Given that he is using plaintext as the auth method, I guess anyone
who wants that
password can have it by snooping anyhow. :-)
T.
On 1 Jun 2009, at 07:18, Rob Hillis wrote:
The clue in the log is "no authority found". Something in the
configuration at the other end doesn't match the config
The clue in the log is "no authority found". Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which t
my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go
to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5
-- Executing [4...@sip:1] Dial("SIP/312-09f9a720",
"IAX2/trun...@147.120.203.98/4567,10,t") in new stack
-- Called trun...@147.120.203.98/4567
[Jun 1 11
Asterisk versions may differ. I do IAX trunk successfully even
between Asterisk 1.0.2 and 1.4.xx
please post your Dial command.
On Fri, May 29, 2009 at 11:33 AM, Tharanga wrote:
> Hi All,
>
> Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and
> asterisk 1.2.14 ?
>
> i tried
Nice to know, luv to have this practical numbers.
On 8/28/07, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I thought I'd give a follow up to this discussion for the archives...
>
> Currently I'm trunking 30 channels of g.729 traffic (no transcoding going
> on, the call comes in and goe
Hi,
I thought I'd give a follow up to this discussion for the archives...
Currently I'm trunking 30 channels of g.729 traffic (no transcoding going
on, the call comes in and goes out as g.729) and it takes about 350 kbps
bandwith bidirectional.
So on average each call takes 11.5 - 12 kbps of
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 tr
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 - while doing IAX2 trunking vs NOT doing it seems to
> half the bandwith requirem
I would suggest to use the branch with the iax bug fixes, i trust it
more than the stable version.
Zoa
Jean-Michel Hiver wrote:
>> 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 s
> 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 th
Jean-Michel Hiver wrote:
>> 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 su
> 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 ab
On 8/24/07, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote:
> 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
I can take 30 calls in one trunk with good voice quality
more calls cause awesome sounds___
--Bandwidth and Colocation provided by Easynews.com --
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To UNSUBSCRIBE or update options visit:
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On 3 Aug 2006, at 09:14, Jon Schøpzinsky wrote:
Hello
Im trying to decide whether or not I want to use IAX2 trunking on
our WRAP based customer computers.
As it only has a 200mhz processor, I want to make shure that the
trunking part does not affect call quality.
Does anybody know if tru
verify property of dev/zap; if your asterisk running in non-root mode;
change /dev/zap chown into asterisk non-root user.
Regards.
Jon Schøpzinsky wrote:
Hello list
We are having some strange problems.
When we setup trunking between two of our servers, the connection only uses
trunking one
On Thu, 2006-02-16 at 14:58 +0200, yusuf wrote:
> also doing IAX2 trunking. What do yuo mean you dont run asterisk out of
> the box. Also want to know what is you bandwith usage for 100 calls and
> g729
I run a modified version of asterisk. There are a few things that I
felt needed to be adde
On Thu, 2006-02-16 at 14:54 +0200, Zoa wrote:
> When you have a lot of calls, try doing a show channels and iax2 trunk
> debug. (they are killers)
>
> Zoa
not having trunks set up yet, I dont do the latter but I do the former
all the time. Mostly becuase this is a new server and I wanted to mak
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
At 100 with g
When you have a lot of calls, try doing a show channels and iax2 trunk
debug. (they are killers)
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 6
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
> The trunks were made to be maximum 60 simultaneous channels iirc.
> I doubt seriously you will be able to do 600 simultaneous on any system.
> (with or without trunking). (at least out of the box).
>
> Zoa
At 100 with g.729 its running 95% idle, in
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote:
I think, but
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote:
> I think, but am not sure, that with a lot of calls inside the trunk,
> some calls seemed to go suddenly go outside of the trunk in one or more
> directions, bursts of error messages appeared on the cli etc.
>
> i didnt investigate it a lot more, my
I think, but am not sure, that with a lot of calls inside the trunk,
some calls seemed to go suddenly go outside of the trunk in one or more
directions, bursts of error messages appeared on the cli etc.
i didnt investigate it a lot more, my problems went away with splitting
them up in smalle
On Thu, 2006-02-16 at 13:38 +0200, Zoa wrote:
> A long time ago i tried to make one big iax2 trunk for one of my
> customers, i soon changed this to several small trunks. (bandwith doesnt
> rise all that much if you use 2 trunks instead of 1.) Asterisk didnt
> seem to like my big trunk very much
A long time ago i tried to make one big iax2 trunk for one of my
customers, i soon changed this to several small trunks. (bandwith doesnt
rise all that much if you use 2 trunks instead of 1.) Asterisk didnt
seem to like my big trunk very much (i don't remember how big it was,
but probably ove
We did a bunch of audio quality tests with slinear some weeks ago, it
seems completely broken. (not limited to iax2 trunking)
It gave us extremely bad results in 30% of all calls, iax2, sip, jb, no
jitter buffer.
Its on my todo list to find out why this happened and file a bug report
if needed.
Adam Robins wrote:
> I have two Asterisk boxes that I thought were trunked, but based on not
> seeing the (T) in iax2 show peers, now I'm not sure.
Make sure you have some form of Zaptel timing (i.e. Digium Cards/ZTDummy)
--
Cheers,
Matt Riddell
___
On Wednesday 21 September 2005 13:52, Adam Robins wrote:
> Should I plug in the actual IP addresses instead of host=dynamic? Also,
> I do not currently have "register" statements.
> In iax.conf for these.
register => each to the other.
-A.
___
--Bandwi
ED] On Behalf Of Matt
Riddell
Sent: Wednesday, September 21, 2005 10:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iax2 trunking wackyness
Andrew Kohlsmith wrote:
> On Wednesday 21 September 2005 07:27, Clive wrote:
>
>>My setu
Andrew Kohlsmith wrote:
> On Wednesday 21 September 2005 07:27, Clive wrote:
>
>>My setup is: telco-asterisk(voip)-asterisk{ITSP}telco
>
>
> Are both your asterisk boxes peered to each other? IIRC trunking ONLY works
> between peers.
If you do iax2 show peers in the console,
On Wednesday 21 September 2005 07:27, Clive wrote:
> My setup is: telco-asterisk(voip)-asterisk{ITSP}telco
Are both your asterisk boxes peered to each other? IIRC trunking ONLY works
between peers.
-A.
___
--Bandwidth and Colocation
On 21 Sep 2005 at 19:48, Matt Riddell wrote:
> Clive wrote:
> > Hi
> >
> > I was doing some bandwidth testing, and my incomming usage is
> > 36% more than my outgoing bandwidth.
>
> In my case the calls come in separately (i.e. untrunked) and get trunked by
> the Asterisk machine and sent out.
Clive wrote:
> Hi
>
> I was doing some bandwidth testing, and my incomming usage is
> 36% more than my outgoing bandwidth.
In my case the calls come in separately (i.e. untrunked) and get trunked by
the Asterisk machine and sent out. This causes an imbalance.
Are your calls coming from many to
Kris Boutilier wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Clive
Sent: Thursday, July 07, 2005 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 Trunking - CVS-Head
Is anyone successfu
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Clive
> Sent: Thursday, July 07, 2005 2:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] IAX2 Trunking - CVS-Head
>
> Is anyone successfully using iax2 trunki
Hi, try this simulation:
http://www.asteriskguru.com/tools/bandwidth_calculator.php
It shows clearly there is a gain starting at the second call.
Zoa
[EMAIL PROTECTED] wrote:
Hi,
I've just setup IAX2 trunking between two Asterisk servers. It seems ok
except for the reduction on bandwidth
On Tue, 2004-11-16 at 15:45 +0100, HÃkan KÃllberg wrote:
> Ohh, sorry, rtc ( real time clock ) I mean...
I don't think zaprtc is more accurate than ztdummy. Iirc it has to do
with the fact that ztdummy doesn't generate the exact same amount of
interrupts as the Digium cards do. The difference in i
On Tue, Nov 16, 2004 at 03:22:04PM +0100, Patrick wrote:
> On Tue, 2004-11-16 at 14:39 +0100, Håkan Källberg wrote:
> [snip]
> > Well, if it works... Compared to the crt package??
>
> Yes it does work. I use ztdummy myself on my hardly loaded home * box
> because I am too lazy to plug my X100P bac
On Tue, 2004-11-16 at 14:39 +0100, HÃkan KÃllberg wrote:
[snip]
> Well, if it works... Compared to the crt package??
Yes it does work. I use ztdummy myself on my hardly loaded home * box
because I am too lazy to plug my X100P back in :) The 3% off will have a
negative impact when you start to get
On Tue, Nov 16, 2004 at 01:56:07PM +0100, Patrick wrote:
> I don't think you need to do anything with zaptel.conf when you are
> using ztdummy. Someone on irc mentioned that ztdummy's timing is off by
> about 3%. So if you are serious about your timing why don't you get a
> X100P to make sure you h
On Mon, 2004-11-15 at 21:03 +0100, HÃkan KÃllberg wrote:
[snip]
> >I don't know how to configure zaptel ( /etc/zaptel.conf )
I don't think you need to do anything with zaptel.conf when you are
using ztdummy. Someone on irc mentioned that ztdummy's timing is off by
about 3%. So if you are serious a
On Mon, Nov 15, 2004 at 10:02:43AM -0800, Bob Knight wrote:
> Håkan Källberg wrote:
> >I want to trunk two Asterisk systems with each other. System A,
> >behind a NAT-Firewall and System B with a real IP address.
> >
> >aix.conf on B:
> >
> >[mytrunk]
> >host=dynamic
> >username=mytrunk
> >auth=md5
Håkan Källberg wrote:
Hello!
Perhaps someone can spread i little bit light on this:
I want to trunk two Asterisk systems with each other. System A,
behind a NAT-Firewall and System B with a real IP address.
aix.conf on B:
[mytrunk]
host=dynamic
username=mytrunk
auth=md5
secret=yyy
trunk=yes
iax.con
L PROTECTED] On Behalf Of David Cook
> Sent: Tuesday, June 22, 2004 5:48 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] IAX2 Trunking help!
>
> I was just trying to solve this one myself. I found this
> method worked for me. I'm still calling this "Method 1&q
On Tue, 2004-06-22 at 10:20, David Cook wrote:
> So you're saying that the following would be the same?
>
> iax.conf
> [YOUR_REC_SERVER]
> secret=mysecret
> host=my.receiving.server.ca
> context=local
>
> extensions.conf
> exten => _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
>
> If so, what abou
Perfect! Thanks for the clarification. That's what my brain needed - on
both points.
dbc.
Quoting Kevin Walsh <[EMAIL PROTECTED]>:
> If that's on your outgoing side then you'll also need "type = peer"
> in there. The incoming side would have "type = user".
>
> Outgoing = peer, incoming = user.
David Cook [EMAIL PROTECTED] wrote:
> So you're saying that the following would be the same?
>
> iax.conf
> [YOUR_REC_SERVER]
> secret=mysecret
> host=my.receiving.server.ca
> context=local
>
> extensions.conf
> exten => _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
>
> If so, what about the type=
On Tue, 22 Jun 2004, Tony Nichols wrote:
> I'm trying to get two * boxes to talk no matter what variation I try
> I get No Authority Found and connection refused from 192.168.1.5
>
> I've googled, I've site searched to no avail.
I think you need to match a peer at one end to a user at
On Tuesday 22 June 2004 10:08, Kevin Walsh wrote:
> You really don't want your username and password to appear (in plain
> text) in your logs.
> Put the sensitive details in iax.conf instead of extensions.conf.
> As well as being more secure, it'll make your Dial() string shorter,
> and will mean
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten => _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the type=peer/user/friend thing? I did read the docs
but maybe I'm th
David Cook [EMAIL PROTECTED] wrote:
> [mycontext]
> exten =>
> _5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED])
> exten => _5XXX,2,Hangup exten => _5XXX,102,Hangup
>
You really don't want your username and password to appear (in plain
text) in your logs.
Put the sensitive details
Tony Nichols [EMAIL PROTECTED] wrote:
> I'm trying to get two * boxes to talk no matter what variation I try
> I get No Authority Found and connection refused from 192.168.1.5
>
> [snip]
>
> Server b config (192.168.2.2):
>
> [pbx]
> type=peer
> host=dynamic
> trunk=yes
> secret=test
> quali
I was just trying to solve this one myself. I found this method worked
for me. I'm still calling this "Method 1" in my document because I don't
fully understand the "switch" and the "register" versions and pros/cons
to implementation of each. But this one does work.
Method 1
Receiving Server
Iax.c
I seem to have the same problem now,
were you able to resolve this ?
joachim.
At 22:41 6/11/2003 -0500, you wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot fi
Thorsten Lockert wrote:
And, given that a context= entry is not used for a "peer", only for a
"user", it also goes to say that if you use a "friend", context is still only used
for the *inbound* portion of it. So I just don't really see why you so
strongly recommend against "friend" entries as op
> No, you actually don't need to use a context in the peer. Asterisk will
? leave it up to the far end to decide what context to use.
> We use it to avoid any possibility of confusion in the process, but it
> is not necessary.
>
> In fact, I just verified this with the master himself and we will
TeleSIP wrote:
Mark's words to me, when I was a newbie:
[00:08] a user is to authenticate an incoming call
[00:08] a peer is someone you send a call to
[00:08] friend, of course, is both
I am still at a loss here. If both are set to peer then how can either end
originate the call? You wo
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, October 09, 2003 2:57 PM
Subject: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2
Trunking confirmation?)
> WipeOut wrote:
>
Thorsten Lockert wrote:
and the peer simply has the required information:
[NuFone]
type=peer
secret=his_secret
context=NANPA
host=switch-1.nufone.net
Uh. Why would you want to specify a context for a peer at all...? Aren't
those used
only for inbound anyhow?
No, you actually d
>On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote:
> > > and the peer simply has the required information:
> > >
> > > [NuFone]
> > > type=peer
> > > secret=his_secret
> > > context=NANPA
> > > host=switch-1.nufone.net
> >
> > Uh. Why would you want to specify a context for a peer a
On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote:
> > and the peer simply has the required information:
> >
> > [NuFone]
> > type=peer
> > secret=his_secret
> > context=NANPA
> > host=switch-1.nufone.net
>
> Uh. Why would you want to specify a context for a peer at all...? Aren't
>
Jeremy McNamara wrote:
WipeOut wrote:
Jeremy, Can you elaborate on how using type=friend would restrict the
dialplan.. Just so I am aware of the pitfalls.. :)
Mark's words to me, when I was a newbie: [00:08] a user is to
authenticate an incoming call
[00:08] a peer is someone you send a cal
> and the peer simply has the required information:
>
> [NuFone]
> type=peer
> secret=his_secret
> context=NANPA
> host=switch-1.nufone.net
Uh. Why would you want to specify a context for a peer at all...? Aren't
those used
only for inbound anyhow?
Thorsten
WipeOut wrote:
Jeremy, Can you elaborate on how using type=friend would restrict the
dialplan.. Just so I am aware of the pitfalls.. :)
Mark's words to me, when I was a newbie:
[00:08] a user is to authenticate an incoming call
[00:08] a peer is someone you send a call to
[00:08] friend, of c
tcpdump is the easiest way. From 1 call to 50 calls the number of packets
should be about the same, and they should just get larger.
Mark
On Thu, 9 Oct 2003, Jared Smith wrote:
> On Thu, 2003-10-09 at 11:39, WipeOut wrote:
> [snip]
> > He states that in order for "trunking" to work the type has
TeleSIP wrote:
Hi Jeremy,
The handbook says:
"user: A user can place calls to or through the Asterisk server.
peer: A peer receives calls from the Asterisk server, but does not
place them
friend: A friend both sends and receives calls through the Asterisk
server. This makes the most sense for
Jared Smith wrote:
I think you may be confused as to what the "trunking" is. Just because
you can make calls over IAX doesn't necessarily mean you have trunking
working. (Trunking combines packets from multiple calls to reduce
overhead.)
Jared Smith
I do understand what trunking is, I was ju
John Todd wrote:
Hi,
My question is in refernece to the posting by Jeremy McNamara here..
http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html
He states that in order for "trunking" to work the type has to be
peer.. When I set mine up I did so using type=friend just to ma
Jeremy McNamara wrote:
A friend is both a user and peer. However, I would discurage the use
of a friend as it will severely restrict your dialplan, espcially once
you are dealing with more than just a couple Asterisk boxes.
Jeremy, Can you elaborate on how using type=friend would restrict the
se?
Thanks,
Ricardo
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, October 09, 2003 12:48 PM
Subject: Re: [Asterisk-Users] IAX2 Trunking confirmation?
> From the chan_iax2 source (around line 3712):
>
>
Hi,
My question is in refernece to the posting by Jeremy McNamara here..
http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html
He states that in order for "trunking" to work the type has to be
peer.. When I set mine up I did so using type=friend just to make it
simple.. So
On Thu, 2003-10-09 at 11:39, WipeOut wrote:
[snip]
> He states that in order for "trunking" to work the type has to be peer..
> When I set mine up I did so using type=friend just to make it simple..
> So when I read the above posting I thought well maybe my "trunking" has
> not been working prop
From the chan_iax2 source (around line 3712):
if (!peer) {
ast_log(LOG_WARNING, "Unable to accept trunked packet from '%s:%d':
No matching peer\n", intoa(sin.sin_addr), ntohs(sin.sin_port));
return 1;
}
A friend is both a user and peer. However, I would discurage the use of
a friend as i
>- The number for "Estimated IP Overhead" was obtained by
subtracting (additional channel usage) from (single channel usage.)
This is possibly inaccurate.
It should generally be pretty accurate. You might try running 3 calls
just to confirm.
I'll try this shortly after I return from dinner.
>- The number for "Estimated IP Overhead" was obtained by
> subtracting (additional channel usage) from (single channel usage.)
> This is possibly inaccurate.
It should generally be pretty accurate. You might try running 3 calls
just to confirm.
> ILBC:
> one call: 56134.91 bps/67.45 pps
> I think it's only being tested with GSM right now, but I'm not aware of
> any reason why you couldn't use another codec. Maybe Mark can enlighten
> us with some details?!? (Thanks Mark for implementing this! I know I'll
> use it a LOT!)
In principle it will work with any choice of codec suppor
On Mon, 2003-03-17 at 11:00, John Harragin wrote:
> >I think it's only being tested with GSM right now, but I'm not aware of
> any reason why you couldn't use another codec. Maybe Mark can enlighten
> us with some details?!? (Thanks Mark for implementing this! I know I'll
> use it a LOT!)
>
> Th
Many of the hardware-based solutions I've seen use G723 and G729... I'm
hoping to find a smaller codec than GSM that doesn't require *too much*
CPU. (I'd like to use something like speex, but rumor has it that the
CPU requirements are way out of my league.)
Jared
On Mon, 2003-03-17 at 10:00, Joh
>I think it's only being tested with GSM right now, but I'm not aware of
any reason why you couldn't use another codec. Maybe Mark can enlighten
us with some details?!? (Thanks Mark for implementing this! I know I'll
use it a LOT!)
This brings up another question. As long as T1s have been around
I think it's only being tested with GSM right now, but I'm not aware of
any reason why you couldn't use another codec. Maybe Mark can enlighten
us with some details?!? (Thanks Mark for implementing this! I know I'll
use it a LOT!)
Jared
On Mon, 2003-03-17 at 03:04, Roy Sigurd Karlsbakk wrote:
>
On Sunday 16 March 2003 21:17, Mark Spencer wrote:
> IAX2 now has support for a "trunk" mode ("trunk=yes" in the appropriate
> friend section). Trunk mode allows IAX2 to use bandwidth extremely
> effectively. The original impetice (and strategy) was a result of a
> mistake in which it was claimed
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