On 12-07-15 10:23 PM, Klaverstyn, David C wrote:
Hi All,
I need some assistance with QoS. We have multiple Asterisk servers on
a MPLS network. We have just moved over to Verizon and for us to get
QoS Verizon are saying we need to use af41.
Rather than get the application to do this, get
Hi All,
I need some assistance with QoS. We have multiple Asterisk servers on a MPLS
network. We have just moved over to Verizon and for us to get QoS Verizon are
saying we need to use af41.
I need to check what exactly I need to do as this is a new area for me.
We only use IAX over our WAN
On 07/15/2010 11:13 AM, hin lee wrote:
> I have discussed QoS with our ISP and in order to implement this, I
> need to make sure all VoIP packets are marked in the IP packet header
> (IPP bits?). Does Asterisk automatically marks the VoIP packets or
> do I need to do something in Asterisk? I ne
I have discussed QoS with our ISP and in order to implement this, I need to
make
sure all VoIP packets are marked in the IP packet header (IPP bits?). Does
Asterisk automatically marks the VoIP packets or do I need to do something in
Asterisk? I need to do this for SIP and H323 protocols.
PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] QOS/DSCP for IAX?
>
> Michelle Dupuis wrote:
>> I actually see the TOS setting in iax.conf, but the default (commented
>> out) is EF - which doesn't even match a valid bit combination
>> according t
ts.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, October 01, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] QOS/DSCP for IAX?
Michelle Dupuis wrote:
> I actually see the TOS setting in iax.conf, but the default (commented
>
> an ADSL connection?
>
> _
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
> Dupuis
> Sent: Thursday, October 01, 2009 2:27 PM
> To: Asterisk Users List
> Subject: [asterisk-users] QOS/D
erisk Users List'
Subject: Re: [asterisk-users] QOS/DSCP for IAX?
I actually see the TOS setting in iax.conf, but the default (commented out)
is EF - which doesn't even match a valid bit combination according to
voip-info wiki
If this is the right place, what TOS value are people us
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 01, 2009 2:27 PM
To: Asterisk Users List
Subject: [asterisk-users] QOS/DSCP for IAX?
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.con
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.conf but not iax.conf
Thanks
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http:
At 06:37 AM 7/15/2009, you wrote:
>Ours is just internal, but the concept should be the same. My boss could
>talk on his phone fine until he cranked up Foxnews feed. Once the video
>started, he couldn't talk on his phone anymore (bad quality or total loss of
>call).
What I've done here is probab
.@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, July 14, 2009 5:04 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] QoS
>
>
> Howdy,
>
> Getting ready to play with QoS setting
s-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Wednesday, July 15, 2009 8:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] QoS
>
>
> On Wed, 15 Jul 2009, Danny Nicholas wrote:
>
>
>>
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, July 15, 2009 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] QoS
On Wed, 15 Jul 2009, Danny Nicholas wrote:
> In my shop, we got
se at the same time.
Cheers,
j
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, July 14, 2009 5:04 PM
> To: asterisk-users@lists.digium.com
>
LaCoursiere
Sent: Tuesday, July 14, 2009 5:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] QoS
Howdy,
Getting ready to play with QoS settings. We have an asterisk 1.4.23
server running in a colo bunker in the US Virgin Islands under a large
radio tower. That tower has
Howdy,
Getting ready to play with QoS settings. We have an asterisk 1.4.23
server running in a colo bunker in the US Virgin Islands under a large
radio tower. That tower has multiple "sector" radio/antenna pairs that
blanket a valley in 802.11a. The customers have directed dishes aimed at
Jeremy Mann wrote:
> Access-list 100 permit ip host any
>
> Class-map match-any voip
> Match access-group 100
>
> Policy-map voip
> Class voip
> Priority 256
> Class class-default
> Fair-queue
>
> Interface fastethernet 0
> Service-policy output voip
>
>
> Above is what I do to prioritize
David Backeberg wrote:
On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
wrote:
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each office also has a 1Mb
downst
estination in ~45 msec.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Garth van
Sittert
Sent: Friday, May 08, 2009 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [aste
Dave Platt wrote:
> OpenVPN doesn't suffer from this problem. Although it's SSL-based
> (and one might think it does everything through SSL-over-TCP),
> it actually sends the VPN traffic via UDP... it uses TCP only
> for the negotiation and administrative aspects of setting up
> the VPN connecti
> I would think that VoIP over VPN is a bad idea as UDP packets need to be
> in realtime not corrected by the TCP of the VPN.
That depends very much on the VPN in use.
OpenVPN doesn't suffer from this problem. Although it's SSL-based
(and one might think it does everything through SSL-over-TCP)
On Friday 08 May 2009 10:07:43 Garth van Sittert wrote:
> I would think that VoIP over VPN is a bad idea as UDP packets need to be
> in realtime not corrected by the TCP of the VPN.
Not all VPNs use TCP. OpenVPN, in particular, uses UDP for the backbone.
--
Tilghman
___
I would think that VoIP over VPN is a bad idea as UDP packets need to be
in realtime not corrected by the TCP of the VPN.
Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za
Aurimas Skirgaila wrote:
> Despite the VPN overhead, running VOIP through VPN is good idea
> because
On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
wrote:
> I've got multiple satellite office all linked back to the main office
> via VPN. Each office has their own asterisk server which registers back
> to the main office's Asterisk server. Each office also has a 1Mb
> downstream / 384k - 768k up
On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere wrote:
>
> On Fri, 8 May 2009, Aurimas Skirgaila wrote:
>
> > Despite the VPN overhead, running VOIP through VPN is good idea because
> VPN
> > reorders encapsulated UDP packets in correct order. Security matters as
> > well.
>
> Reorders? How so?
Access-list 100 permit ip host any
Class-map match-any voip
Match access-group 100
Policy-map voip
Class voip
Priority 256
Class class-default
Fair-queue
Interface fastethernet 0
Service-policy output voip
Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls,
On Fri, 8 May 2009, Aurimas Skirgaila wrote:
> Despite the VPN overhead, running VOIP through VPN is good idea because VPN
> reorders encapsulated UDP packets in correct order. Security matters as
> well.
Reorders? How so? I think it will maintain the order, only if they have
arrived in the "
Despite the VPN overhead, running VOIP through VPN is good idea because VPN
reorders encapsulated UDP packets in correct order. Security matters as
well.
I'd suggest to route VNC packets rather over internet than VPN (so do I), as
VPN usually has the highest priority.
On Thu, May 7, 2009 at 11:33
I do not have examples, but if you are using the 1700 series router in order
to originate the ipsec vpn, you may use command qos pre-classify (please
search for it on cco.cisco.com)
On Thu, May 7, 2009 at 9:54 PM, Brent Davidson
wrote:
> I've got multiple satellite office all linked back to the m
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each office also has a 1Mb
downstream / 384k - 768k upstream connection. The branches are using
Speex for their
Anael DIAZ wrote:
> Hi!
> I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2
> and this didn't accept voip QoS and can't route the packets having voip
> QoS.
> So I should change voip packets to be routing with centOS.
> I want to use iproute2 but i don't what to do after i
A clearer explanation of your problem, including examples and output, is
needed.
Anael DIAZ wrote:
> Hi!
> I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2
> and this didn't accept voip QoS and can't route the packets having voip
> QoS.
> So I should change voip packets
Hi!
I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2
and this didn't accept voip QoS and can't route the packets having voip QoS.
So I should change voip packets to be routing with centOS.
I want to use iproute2 but i don't what to do after installing iproute2.
Anyone cou
You SHOULD be concerned with QOS. All the way to an including the vendor
or your service cold really sucku
Michael Graves wrote:
> On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote:
>
>
>> I will have a small shop with ~4 phones using an HP server with Asterisk on
>> it, it has two N
On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote:
>I will have a small shop with ~4 phones using an HP server with Asterisk on
>it, it has two NICS and so I planned on plugging one into the cable modem, and
>the other into the switch. I was going to let this box perform NAT for the
>c
I will have a small shop with ~4 phones using an HP server with Asterisk on it,
it has two NICS and so I planned on plugging one into the cable modem, and the
other into the switch. I was going to let this box perform NAT for the company
but I am concerned about QOS for the VOIP portion.
Anyone
Chris Mason wrote:
> QOS can only be on outgoing, you can't set the priority of a packet
> after you receive it. The only other solution would be the cooperation
> of the ISP to provide QOS upstream of you. Good luck.
QOS is probably not the most precise term as it's normally associated with
RSVP,
J. Oquendo wrote:
> it does, when someone can realistically point this out please let me
> know so I can switch from a DS3 to T1 and save money.
>
Use the T1 for voice and get a DSL modem for your data use? :)
___
-- Bandwidth and Colocation Provided
J. Oquendo wrote:
> Its fine and dandy, but the problem is you're still getting 5 packets.
> You're still saturated period. No QoS in the world outside of your
> provider and more bandwidth can alleviate that. Your provider is not
> going to care what you do once its passed to the CPE. So look at i
At 05:59 AM 4/17/2008, you wrote:
>Not at all "little". If you have a lot of low priority outgoing traffic
>(i.e. p2p) saturating your link, uplink traffic shaping will mean the
>difference between a completely unintelligible call and something very
>acceptable.
My network looks like this:
Cable
On Thu, 2008-04-17 at 11:40 -0400, Chris Mason (Lists) wrote:
> Mike wrote:
> > do a decent
> > job of providing QoS on the upstream, which is where you (usually) need it
> > anyways.
>
> QOS can only be on outgoing,
Which is what he meant when he said "upstream" I believe.
> you can't set the
Mike wrote:
> do a decent
> job of providing QoS on the upstream, which is where you (usually) need it
> anyways.
QOS can only be on outgoing, you can't set the priority of a packet
after you receive it. The only other solution would be the cooperation
of the ISP to provide QOS upstream of you
: Thursday, April 17, 2008 10:34
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] QOS for outgoing SIP ... Who
> needs QoS anyway!
>
>
> May I suggest the following read:
>
> A Beginners Guide To Successful VOIP Over DSL
>
>
May I suggest the following read:
A Beginners Guide To Successful VOIP Over DSL
http://www.smallnetbuilder.com/content/view/30340/83/
Which covers both QoS and traffic shaping in small routers. It was
written based upon my own experience with both Asterisk and hosted PBX
providers.
Michael
On
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Brian J. Murrell wrote:
| I think I've made it clear that my argument is only about uplink shaping
| and the requirement for it given the asymmetric nature of a lot of last
| mile connections existing today. Funny enough that is *exactly* what
| th
On Thu, 2008-04-17 at 09:25 -0400, J. Oquendo wrote:
>
> Is it? So you're telling me if you're saturated on the way in, fixing up
> your packets on the way out is the solution.
I think I've made it clear that my argument is only about uplink shaping
and the requirement for it given the asymmetric
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Brian J. Murrell wrote:
| Not at all "little". If you have a lot of low priority outgoing traffic
| (i.e. p2p) saturating your link, uplink traffic shaping will mean the
| difference between a completely unintelligible call and something very
| acc
On Thu, 2008-04-17 at 08:36 -0400, J. Oquendo wrote:
>
> Brian J. Murrell wrote:
>
> | But certainly at my choke point which is of course my Internet uplink,
^^^
> I
> | can apply QOS (i.e. traffic shaping, which is what the O
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Brian J. Murrell wrote:
| Maybe your understanding of QOS and mine is different. Of course I have
| no illusions that I can assign a priority to my packets that is going to
| be meaningful to anyone once they leave my network.
|
| But certainly at
On Thu, 2008-04-17 at 07:54 -0400, sil wrote:
>
> Apparently man people don't understand that those QoS settings on
> routers mean little most of the time. Most providers resell QoS as a
> premium service, so while many waste their time "painting their packets"
> those markings get stripped.
Mayb
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Brian J. Murrell wrote:
|
| Yeah, well, that's all fine and dandy as long as "more capacity" is an
| option. Many people are already subscribed to the most capacity
| available to them and using it.
|
| b.
Apparently man people don't understand th
On Thu, 2008-04-17 at 07:16 -0400, sil wrote:
> Simon wrote:
>
> | Is this worth doing? If so, what ports should i specifiy?
>
>
> http://www.bricklin.com/qos.htm
Yeah, well, that's all fine and dandy as long as "more capacity" is an
option. Many people are already subscribed to the most capac
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Simon wrote:
| Is this worth doing? If so, what ports should i specifiy?
http://www.bricklin.com/qos.htm
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iQIVAwUBSAcx
Simon wrote:
> Hi There,
>
> We have our Asterisk box using a external SIP provider for outgoing
> calls over our DSL line. This seems to be going well... But i do have
> the ability to set some QOS ports in our linksystem DSL router... Its
> faily basic, so im wondering if it will help at all...
On Wed, Apr 16, 2008 at 11:49 PM, Simon <[EMAIL PROTECTED]> wrote:
> Hi There,
>
> We have our Asterisk box using a external SIP provider for outgoing
> calls over our DSL line. This seems to be going well... But i do have
> the ability to set some QOS ports in our linksystem DSL router... Its
>
Hi There,
We have our Asterisk box using a external SIP provider for outgoing
calls over our DSL line. This seems to be going well... But i do have
the ability to set some QOS ports in our linksystem DSL router... Its
faily basic, so im wondering if it will help at all...
We can specify High, Med
I would like to be able to monitor my QoS.. I see that Qwest is using this
QoS Manager (Firehunter)
http://www.home.agilent.com/cgi-bin/pub/agilent/Product/cp_Product.jsp?NAV_ID=-536885714.536882909.00&LANGUAGE_CODE=eng&CONTENT_KEY=49888&ID=49888&COUNTRY_CODE=US
I have some buddies who work at
> >How much of an impact can/does local network traffic have on call quality?
> >Would opening large files on local servers affect call quality? We are
> >running QoS on the router but that will only prioritize traffic in/out of
> >the network.
>
> Sure it can. If you have a network segment that
On 14:35, Tue 09 Aug 05, Geoff Manning wrote:
> Julio Arruda wrote:
> > Half duplex by itself doesn't hurt (depends in number of calls and etc
> > really, but anyway...)
> > What is a killer for VOIP is duplex mismatch.
> > If you have autonegotiation enabled, and your peer (the switch ?) has
> > a
Julio Arruda wrote:
> Half duplex by itself doesn't hurt (depends in number of calls and etc
> really, but anyway...)
> What is a killer for VOIP is duplex mismatch.
> If you have autonegotiation enabled, and your peer (the switch ?) has
> autoneg off, and 100/Full-duplex hard coded, you WILL have
Half duplex by itself doesn't hurt (depends in number of calls and etc
really, but anyway...)
What is a killer for VOIP is duplex mismatch.
If you have autonegotiation enabled, and your peer (the switch ?) has
autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex
mismatch.
And th
Geoff Manning wrote:
Eric Wieling aka ManxPower wrote:
Are your phones on shared links to the switch?
i.e.
PC -> Phone -> Switch?
Actually it is a legacy PBX - Asterisk integration
Legacy Handset --> Mitel SX 200 --> Asterisk --> Switch --> Router
The calls come inbound over the internet
Eric Wieling aka ManxPower wrote:
> In my experience, for local LAN audio issues, duplex problems are the
> problem, not LAN traffic.
>
Rock on!
I am in half duplex mode:
serv01:~# ethtool eth0
Settings for eth0:
Supported ports: [ MII ]
Supported link modes: 10baseT/Half 10b
Eric Wieling aka ManxPower wrote:
> Are your phones on shared links to the switch?
>
> i.e.
>
> PC -> Phone -> Switch?
Actually it is a legacy PBX - Asterisk integration
Legacy Handset --> Mitel SX 200 --> Asterisk --> Switch --> Router
The calls come inbound over the internet as SIP to Asteri
Geoff Manning wrote:
Michael Graves wrote:
Oh, yes! That's a good possibility as well, expecially with some Cisco
gear.
One problem that I had was related to saturating a segment during an
automated backup procedure. When a server in the UK started its backup
processes at an apparently idel ti
On Tue, 09 Aug 2005 11:26:11 -0500, Eric Wieling aka ManxPower wrote:
Geoff Manning wrote:
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated
and
you're also pushing VOIP data over that segment you'll have
problems.
In practice most networks are not t
Michael Graves wrote:
> Oh, yes! That's a good possibility as well, expecially with some Cisco
> gear.
>
> One problem that I had was related to saturating a segment during an
> automated backup procedure. When a server in the UK started its backup
> processes at an apparently idel time callers in
On Tue, 09 Aug 2005 11:26:11 -0500, Eric Wieling aka ManxPower wrote:
>Geoff Manning wrote:
>> Michael Graves wrote:
>>
>>>Sure it can. If you have a network segment that's fully saturated and
>>>you're also pushing VOIP data over that segment you'll have problems.
>>>In practice most networks ar
Geoff Manning wrote:
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated and
you're also pushing VOIP data over that segment you'll have problems.
In practice most networks are not that busy, but it can happen. If
your phones, switch and NICs are VLAN capable
Michael Graves wrote:
> Sure it can. If you have a network segment that's fully saturated and
> you're also pushing VOIP data over that segment you'll have problems.
> In practice most networks are not that busy, but it can happen. If
> your phones, switch and NICs are VLAN capable you can setup a
On Tue, 9 Aug 2005 12:07:07 -0400, Geoff Manning wrote:
>How much of an impact can/does local network traffic have on call quality?
>Would opening large files on local servers affect call quality? We are
>running QoS on the router but that will only prioritize traffic in/out of
>the network.
Sur
How much of an impact can/does local network traffic have on call quality?
Would opening large files on local servers affect call quality? We are
running QoS on the router but that will only prioritize traffic in/out of
the network.
___
Asterisk-Users ma
I know this isn't directly related to * but I found it works very well in my
voip environment. Check out cFosSpeed @ www.cfos.de. It gives you QoS based
on applications and also seems to have increased my network throughput.
Cheers,
S.
smime.p7s
Description: S/MIME cryptographic signature
___
Hi All,
There are two option in QoS settings of the SIPURA ATA. ( I can't just
remember them). please tell me what is better and which one should I choose
for my DSL line (128kbps) with a small LAN.
thank you
kumara
___
Asterisk-Users mailing list
Asteri
Hi All,
There are two option in QoS settings of the SIPURA ATA. ( I can't just
remember them). please tell me what is better and which one should I choose
for my DSL line (128kbps) with a small LAN.
thank you
kumara
___
Asterisk-Users mailing list
Asteri
Hi All
From where we can get the data for
1) ASR on various countries
2) Average Call drop on VoIP
3) Average Call Quality
This we require to get an idea of what types of
problem normally users use to face on voip and what is the average percentage of
those problems.
Pls. help me if
> -Original Message-
> From: cmisip [mailto:[EMAIL PROTECTED]
> Sent: Friday, May 06, 2005 9:41 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Qos betwenn WIFI machines in LAN? oh323?
>
>
> How do you do Qos betwe
How do you do Qos between two machines when the bandwidth changes such
as with WIFI? I normally get about 15 Mbit/s but this changes between 9
to 19 Mbits/s at times. Also, I use ohphone. How does one prioritize
these oh323 packets or tag them for higher priority? I also have mythtv
running in
lto:[EMAIL PROTECTED] On Behalf Of
Jean-Christophe Heger
Sent: Friday, 06 May 2005 18:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] QoS for improvements
That's funny, people having good bandwidth always have a better way to do
it. You should feel lu
Jean-Christophe,
Thank you for the explanation. I've never been in a situation demanding
adjustment of MTU. It's not so much that I think I have a better way,
only that my circumstances lend themselves to a simple solution. I did
start out using * with only 256k upload speed. I decided to stay wit
That's funny, people having good bandwidth always have a better way to
do it. You should feel lucky, because no one provides 768kbps upstreams
in Switzerland, except if you want to pay 1'000 USD per month for a
leased line.
There is nothing complicated, just mathematics.
Here is the formula:
MTU
Sometimes this all sounds so complicatedbut it needn't be. I
suppose it can vary with the size of your installation.
I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic
shaping feature I establish inbound and outbound pipes which are
bandwidth restricted to just less than my mesu
I've spent may hours to play with HTB QoS settings on the firewall, but with
absolutely no effect. In fact, this is normal, because the time required to let
a data packet going through the ADSL line will break the voice jitter. The only
right way to handle this issue is to modify the MTU on the
Kumara Jayaweera wrote:
I want to run VoIP in the same LAN (15 windows clients) which we use for
surfing the Internet.
Some magic words: QoS Asterisk HTB TC. Not easy to find good material
over the internet, but Google may give you some ideas - how to use them
is another problem, which you have t
> don't know what it (QoS= Quality of Service) is. I hope you may help me
> giving "Links" to read and briefing me your ideas.
1 minute of google search I found this :
http://compnetworking.about.com/od/networkdesign/l/bldef_qos.htm
which looks like a pretty nice explanation
hth
_
Hello! Everybody!!,
I want to run VoIP in the same LAN (15 windows clients) which we use for
surfing the Internet. 6-7 softphones in the same client's machines is 'the
target'. My DSL is 128kbps, (I can go to 256kbps if required). So, I am told
to install some QoS's in the LAN to improve the voice
Hi -
We've been using IAX forwards between sites for a little while now
(with centralized VM). For the most part, it is fine, but I have some
very minor, yet persistent QoS issues on calls over the IAX forwards.
For most normal calls, there are very occasional minor glitches, just
an infreque
I run m0n0wall (http://m0n0.ch/wall) on a Soekris 4501 embedded PC
(http://www.soekris.com). Very tweakable. Under $200.
Michael
On Fri, 22 Apr 2005 10:42:20 -0700, Max Clark wrote:
>Hi all,
>
>I am looking for good (sub $200 dollars) routers to support VoIP
>installations. What is available at
-Users] QOS Routers
How about a linksys wrt54g with sveasoft firmware? Has some
shaping and many other nice features...
Jay: can you elaborate on your standpoint on the svea firmware?
thx
/rg
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Hi all,
I am looking for good (sub $200 dollars) routers to support VoIP
installations. What is available at this point? I've used Netscreen and
Checkpoint in the past, they are just too much overkill for this
application.
TIA,
Max
--
Max Clark
max [at] clarksys.com
http://www.clarksys.co
Sveasoft is useless -- use hyperWRT instead.
> -Original Message-
> From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED]
> Sent: Friday, April 22, 2005 4:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] QOS Rout
ailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] QOS Routers
>
>
> Hi all,
>
> I am looking for good (sub $200 dollars) routers to support VoIP
> installations. What is available at this point? I've used
> Netscreen and
> Checkpoint in t
Max Clark wrote:
Hi all,
I am looking for good (sub $200 dollars) routers to support VoIP
installations. What is available at this point? I've used Netscreen and
Checkpoint in the past, they are just too much overkill for this
application.
I was using a boot off cd distro called Devil linux... wit
You may want to check out edgewaternetworks.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Max Clark
> Sent: Friday, April 22, 2005 11:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:
Subject: [Asterisk-Users] QOS Routers
Hi all,
I am looking for good (sub $200 dollars) routers to support VoIP
installations. What is available at this point? I've used Netscreen and
Checkpoint in the past, they are just too much overkill for this
application.
TIA,
Max
--
Max Clark
You may want to check out edgewaternetworks.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Max Clark
> Sent: Friday, April 22, 2005 11:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:
Good day all
I'm looking for a type of QOS test tool(software)
I want to test if a link is good enough for voip and test witch ones
will be the best..ens
any ideas
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tos=0xb8 will set the the packet to be DSCP EF (Cisco likes to use DSCP)
Rich Adamson wrote:
Does anyone know how setting the TOS bits in iax.conf corresponds to
the Cisco TOS types?
For example, if I set:
tos=0x04
in iax.conf, and on the Cisco, I use:
access-list 110 permit ip any any tos 4
I ca
> Does anyone know how setting the TOS bits in iax.conf corresponds to
> the Cisco TOS types?
>
> For example, if I set:
>
> tos=0x04
>
> in iax.conf, and on the Cisco, I use:
>
> access-list 110 permit ip any any tos 4
>
> I can't get the Cisco to match any packets. I've tried various
> c
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