
It would be wiser to know that it has a
So, 215 - 40 - 110 = 65kbps
You only have about 65kbps to spare, and all of this is based on ideal
(theoritical) conditions. I doubt that those 12 calls will sound
okay, or even work at all...
But, you can always try!
The thing is: how do I do that? What tools are there to test how many
Jeromie Reeves wrote:
I have been looking around for VoIP providers but have not found a
good listing.
Is there no yellow pages for VoIP providers? Google mostly returns
services
like Vonage, Packet8, NuFone, ect. None seam to be very reseller
friendly and
none offer LNP or local DID's for my
Daniel Nyström wrote:
Do anyone knows abount European resellers of these products:
* Digium Wildcard TE410P
* CarrierAccess Adit 600
Preferably in Sweden, but Europe is also better.
Have anyone within EU ordered products from these companies directly from the US?
In that case, how is the service
pstn - gw - asterisk
|
phone
can you recommend me some hardware (cheapest than PC+fxo card+asterisk)?
That's the kind of stuff the sipura 3k really shines at.
It offers one FXS, one FXO, 2 VoIP channels and decent routing
capabilities for about $100.
I've never managed to get echo
Stig Thune wrote:
Searching through wiki and google.
http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
but there are also other products on the market.
---
Wondering if its possible to connect as follows:
Extension - Asterisk - ZyxelAnalogTelephoneAdapter - GSM gateway.
If I
Michael Giagnocavo wrote:
Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
only game in town. I know that the Farfon device will be out soon and
we'll be able to judge its quality at that time.
Or any PA168 phones, which are already out, and support IAX2, SIP, H323,
MGCP
marius baranescu wrote:
Hi ,
I have a running Asterisk box . It is running great
My problem is that I can not get connected to the world :) .
Well, the sensible option then is to rent a cheap server somewhere with
static IP and do VPN / Tunneling.
My only option available here is a satellite
I am trying to get a SPA-3000 to work behind NAT - for the sake of the
exercice.
The SPA is on the local network at the address 192.168.0.125 behind a
NATted linux router.
The machine I am trying to work with is a friend's (let's call it
lolo.dyndns.org) and I've installed Asterisk 1.0.3 on
Nabeel Jafferali wrote:
I am trying to get a SPA-3000 to work behind NAT - for the
sake of the exercice.
Post the relevant entries from sip.conf and extensions.conf, and the
relevant fields from the SPA-3000 Line 1 tab.
Sure!
From sip.conf
=
[2001]
type=friend ; This
Right now our only solution is to buy $5000 worth of XTen Pro; then we get
co-branding and 729. But if we can save $5000, all the better.
I've read somewhere on ZdNet that this 'wengo' french SIP provider was
gonna open source their client, might be worth sending them an email -
it might be
[EMAIL PROTECTED] wrote:
I've created a pretty complete HOWTO on creating a Linux Bridge (using
Fedora) to shape LAN -- WAN traffic. It includes installation
instructions, a script to configure the bridge (which you install as a
service), and 2 scripts to configure the network interfaces using
Hi Guys,
After days of fiddling, I can't really get my SIP device to work
communicate with Asterisk behind NAT. Sometimes the STUN server is
flaky, sometimes the device isn't reachable if the connection is dropped
and then put back on, sometimes it registers OK, sometimes it doesn't, etc.
I've
If you're going to promote your own company, that is fine, but you
should do it in the -biz list.
If you're going to try hard to sound like you are an objective third
party, next time try to remember to set your email address properly.
At least this kind of astroturfing brings some sample
Ken D'Ambrosio wrote:
Howdy! I'm VERY interested in your HOWTO... but the link you have,
below, times out. Any chance you could mail me the HOWTO, or point me
to a new link?
Well, linux bridging is *really* easy, here is what I have on my box
(eth0 goes to the LAN, eth1 to the netgear modem).
Jer wrote:
Dear list
I need to find a way to hook up 4 analog phones to *
was wondering if anyone had old hardware not being used and they are
willing to sell
or know of any places to buy that are cheaper then digium :/
You could always get a couple of SIPURA 2000 (SIP devices that can talk
to
Hi List,
After some research, it seems the only reasonable thing to do in order
to get SIP phones behind NAT working reasonably well without fiddling
with the DSL router is to have some kind of far end nat traversal mechanism.
Is there any way to do this with open source tools? I've seen
Yair Hakak wrote:
Hello,
i'm using ser+nathelper+rtpproxy in front of asterisk. It has been
terrific.
This sounds pretty cool... could you share some config files, maybe
stick them on the wiki somewhere? Or if you want you can send them to me
privately.
If I manage to get it to work thanks to
[EMAIL PROTECTED] wrote:
The IAX protocol gets around all NATn?
It's much more NAT friendly than SIP.
You know I can't believe that something that such a new, big standard as
SIP doesn't account for all environments.
Roughly, at the moment, this is the situation *to my knowledge*. If I'm
wrong
Leo Ann Boon wrote:
Another question... Are you aware of a SIP ATA or phone that has some
kind of VPN (i.e. PPTP) client embedded in? This would make the NAT
problem go away nicely and provide added security...
The Zulty's phones support VPN. Then again, many firewalls don't pass
through VPN
Hi List,
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
Also it seems that VoIPJet takes forever to return 'circuit busy' while
NuFone does it instantly.
At any rate, is there like a reliable third VoIP provider I can use for
fallback when the two
asterisk wrote:
Assuming I'm using a VOIP provider of some sort, what kind of
bandwidth requirements / line should I expect to have in place? I
currently have 8 traditional voice lines, and a FAX line that doubles
as my DSL source. Ballpark, what do I need to have in place to move
everything
Hi List,
I'm wondering if anybody on the list managed to get one of these beasts
working with asterisk?
FYI They're Windows NT embedded (yuk!) based H.323 / SIP compliant
devices with a *very* complicated admin interface. Can't figure out how
to get it working... yet :)
Cheers,
Jean-Michel.
John Goerzen wrote:
I've heard good things about VoipJet here, so I was going to set up an
account. Then I noticed their Terms of Service here:
https://www.voipjet.com/tos.php
Several things there are very concerning to me, and I'm interested in
what other people here think of them.
* The ToS
nkb wrote:
Hi.
I'm really new.
I was just wondering if it is possible at all to do a IP to IP call
without a * server (or as a matter of fact, any other kind of server)?
say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at
hisdomain.com's 192.168.0.3. Is this sort of things
Tomasz Chmielewski wrote:
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my
PC-Asterisk server (which I yet have to install) and use them as 4
lines in case anyone has to call me in
From what I have been told on this very list you can only use Diva
Server cards with asterisk because the 'cheaper' diva cards do not
support some stuff called 'capi'.
Or off course you can buy digium cards. They look pretty cool anyway -
can't wait to receive the onces I have ordered :-)
It's a very interesting idea. The more I think about it, the more I
wonder . . .
Sounds like something that would end up being used as a dating phone
service to me :-)
Cheers,
Jean-Michel.
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Corvin wrote:
Hi all,
I can see huge traffic here over 400 post in 4 days.
My proposal is to create asterisk newsgrup proposal or phpBB forum
what do think about it ?
I think it's a terrible idea :-/
Sorry if this seems crude to some people, but mailing list have this
wonderful property of
I have had this book for a while and I must say I'm very disapointed
with it.
It's presented as a technical overview of the open source PBX but IMHO
it's more like a collection of howtos with rather confusing examples.
It has about zero structure and doesn't really help you understand how
Lee wrote:
I apoligize in advance for this newbie question on what I perceive as
a mostly advanced level list... I did some searching, but would like
some of your expert opinions.
I'm certainly no expert but I'll give it a shot anyway :-)
2. What is a good, inexpensive FXS solution?
I simply
Hi List,
I have set up the following in my extensions.conf
; local numbers look like 0262XX
; but must be dialed 262 262XX
exten = _0262XX,1,Dial,IAX2/[EMAIL PROTECTED]/011262262${EXTEN:4}
exten = _0262XX,2,Dial,IAX2/[EMAIL PROTECTED]/011262262${EXTEN:4}
exten =
Hi List,
I know this a little outside asterisk but I couldn't resist.
I wrote just a little perl toy I wrote to find words in existing phone
numbers. For example, using the script on Digium's toll free line, you
discover that it can be 877-LINUX-ME but also 8775-GNU-WOE or
8775-HOT-ZOE ;-)
The X100P is working - partly. I can make outgoing calls. But the
card has got a problem detecting incoming calls. Even in verbose mode
I don't see any hint that the card detects a call.
Now it works. I changed the following items in the file wcfxo.c:
#define PEGTIME 1000 * 8
#define PEGCOUNT
Adam Goryachev wrote:
Hi all,
I have the opportunity to demo asterisk to a large group of people, and
was just wandering *how* to do that?
I've been setting * at home just to train myself with it. Here is what I
have:
- IVR menu
- music on hold / transfer
- voicemail
- transparent Zap or IAX
So I thought of installing a combination of four pci cards in the
machine, and everybody on the list just keeps telling me it won't work.
You have 5 POTS lines and 4 X100P cards? Sounds like a complete drag...
At any rate, why don't you buy a TDM400P with 4 FXO ports? I've bought
one off
Jim Guy wrote:
Hello,
I am just starting to research Asterisk and I would like to install it
on a PC to try out. I have looked around quite a bit but I haven't
found much information on the Linux part. I know you need to put Linux
on the PC first but what version or flavor of Linux do you
What leds are lit?
Looking with the orange bit facing you, the network led on the left
(network) is permanently lit. The led on the right blinks once every 7
seconds or so. There is also the network plug's led which is lit. That's
all.
What kind of phone is connected to it?
France
Dorn Hetzel wrote:
Would anyone care to offer opinions as to the FXO interface which sucks
the least :)
So far, for me, using VoIP - PSTN termination provider has been the
solution which sucked the least.
My FXO card doesn't seem to work so well. Never tried my SIPURA as an
FXO device though...
Hi List,
I was wondering if there was any device I could use to connect * to GSM
networks. I don't need much capacity, maybe 2-4 GSM channels. As usual,
cheap is better :-)
Any tips on this?
Cheers,
Jean-Michel.
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What applications (osx or linux) are best? Optimal settings?
linux 'grip' is very nice.
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Hi List,
I have this good looking IAXy device... I have managed to provision it,
i can see it registering to my asterisk box, however when I pick up the
phone which is plugged in the IAXy I have no dialtone, nothing.
Any ideas what might be going on?
Cheers,
Jean-Michel.
Hi List,
I've got * randomly hanging up on inbound or outbound calls on zap
channels. I use a Digitnetworks X100P clone card. Any idea of what might
be happening?
Cheers,
Jean-Michel.
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Hi List,
I have installed a debian system through a knoppix CD and the knx2hd
tool. I am using 2.6.7 kernel because 2.4 doesn't seem to support the
network card I have in the box - which is a drag: from what I have read
so far asterisk works better with 2.4.
After doing a apt-get update
Dave Cotton wrote:
On Fri, 2004-10-29 at 10:48 +0400, Jean-Michel Hiver wrote:
Now I don't have any digium hardware in this box, so I wanted to use the
ztdummy driver before starting asterisk. However 'modprobe ztdummy'
tells me that module ztdummy is not found.
Is there a way to install
You'll need to get the kernel source for 2.6.7.
apt-get install kernel-source.2.6.7
John, thanks for that.
Which version of Asterisk are you using? head or stable?
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This is how I do it - I know the 2.6 kernel is supposed to have an
easier way, but I've not seen/read how to do it yet.
That did it for CVS head on a knoppix distro. Thanks!
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Hi List,
I have managed to compile asterisk but I can't start it. What I have
done so far as asterisk config is concerned is cut and paste the sample
config files from the ONLamp article on Asterisk.
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
When I start asterisk -vvvp I get
Thanks for the tip! I'm still having a couple of quirks though...
Adjust devices= with the number of B channels supported by your card. For
ISDN BRI, it's 2, for PRI, it's 30.
Okay, I did that but then I had the exact error you describe below...
You need a kernel support for you card and you
It's not enough, you must compile the correct Eicon driver.
Read /usr/src/linux/Documentation/isdn/README.eicon
Okay... Well, since my goal is to get asterisk to somehow work, I have
removed the card from the box. To my surprise, I still have the same
error! I have tried re-compiling
This card does not have CAPI drivers. Only the Eicon Diva SERVER cards
have capi drivers.
Fine... I have removed the card from the box anyway since my current
goal is to get asterisk to start.
Still, asterisk still moans the following when I start it with asterisk
-vvvp
[chan_capi.so] =
In modules.conf put
noload=chan_capi.so
and any other module that gets complained about.
Hurray! Asterisk now actually starts! I had to disable quite a few
modules though:
[skipping chan_capi.so]
[skipping app_capiCD.so]
[skipping app_capiHOLD.so]
[skipping app_capiRETRIEVE.so]
[skipping
If you refer to the urban legend that IAX always needs a server to
stay in the media path, then you would be wrong. IAX has a mechanism
that for all practical purposes is equivalent to a SIP reinvite
through which the end points then transition to a mode by which they
communicate directly peer to
Actually, I assume the above (2 x IAX devices behind a single NAT
router) would work perfectly without any special configuration EXCEPT in
the (perhaps most common case) where both IAX devices are talking to the
same IAX server.
Could you explain why it would be a problem if both devices were
Hi List,
I am trying to get my budget sip phone to work with asterisk, which in
turn is configured to work with NuFone. I can get the phone to ring my
home PSTN'ed phone but as soon as I pick up my home phone it hangs.
Here's what I get in the log:
Nov 4 18:37:44 WARNING[1191013296]:
Hi,
For starters, I was hoping that some of the experts on this board could
give me some tips on what I need to do to allow one phone to successfully
call the other phone. I did a similar thing several years ago using a SIP
proxy server (from Dynamicsoft, albeit, with help from their support
iax.conf:
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
sip.conf:
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
I think that should do what you need. Please read up on codecs at
http://www.voip-info.org/wiki-Asterisk+codecs
Wow, it works! And it works great too.
I am quite impressed really. I have
Hi List,
I have a 2Mbps SDSL link which gets saturated during peak time because
about I have about 3 E1 worth of g729 traffic going thru. So I'm planning
to use IAX2 trunking to reduce bandwith requirement and squeeze out each
and every bit of this (expensive) bandwith.
I've set up two
Le Fri, 24 Aug 2007 20:50:05 +0400, Mark Quitoriano
[EMAIL PROTECTED] a écrit:
What is a good softswitch that is also open source rather than asterisk?
You may want to check out freeswitch.
___
--Bandwidth and Colocation Provided by
So you are using an asterisk box as an E1 gateway. You want to know if
switching from not using IAX trunking to using IAX trunking will have
any effect? Yes it will lower your bandwidth usage a little. It
will not increase the CPU load. If your system can support x calls it
will be able
I used to do it, but its a while ago. (Before iax2 got some more fixes)
The trick was to keep the trunks small (like 40 per trunk, just make
multiple), this should no longer be needed.
Cpu utilisation with trunking should be lower than without trunking.
Hi Zoa,
Thanks for your input. I think
Le Sun, 26 Aug 2007 20:20:01 +0400, Andrew Joakimsen [EMAIL PROTECTED]
a écrit:
On 8/25/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
I'm already receiving the calls as g.729, so there is little gain
(slightly less bandwith usage, slighly worse sound) in doing g.729 -
g.723 transcoding
of bandwith. The solution
seems stable and the QoS is identical... so for the price (2 commodity
PCs...), IAX2 trunking is well worth the effort since it reduces bandwith
usage by a factor of 2.
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - YKOZ
+262 (0)692 828 070
I have done some testing with VOIP provider though my firewall to FWD and
VOIPSTUNT.
Where might SER help?
Why are people using it with Asterisk?
SER's SIP implementation is very stable. It will handle a lot more
phones than asterisk (but does a lot less than Asterisk too).
I use SER to
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