Re: Networkers, pt. 2 [7:70768]

2003-06-19 Thread Steven A. Ridder
I'll be there.  Looking forward to it.

--

RFC 1149 Compliant.



Mossburg, Geoff (MAN-Corporate)  wrote in
message news:[EMAIL PROTECTED]
 I know Robert McCallum already asked this, but who is going to Networkers
in
 Orlando next week? Any cool GroupStudy router config parties gonna happen?
 :-)
 Geoff Mossburg




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Re: Upgrade license [7:70919]

2003-06-19 Thread Steven A. Ridder
I believe it's just a new activation key.

Copied from Cisco's website:
There are a couple of reasons that you may need to upgrade the activation
key on your PIX.

  a.. Your PIX does not currently have VPN-DES or VPN-3DES encryption
enabled.

  Note: VPN-DES encryption must be enabled for you to manage your PIX using
PDM. Registered users may obtain a free 56-bit VPN-DES activation key by
completing the PIX 56-bit License Upgrade Key form. VPN-3DES activation keys
must be purchased through your local reseller or Cisco sales representative.

  b.. Your PIX currently does not have failover activated.

  c.. You are upgrading from a connection-based license to a feature-based
license.


--

RFC 1149 Compliant.



maine dude  wrote in message
news:[EMAIL PROTECTED]
 Hi,

 Can someone please tell me the process of upgrading a restricted license
to
 a unrestricted one a PIX firewall please.

 Is it just as simple as downloading a new IOS or more.

 Thanks is advance,

 -Dj





 -
 Yahoo! Plus - For a better Internet experience




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Re: Site to Site VPN Monitering on PIX [7:62676]

2003-02-09 Thread Steven A. Ridder
CiscoWorks VMS 2.1

--

RFC 1149 Compliant.



Curious  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I have setup Site to Site VPN between our corporate PIX 515 and our
 developers PIX 501, i want to moniter the VPN traffic of these Site to
Site
 VPN connections.
 Please tell me what tools are available to accomplish this.

 thanks,


 --
 Curious

 MCSE, CCNP




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Re: VoIP from behind PIX [7:60859]

2003-01-13 Thread Steven A. Ridder
Just so I understand (crypto is a tough subject for me), if one knows the
length of a packet before crypto processing, it becomes a weakness
because(fill in the blank).







Howard C. Berkowitz  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 At 12:03 AM + 1/13/03, Steven A. Ridder wrote:
 I agree with Matt. The PIX 515 introduces jitter.  Not sure what the
Cisco
 IPT Safe document is talking about.

 This may be a rather obscure point, but if a cryptographic device
 takes different amounts of time to encrypt and decrypt equal-length
 blocks of text with different contents, it is a cryptographic
 vulnerability and may also provide a covert channel.

 These time differences, however, have to be constant.  If they are
 simply a function of processing load, there is no vulnerability.

 Latency is not a cryptosecurity issue, although, obviously, it can
 affect speech intelligibility.

 
 
 Matt Hill  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   Good luck..
 
   However you will get latency and jitter issues during the time the
PIXs
   encrypt/decrypt the voice packets...
 
   Matt
 
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf
   Of
Simer Mayo
Sent: Friday, 10 January 2003 6:05 PM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: VoIP from behind PIX
   
1. Will PIX 515 handle VoIP traffic?
2. Will PIX 501 handle VoIP traffic?
3. Can we VPN between 2 (site-to-site) and pass VoIP traffice thru
the
VPN
   
Thanks
   
 Simer




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Re: fragmentation question [7:60643]

2003-01-13 Thread Steven A. Ridder
Thanks!  I was just curious.  What about L2 headers in Frame Relay
Fragmentation (frf.12)?


Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Steven A. Ridder wrote:
 
  Does anyone know if a packet is fragmented, that the specific
  values in a
  field are replicated across all headers of the fragmented
  packets, or just
  the first one?
 
  Meaning, if I have a packet that has IP Prec 5, and a router
  along the way
  has to fragment the packet, would it be so kind as to put IP
  Prec on all the
  headers?

 Yes, it should. Per RFC 791, a router (or gateway as the RFC calls it)
 copies the contents of the header fields from the original datagram into
the
 new headers of all the fragments. Of course, the following fields may
 change, however:

   (1) options field
   (2) more fragments flag
   (3) fragment offset
   (4) internet header length field
   (5) total length field
   (6) header checksum

 Also, with the options field, options may or may not be copied into each
 fragment. There's a bit that the sender can set saying whether they must
be
 or not. But in general, all bits and bytes are copied into each fragment
IP
 header.

 Prscilla

 
  Steve
 
 
  Priscilla Oppenheimer  wrote in
  message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   Mossburg, Geoff (MAN-Corporate) wrote:
   
Thanks for clearing that up;
  
   No problem.
  
I don't mind being told I'm
mistaken. I
recently decided that the only way I'm really going to learn
from this group
is to take a chance on confirming what I THINK I know, and
asking questions
about what I DON'T know. :) A lesson in humility, to be
  sure.
  
   I know what you mean. I like to pretend to be an uber goddess
  of all
  things
   tech, but to learn, I have to admit to lots of cluelessness
  in some areas.
   It can be a bit painful, but definitely worth it! :-)
  
   Priscilla
  
GM
   
-Original Message-
From: Priscilla Oppenheimer [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, January 08, 2003 5:35 PM
To: [EMAIL PROTECTED]
Subject: RE: fragmentation question [7:60643]
   
   
Mossburg, Geoff (MAN-Corporate) wrote:

 Someone correct me if I'm wrong:
   
OK, you're wrong. :-) Look it up or use a protocol analzyer.
   
 All the fragments have the TCP/UDP/IP headers, or else
  they
 can't be
 routed to their destination.
   
Routing to their destination just requires the IP header,
  which
is in each
fragment. The TCP or UDP headers are not in the fragments,
  past
the first
one. The IP layer at the end device puts it all back
  together
and hands the
packet to the TCP or UDP layer. TCP or UDP get the full
  packet
and can
route it to the correct process, based on the destination
port number.
   
 Fragmentation is just a way of breaking up the data
  payload
 into smaller
   
Data payload from IP's point of view.
   
 packets, but it puts individual headers on each packet.
 MTU is the total size of each packet, including the
  header.
   
The term isn't always used that way, though.
   
 GM

 -Original Message-
 From: Paul Dong So [mailto:[EMAIL PROTECTED]]
 Sent: Wednesday, January 08, 2003 4:19 PM
 To: [EMAIL PROTECTED]
 Subject: fragmentation question [7:60643]


 Hi All,

 Please shed a light on this as I am confused.

 Fragmentation for UDP/TCP:
  * Only the first fragment contains the UDP or TCP
  header, not
 the
 sequencial fragments?

 Fragementation for IP packets
  * every fragmented packet will contains ip header?

 MTU 1500 bytes, doesn't it mean the data payload can not
exceed
 1500
 bytes or the whole packet size(payload+header) can not
  exceed
 1500
 bytes?

 Thanks in advance

 Paul




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Re: VoIP from behind PIX [7:60859]

2003-01-12 Thread Steven A. Ridder
I agree with Matt. The PIX 515 introduces jitter.  Not sure what the Cisco
IPT Safe document is talking about.


Matt Hill  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Good luck..

 However you will get latency and jitter issues during the time the PIXs
 encrypt/decrypt the voice packets...

 Matt

  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf
 Of
  Simer Mayo
  Sent: Friday, 10 January 2003 6:05 PM
  To: [EMAIL PROTECTED]
  Cc: [EMAIL PROTECTED]
  Subject: VoIP from behind PIX
 
  1. Will PIX 515 handle VoIP traffic?
  2. Will PIX 501 handle VoIP traffic?
  3. Can we VPN between 2 (site-to-site) and pass VoIP traffice thru the
  VPN
 
  Thanks
 
  Simer




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Re: fragmentation question [7:60643]

2003-01-12 Thread Steven A. Ridder
Does anyone know if a packet is fragmented, that the specific values in a
field are replicated across all headers of the fragmented packets, or just
the first one?

Meaning, if I have a packet that has IP Prec 5, and a router along the way
has to fragment the packet, would it be so kind as to put IP Prec on all the
headers?

Steve


Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Mossburg, Geoff (MAN-Corporate) wrote:
 
  Thanks for clearing that up;

 No problem.

  I don't mind being told I'm
  mistaken. I
  recently decided that the only way I'm really going to learn
  from this group
  is to take a chance on confirming what I THINK I know, and
  asking questions
  about what I DON'T know. :) A lesson in humility, to be sure.

 I know what you mean. I like to pretend to be an uber goddess of all
things
 tech, but to learn, I have to admit to lots of cluelessness in some areas.
 It can be a bit painful, but definitely worth it! :-)

 Priscilla

  GM
 
  -Original Message-
  From: Priscilla Oppenheimer [mailto:[EMAIL PROTECTED]]
  Sent: Wednesday, January 08, 2003 5:35 PM
  To: [EMAIL PROTECTED]
  Subject: RE: fragmentation question [7:60643]
 
 
  Mossburg, Geoff (MAN-Corporate) wrote:
  
   Someone correct me if I'm wrong:
 
  OK, you're wrong. :-) Look it up or use a protocol analzyer.
 
   All the fragments have the TCP/UDP/IP headers, or else they
   can't be
   routed to their destination.
 
  Routing to their destination just requires the IP header, which
  is in each
  fragment. The TCP or UDP headers are not in the fragments, past
  the first
  one. The IP layer at the end device puts it all back together
  and hands the
  packet to the TCP or UDP layer. TCP or UDP get the full packet
  and can
  route it to the correct process, based on the destination
  port number.
 
   Fragmentation is just a way of breaking up the data payload
   into smaller
 
  Data payload from IP's point of view.
 
   packets, but it puts individual headers on each packet.
   MTU is the total size of each packet, including the header.
 
  The term isn't always used that way, though.
 
   GM
  
   -Original Message-
   From: Paul Dong So [mailto:[EMAIL PROTECTED]]
   Sent: Wednesday, January 08, 2003 4:19 PM
   To: [EMAIL PROTECTED]
   Subject: fragmentation question [7:60643]
  
  
   Hi All,
  
   Please shed a light on this as I am confused.
  
   Fragmentation for UDP/TCP:
* Only the first fragment contains the UDP or TCP header, not
   the
   sequencial fragments?
  
   Fragementation for IP packets
* every fragmented packet will contains ip header?
  
   MTU 1500 bytes, doesn't it mean the data payload can not
  exceed
   1500
   bytes or the whole packet size(payload+header) can not exceed
   1500
   bytes?
  
   Thanks in advance
  
   Paul




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Re: callmanager 3.3 [7:59160]

2002-12-13 Thread Steven A. Ridder
Supposedly by end of the month.  Docs and stuff are slowly trickling out,
but noting good yet.

--

RFC 1149 Compliant.



supernet  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Several months ago, Cisco TAC told me that CallManager 3.3 would be
 released in Nov. this year. Is it out yet? I don't see it in Cisco
 download area. Thanks.




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Re: Off Topic - Missed it by that much - CCIE Lab report [7:58587]

2002-12-04 Thread Steven A. Ridder
was your problem split horizon?


The Long and Winding Road  wrote in
message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 In the words of the esteemed Maxwell Smart, missed it by that much.

 Good test. Liked it a lot.  Can't say much about the content, obviously.

 The 3550's were there. I think that those who have expressed reservations
 about this will find little to worry about. The Lab writers did a pretty
 good job of integrating the devices into the rack and the exam.

 I think I was more surprised by what I did NOT see than by what I did see.
 Wish I could say more.

 There were the usual off the wall requirements. I knew the names, or had
 heard of the technology, but had never practiced it. Here's where the doc
CD
 came in handy. It was very easy to locate the information and do the
 required configuration.

 I did have one very odd problem I was unable to solve. My own practice,
not
 to mention the doc CD configuration guide, told me that a particular
 configuration should have worked. But it didn't. I've mocked up the
 configuration here at home, and it took a total of 10 minutes to start
from
 a router with no configuration and have it up and running correctly. But
in
 the Lab it just would not work. I have an inquiry in to the CCIE Lab
folks,
 asking them to check the rack. I believe there is a physical problem,
 although for the life of me I cannot come up with a plausible explanation
as
 to why. I would get more specific, except this would be a direct violation
 of NDA. I will say that anyone who sits at rack 12 in San Jose - if you
are
 absolutely certain your configuration is correct, tell the proctor. I
 hesitated to do so, and I paid the price. You'll know when you see it ;-)

 I did one stupid thing, and the more I think about it, I should have
 corrected it immediately when I discovered it. When I first created my
 notepad file with my alias commands, I stupidly did most of them as alias
 configure rather than alias exec Given that the lab is graded pretty
much
 by scripts, I have this bad feeling that this mistake may have interfered
 with the operation of some of those scripts, meaning that I was not given
 credit for successfully completed tasks just because the script was unable
 to function properly.

 You are no longer given a point total in your report. When I counted up
 points in the late afternoon, I thought I had between 60 and 70. I had no
 reachability problems, save to one interface, and that interface had
nothing
 depending upon it. I knew I didn't have enough points to pass, but I
thought
 I was close. To judge from my score report, the final total was maybe
35-45
 depending. As those of you who have been there know, the dependings will
 kill you every time. :-)

 I can say I had a lot of fun doing this test. That's probably part of the
 reason I failed - I'm having too much fun. I can also say I'm hot to trot.
I
 can taste it. I'm yay close to passing, and I want back in as soon as I
can
 get there. You can bet I'm checking CCO regularly for those open dates.

 Afterwards, I had the pleasure of hooking up with groupstudy regular Larry
 Letterman. Larry - thanks for the tour - it was impressive. I was reminded
 of exactly why I got into the tech business in the first place - the
desire
 to do things like you are doing, important things, things that keep
 businesses competitive. You're doing a great job and I appreciate your
 taking some time to show me what you're working on.

 well, another time.

 Back on the road again.
 --
 TANSTAAFL
 there ain't no such thing as a free lunch




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Re: is there anyone migrating isdn backup to dsl b [7:58568]

2002-12-04 Thread Steven A. Ridder
I say DSL has no multi-service (or very limited) capabilities.  There
isn't much in terms of QoS, LFI or other voice/video tools.  Plus there is
no QoS across the DSL network (if over Internet) and no standard nation-wide
(no National provider).

If you say, there is no voice going across network, or video, then you are
doing your client a dis-service by providing no upgrade path towards that
eventual path.

Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 It sounds like DSL has a low mean time between failure (MTBF) but a high
 mean time to repair (MTTR), which can be just as bad, especially if it's
 your only backup. Of course, your mileage may vary (YMMV), depending on
the
 service provider. Also, a service level agreement (SLA) would help, as
Chuck
 mentions.

 Does that message set a record for the number of acronyms used? :-)

 Priscilla

 The Long and Winding Road wrote:
 
  Mirza, Timur  wrote in
  message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   we are looking to migrate isdn backup at our retail stores to
  dsl...is
  there
   anyone that has performed this already?
 
 
  CL: having done a number of data networks that were DSL based (
  but none
  migrating ISDN to DSL ) I can offer this consideration: if a
  DSL link goes
  down for whatever reason, it may take more than a couple of
  days for your
  telco to get it back up and working. You will want to have some
  solid
  service level agreements in place. DSL on the whole is
  extremely reliable.
  The problem tends to be during those rare instances when it is
  down for
  whatever reason, some telcos seem to have DSL repair low on
  their priority
  list.
 
  CL: other than that caviat, why not?
 
 
  
   Timur Mirza
   Principal Network Engineer
   Network Planning  Engineering, West Region
   15505-B Sand Canyon Avenue
   Irvine, California 92618
   Verizon Wireless
   949.286.6623 (o)
   949.697.7964 (c)




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Re: Enterprise technologies [7:58493]

2002-12-04 Thread Steven A. Ridder
I'd focus on Avvid technologies, centraly managed security and storage
solutions across nation-wide networks and public Internet (Cisco Works/ACS),
and on-line collaboration tools using open standards like LDAP, X.509,
h.323/SIP, etc.

That is where Enterprises are moving.


Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I may be starting a new project doing some writing about technologies used
 in enterprise networks. (read not service provider)

 Do I need to cover IS-IS? Or is it mainly ISPs that use this?

 How about MPLS? I should discuss it briefly, but aren't the main users of
 MPLS ISPs, not enterprise networks?

 Anyone using GARP? That's on my list to research too. I thought that Garp
 was a hero in a John Irving book.

 Alas, I have a lot to learn. Thank-you VERY much for answering these quick
 questions.

 ___

 Priscilla Oppenheimer
 www.troubleshootingnetworks.com
 www.priscilla.com




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Re: AW: Port Security on 3550 based on given MAC-Addre [7:58591]

2002-12-04 Thread Steven A. Ridder
You are correct.  I read it too quickly.


William Lijewski  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hello,

 The default for the maximum number of mac-addresses is one, and the
default
 violation is shutdown.

 Bill




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Re: Port Security on 3550 based on given MAC-Address and [7:58326]

2002-11-30 Thread Steven A. Ridder
Don't worry about the IP address.  The command you had was correct.  Why do
you ask?


--

RFC 1149 Compliant.



MK  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 How can I configure PortSecurity based on MAc-Address and
 IP-Address. I only know about switchport port-security mac-address
 but there must be a way to manage this in conjunction with an IP
 Static ARp entry 





 Thanx




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Re: Port Security on 3550 based on given MAC-Address and [7:58332]

2002-11-30 Thread Steven A. Ridder
No, just do the mac address.  That's what they're looking for.  Then limit
it to 1, because the default is 150.

On another note, what does the AW in the subject line (RE: in English)stand
for in German?  I used to live in y and I can't think of the word...



MK  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Because there must be a way, and I was asked about it in our
 Company. I know there is some secret behind !

 -Urspr|ngliche Nachricht-
 Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Im Auftrag von
 Steven A. Ridder
 Gesendet: Samstag, 30. November 2002 13:42
 An: [EMAIL PROTECTED]
 Betreff: Re: Port Security on 3550 based on given MAC-Address and
 [7:58326]

 Don't worry about the IP address.  The command you had was correct.  Why
 do
 you ask?


 --

 RFC 1149 Compliant.



 MK  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  How can I configure PortSecurity based on MAc-Address and
  IP-Address. I only know about switchport port-security
 mac-address
  but there must be a way to manage this in conjunction with an IP
  Static ARp entry 
 
 
 
 
 
  Thanx




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Re: Port Security on 3550 based on given MAC-Address and [7:58331]

2002-11-30 Thread Steven A. Ridder
MK  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Because there must be a way, and I was asked about it in our
 Company. I know there is some secret behind !

 -Urspr|ngliche Nachricht-
 Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Im Auftrag von
 Steven A. Ridder
 Gesendet: Samstag, 30. November 2002 13:42
 An: [EMAIL PROTECTED]
 Betreff: Re: Port Security on 3550 based on given MAC-Address and
 [7:58326]

 Don't worry about the IP address.  The command you had was correct.  Why
 do
 you ask?


 --

 RFC 1149 Compliant.



 MK  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  How can I configure PortSecurity based on MAc-Address and
  IP-Address. I only know about switchport port-security
 mac-address
  but there must be a way to manage this in conjunction with an IP
  Static ARp entry 
 
 
 
 
 
  Thanx




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Re: Multicast QOS Book .....Any Good?? [7:58137]

2002-11-26 Thread Steven A. Ridder
I heard from Wendel Odom himself that he is coming out with a Cisco-press
QoS book for the Exam, so I'd wait for that.  I thought I heard December.

--

RFC 1149 Compliant.



dre  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Cisco Nuts  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Hello, Has anyone used this book by Carl Timm for the CCIP MCast and Qos
  exam published by Sybex? Is it worth it to buy this book? How much does
  it help just to pass the exam?Thanks for all your input.Sincerely. Ccip:
  Multicast
  and Qos Study Guide
  Carl Timm  Jeff Witkowski

 I would concentrate on passing the BSCI and MPLS tests before bothering
 with the MCAST/QOS CCIP exam (unless assuming you have already
 passed both).  I really like Doyle (Vol. II) and Vegesna (Cisco Press IP
 QoS)
 for studying for this material.  The multicast info on ftp-eng.cisco.com
 should
 suffice for the most part.   I would instead recommend reading Vegesna and
 CCO a few times about QoS/MQC/etc and then going through the formal
 training (the web-based training is generally $499) if you fail the test
 once:
 TRN-QOS: Implementing Cisco QoS (QOS) v1.0

http://www.cisco.com/pcgi-bin/front.x/wwtraining/CELC/index.cgi?action=Cours
 eDescCOURSE_ID=1583
 rather than buying the Sybex book mentioned.

 -dre




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Re: VoIP Testing :MOS Vs PQSM [7:58061]

2002-11-25 Thread Steven A. Ridder
Just use the MOS charts alread yout there and not worry about it.  Why
reinvent the wheel?

--

RFC 1149 Compliant.



neil K.  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Guys,

 VoIP Testing, do you go by MOS or PQSM. I mean when testing  VoIP will
 perform on a network before implementing it. There are many tools that
give
 a MOS score and many other tools give a PSQM report. What do you
recommend?

 Thanks,
 Neil K.




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Re: Multicast Traffic Question [7:57932]

2002-11-23 Thread Steven A. Ridder
ping a multicast address.

--

RFC 1149 Compliant.



H  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I have just started to study for Multicast, and I am wondering whether
there
 is any simulator / programs that can simulate Multicast traffic.  Also,
can
 I use a Cisco router to act as Multicast Source (pumping out Multicast
 traffic), or used it as a Group member??

 Sorry if these are silly questions, but any advice would be greatly
 appreciated.

 Regards,
 H.




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Re: Block MSN Messenger [7:57595]

2002-11-18 Thread Steven A. Ridder
no.  don't waste your time.


Ahed Naimi  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Dear All;

 Is there any way to block MSN Messenger by using the access-list
statements
 on an IOS Cisco router.

 Thanks All.




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Re: Networkers download sessions [7:57587]

2002-11-17 Thread Steven A. Ridder
I lost it.  Could you post it please?

thanks

Steve


Howard C. Berkowitz  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Thanks. I have the information.


 At 1:37 AM + 11/18/02, Howard C. Berkowitz wrote:
 I confess to having download problems with a couple of recent
 Networkers sessions.  I was able to get to the recent European one
 (www.cisco.com/global/EMEA/networkers/), but at least half of the
 optical and routing presentations I downloaded had PDF file errors.
 Has anyone else had this problem?
 
 On going to www.cisco.com/networkers/nw02/pres, I can find the
 abstracts but not the download page similar to the one for '00 and
 '01.  Is there no download page yet for '02?
 
 TIA,
 Howard




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Re: OSPF adjacencies [7:57410]

2002-11-13 Thread Steven A. Ridder
It looks like the options in the packets do not march.  Any way to get a
sniffer on there to see what each is sending as options.  It could also be a
priority issue if the network is a broadcast/nbma network where neither is
being elected a DR?  Finally, could a checksum be bad?

--

RFC 1149 Compliant.



Jenny McLeod  wrote in message
news:200211140127.BAA14210;groupstudy.com...
 OK, I'll admit this is a real-life problem, not strictly a study question.
 I have a couple of OSPF adjacencies that refuse to start up.  Just to make
 this entertaining, these are not router to router - they are Cisco to
 mainframe, over a CIP.
 Five IP stacks neighbour the router - two are OK, three get stuck in
 EXSTART/EXCHANGE.  The five IP stacks also connect to a different router,
 and these adjacencies are fine.
 It looks to me like the classic MTU mismatch symptoms, but a printout of
the
 m/f definitions shows the MTUs to be 4096, as does show int on the
 router.  I'll get the m/f guru to check the definitions for white space -
I
 don't know if that will affect it.  There have been various m/f changes
 lately (and a couple of router ones) errors may have crept into the
configs.

 What has me baffled is some of the debug output from the router (debug ip
 ospf events).

 Nov 14 11:51:14.121 ESuT: OSPF: Rcv DBD from x.x.x.x on Channel6/0 seq
 0x3DCDF2DA opt 0x2 flag 0x7 len 32  mtu 0 state EXCHANGE
 Nov 14 11:51:14.121 ESuT: OSPF: Send DBD to x.x.x.x on Channel6/0 seq
 0x3DCDF2DA opt 0x42 flag 0x2 len 1472

 The debug doco isn't particularly detailed for this command, but I assume
 opt refers to the options field.  RFC 2328 seems to think that the first
two
 bits of the options field should be cleared, so the value of 0x42 being
sent
 by the router surprises me.
 Obviously the value of MTU being reported in the received DBD is also a
 concern!

 Other debug output indicates that the m/f sends the same DBD several times
 (same seq), which the router acks, then after this is received several
times
 the router claims
 Nov 14 11:51:20.037 ESuT: OSPF: EXCHANGE - OPTIONS/INIT not match
 Nov 14 11:51:20.037 ESuT: OSPF: Bad seq received from 92.1.2.20 on
Channel6/0

 Is anyone aware of any other gremlins that cause similar symptoms?  Or any
 other ideas?

 Thanks,
 JMcL




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Re: DQOS course and the CCIE Lab?? [7:57154]

2002-11-08 Thread Steven A. Ridder
I'd recommend knowing QoS in and out for the new lab format, as those are
topics I think Cisco wants you to understand.  As people have been saying
for a while, they took out TR and IPX, but they have to replace it with
something.


Cisco Nuts  wrote in message
news:200211090308.DAA08923;groupstudy.com...
 Hello,Does anyone have any recommendation/comments regarding the DQos
 course from Cisco regarding the CCIE Lab exam? I mean, how much would
 topics out of this course  be covered in the new Lab as of the 4th?
 Topics like Nbar, Diffserv, CBWFQ etc.Is it worth taking the course in
 terms of preparing for the Lab exam? And also, would Qos topics be asked
 in relation to the 3550 switch?Any ideas?Please advise.Thank
you.Sincerely.

 

 MSN 8 helps ELIMINATE E-MAIL VIRUSES. Get 2 months FREE*.




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Re: The 1250$ question [7:56898]

2002-11-05 Thread Steven A. Ridder
I've been studying too hard, because I thought this was a REGEX question.


Greg Nathan  wrote in message
news:200211051415.OAB12451;groupstudy.com...
 Hello fellow ciscoites
 The question should include vat because that is is what cisco charge for
the
 lab, 1500$.
 CCIE topics are well covered by now, and every candidate has a fair idea
of
 what to concentrate his sudies on.
 However, reading through some of the posts on the 3550 and the
speculations
 on topics and features one will be tested on I find little to work on. I
am
 staring at the 3550 on my desk wondering what I should concentrate on
 practising first.
 Any realistic speculations anyone?

 I heard Vlan tunneling, etherchannel etc. But then the magic word: QOS.
This
 can mean quite a lot. Could anyone narrow this down?
 Much appreciated if you could.

 PS
 Being the realist, I see myself carpet bombing all possible topics to have
 any chance of covering the lot. But a bigger degree of focus would really
 help.




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Re: Cisco 3550 study materials and resources [7:56725]

2002-11-04 Thread Steven A. Ridder
I think they may focus on QoS stuff, but we'll see (I'll be prepared for
everything) as I'm taking the new test this Friday.

I know that they'll have 2 Cat 3550's from what they said in the summer, so
I guess trunking, etherchannel and other things like you mention, VLAN
tunneling may pop up as well.  Just as in the Routing and Bridging part,
you'll need to focus on the weird and twisted things they can come up with
with these new switches.

We'll see...

Juan Blanco  wrote in message
news:200211032314.XAA24604;groupstudy.com...
 Chuck,
 Great job, we all appreciate your valuable time on doing this homework
 Keep it up and good luck in your coming LAB

 Juan Blanco

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:nobody;groupstudy.com]On Behalf Of
 The Long and Winding Road
 Sent: Sunday, November 03, 2002 2:39 PM
 To: [EMAIL PROTECTED]
 Subject: Cisco 3550 study materials and resources [7:56725]


 As of this morning, I have been able to verify the following resources for
 Cisco 3550 study and practice. By Monday, not only will these switches be
in
 the CCIE Lab, but test takers will be responsible for all L2 and L3
 functionality, including things like VLAN tunneling, BGP, and a wealth of
 other things. This otter be fun!

 1) Configuration guides and command references on CCO

http://www.cisco.com/univercd/cc/td/doc/product/lan/c3550/12111ea1/index.htm
 watch the wrap

 2) NLI ( www.ccbootcamp.com ) I spoke to Mark Russell. Somewhere on the
site
 there is a free white paper covering 3550 basics. To judge from the web
 site, it appears that 4 of the 5 rental racks have 3550's in them. Mark
also
 said that his package of updated and new scenarios is due real soon now
 I'd like to say in a couple of weeks, but I don't remember if that's
exactly
 what Mark said.

 3) IPExpert ( www.ipexpert.net ) has a new study guide out, which includes
 scenarios with the 3550. The web site says that there are 3550's in the
 rental racks. ( The diagram needs to be updated. )

 4) Hello Computers ( www.hellocomputers.com ) has rental racks that
include
 3550's. They also sell a Lab study book with 24 scenarios and an optional
 rack access purchase, including consulting with a CCIE

 5) There are any number of e - bay auctions of rack rentals. rack rental
 seems to be a going concern these days.

 6) Certification Zone ( www.certificationzone.com ) has announced the
 release of my white paper on the 3550. For a limited time, non subscribers
 may download the two 3550 Lab scenarios with sample configurations that I
 wrote as part of this white paper free of charge. Subscribers get the
white
 paper, the QA, and the labs. ( disclaimer - I was paid to write the Cert
 Zone material, so I have a vested interest in its success )

 Hope this helps.

 Chuck




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Re: Storage Area Networking [7:56857]

2002-11-04 Thread Steven A. Ridder
We in the Cisco world are just entering the SAN arena, but it isn't new
technology.  The only new thing will be iSCSI.  My company is HP and EMC's
largest reseller, so we have been doing this stuff for a while, but it's
brand new to me.  I have been picking everyone's brains the past few months
to understand what all the hubbub is about in the SAN arena.  Here is what I
have learned so far.

The obvious:  First off you need a off disk place to store the data should
the HD fail.  In the beginning there was the tape drive, usually connected
to the same SCSI bus as the hard drives of the server.  Since everything was
SCSI, and local to the server, it was quick and speedy, and you didn't have
to worry about disc timeouts, LUN addressing, or distance etc..  The
limitation was obviously the challenge of managing potentially hundreds of
tape drives.

So someone came out with the idea of creating a large disc system that many
servers could connect to via SCSI.  This offered a more centralized solution
for locally connected servers, but if a large company had many clusters of
servers over a large city, state, country, continent and so on, this
solution couldn't meet that need since the servers still connected to the
central disc system via a SCSI bus.  What was needed was a way to transport
data over a network.  At those times, 10/100 Ethernet was not fast enough,
both because of the 100MB limitation (VS the GB speeds of a local SCSI bus)
and the MTU of Ethernet.  If I tried to transfer even a 512 byte chunk of
data from a SCSI HD to another over Ethernet, the HD would timeout and give
errors.

I think this is where FC came in, with initial speeds of 1 GB and a direct
encapsulation of raw SCSI data, eliminating the timeout issues and the MTU
size, as a raw file could be large than 1500 bytes.  The FC spec also
offered a way to address LUN's on servers.  The only problem I can find with
FC is that there is no standardization as each FC switch vendor offers it's
own flavor of FC, which in turn needs it's own approved FC cards for the
server and each vendor of server/disc system needs to approve it's use.

The next step is iSCSI, which will offer vendor interoperability and
eliminate the separation of IP and FC networks.  On the LAN end, Cisco is
going after Brocade with a new Switch in the 9xxx family (can't remember the
exact name) that, from a technical issue, beats any Brocade switch hands
down (now if only the EMC's, HP's, Hitachi's and IBM's would certify it).
The 9xxx has 128 ports on 1 bus, vs a large brocade that has 32 ports over 2
busses, for a total of 64.  Not only that, the 9xxx switch looks like a Cat
6k, and therefore is modular, and can combine FC/IP/iSCSI all in 1 box.
Cisco hasn't come up with a go-to-market strategy yet, but I have met with
one of the Technical Product Managers at Cisco, and it's coming any day now,
so expect to see Cisco go head to head with Brocade.

That may tackle one issue, but I have other needs where I need Cisco today:

Now the big thing is DR, where I can back up data over WAN's to a remote DR
site.  The problems I am encountering now is two fold:  I can't use a Cisco
WAN router to take FC on LAN end and send over WAN such as a T1 or T3.  I
have customers doing AVVID and storage, but it's over IP, and not FC or
iSCSI.  Cisco is off on the right foot with AVVID, but it needs an S at
the end (S is for storage).  Once I can combine all 4, (from what I can
gather, storage is just another application with it's own needs- *CAN* use a
ton of bandwidth and is latency sensitive like SNA or Video) I can tell
large, LARGE enterprises that we have a great DR solution.  I don't think
that SAN's are for most companies, just the large ones.  The other problem I
have is that none of the Cisco gear is certified, and it doesn't matter how
awesome Cisco's gear is, if the vendors won't certify it, then they will
fail.  If I had to add a third problem, I'd say iSCSI hasn't lived up to
it's hype yet, and there are very few products (servers and disc systems)
out there that offer native iSCSI.

I am not a SAN expert, but I have seen more companies willing to invest in a
SAN than a IP Tel network, so it's a good thing to learn, but not today.


Priscilla Oppenheimer  wrote in message
news:200211050001.AAA21659;groupstudy.com...
 Is anyone using Storage Area Networking? How do you use it? How well does
it
 work? What problems does it solve for you?

 It it really networking, the way we know the term?? It sounds like it's
sort
 of the next generation of file servers, but it also sounds like it's just
a
 new way of managing hard drives.

 I'm having a difficult time figuring out what it is really. Thanks for
 helping me understand it.

 ___

 Priscilla Oppenheimer
 www.troubleshootingnetworks.com
 www.priscilla.com




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Re: Redistributing RIP into OSPF Lab practice [7:56313]

2002-10-26 Thread Steven A. Ridder
THe trick is to see if you can do it in 1 access-list statement.  I think it
can be done in 1.

--

RFC 1149 Compliant.



J B  wrote in message
news:200210252026.UAA12924;groupstudy.com...
 Thanks for the Help

 JB




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Re: redistributing question... [7:56327]

2002-10-25 Thread Steven A. Ridder
without looking at the lab, try summary address at the (or all) ospf asbr's.

--

RFC 1149 Compliant.



Edward Sohn  wrote in message
news:200210260007.AAA12983;groupstudy.com...
 I'm working on Solie's skynet lab...

 If one router (R2) redistributes summarized EIGRP routes (from R5) into
 OSPF (R1,R2,R3,R4), and then OSPF is redistributed into IGRP (R6), how
 do I make the IGRP domain see the EIGRP routes?

 In the /24 mask OSPF domain, the redistributed EIGRP routes show up as a
 /15 mask.  I know this is why they won't go into IGRP, but I don't know
 how to solve the problem without using statics, which I am not allowed
 to do.

 For more info, please see the lab...i can't figure it out using the
 downloaded PDF solutions, either...

 Anyone?

 Thanks,

 Eddie




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Re: Off topic - my first AVVID install [7:56305]

2002-10-25 Thread Steven A. Ridder
I'm gone for a week, and already I'm being attacked :)

--

RFC 1149 Compliant.



The Long and Winding Road  wrote in
message news:200210251844.SAA20015;groupstudy.com...
 Someone a lot smarter than I did the intelligent work - i.e. the call plan
 and the server  configuration. I was one of the warm bodies corralled to
do
 installation of the desk sets.

 Some idle thoughts.
 ( Mr. RFC 1149 Compliant is free to laugh loudly at me and make
denigrating
 comments :-  )

 1) there is no glamour in deploying IP phones. About the only difference
 between deploying a phone and deploying a computer is that phones are a
LOT
 lighter. However, when deploying phones it's still doubly difficult
because
 you end up having to string the PC cable over to the phone ( to get the in
 line power ) and then the phone cable back to the PC.

 2) I was too old for this kind of work 10 years ago, and I'm definitely
too
 old now. My knees hurt. My back huts. And my head hurts. You folks who
crawl
 around under desks and benches to set things up and cable them know
exactly
 how hard those upper surfaces can be ;-

 3) doing this kind of work during business hours is not a real good idea.
It
 was taking neighborhood 15 minutes per station to get a phone deployed. No
I
 did not have the luxury of setting up several phones in an area. Had to do
 it one at a time because of the physical layouts and the user
requirements.

 4) I was overjoyed to finally figure out that it is a lot faster if some
low
 level ( me ) plugged lots of phones directly into a switch, let them go
 through their download and upgrade shenanigans, then hand them out to a
 couple of folks to deploy. If this is done in advance, the process takes
 only a minute or two to register and go through TAPS

 In conclusion, IP telephony intelligence is all in the server, gateway,
and
 router configuration. The phone deployment itself is still monkeywork.




 --

 www.chuckslongroad.info




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Re: Off topic - my first AVVID install [7:56305]

2002-10-25 Thread Steven A. Ridder
I agree that the phone deployment process is monkey work and could be subbed
out for dirt cheap $$, just as long as the unions don't get their hands on
it as they do in the real voice world.

--

RFC 1149 Compliant.



The Long and Winding Road  wrote in
message news:200210251844.SAA20015;groupstudy.com...
 Someone a lot smarter than I did the intelligent work - i.e. the call plan
 and the server  configuration. I was one of the warm bodies corralled to
do
 installation of the desk sets.

 Some idle thoughts.
 ( Mr. RFC 1149 Compliant is free to laugh loudly at me and make
denigrating
 comments :-  )

 1) there is no glamour in deploying IP phones. About the only difference
 between deploying a phone and deploying a computer is that phones are a
LOT
 lighter. However, when deploying phones it's still doubly difficult
because
 you end up having to string the PC cable over to the phone ( to get the in
 line power ) and then the phone cable back to the PC.

 2) I was too old for this kind of work 10 years ago, and I'm definitely
too
 old now. My knees hurt. My back huts. And my head hurts. You folks who
crawl
 around under desks and benches to set things up and cable them know
exactly
 how hard those upper surfaces can be ;-

 3) doing this kind of work during business hours is not a real good idea.
It
 was taking neighborhood 15 minutes per station to get a phone deployed. No
I
 did not have the luxury of setting up several phones in an area. Had to do
 it one at a time because of the physical layouts and the user
requirements.

 4) I was overjoyed to finally figure out that it is a lot faster if some
low
 level ( me ) plugged lots of phones directly into a switch, let them go
 through their download and upgrade shenanigans, then hand them out to a
 couple of folks to deploy. If this is done in advance, the process takes
 only a minute or two to register and go through TAPS

 In conclusion, IP telephony intelligence is all in the server, gateway,
and
 router configuration. The phone deployment itself is still monkeywork.




 --

 www.chuckslongroad.info




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Re: Strange 6509 problems [7:55871]

2002-10-18 Thread Steven A. Ridder
what is the bootflash setting?


Price, Jeffery (TIFPC)  wrote in message
news:200210181305.NAA26473;groupstudy.com...
 All,

 I am hoping that you can help shed some light on a problem we had early
this
 morning.  We lost power to our data center and when the power came back on
 our 3 core 6509 switches came back up with out any configs on them.  We
 restored the configs from backups but the real puzzler is why they lost
the
 configs at all.  Anyone out there ever run in to this kind of problem.

 Thanks

 Jeff




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Re: Would this break the NDA [7:55799]

2002-10-18 Thread Steven A. Ridder
NO.


Thomas Crowe  wrote in message
news:200210171509.PAA16135;groupstudy.com...
 I remember a while back I had the question of which terminal emulator is
 being used for the CCIE lab.  Well after taking the Lab (and yes I was
 honored with an invitation to come back and try again one day soon :-o ) I
 now definitively know the answer to this question.  As I recall others
also
 had this question, in trying to avoid a flaming war, what is the group's
 consensus on this.  Do you feel that it would violate the NDA to disclose
 this information, it doesn't address any of the technical content of the
lab
 (and NO I will not disclose any of those, so please do not ask) so I don't
 feel that it would.  This is simply an effort to help out some people with
 their studying efforts so that they are not wasting time getting
accustomed
 to a new and totally different terminal emulator.

 __

 Thomas Crowe
 Senior Systems Engineer / Senior Architect
 EMC Proven Master Architect
 EMC Proven Master Operator
 CTS Professional Services - Atlanta
 __

 [GroupStudy.com removed an attachment of type text/x-vcard which had a
name
 of Thomas Crowe.vcf]




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Re: Public Internet Access [7:55898]

2002-10-18 Thread Steven A. Ridder
Not sure I understand how you are running your network, but if you deny the
lawyers VLAN from accessing the other VLAN's in your network, you should be
all set.  That way you only have one deny statement to add to each VLAN.  I
think what's throwing me is the 300 line access-list statement.  There's a
ton of solutions out there for you, but you need to be more clear in terms
of describing your internal network.


Robert Edmonds  wrote in message
news:200210181908.TAA09447;groupstudy.com...
 I work for a county government.  As part of building a new courthouse, I
am
 tasked with providing attorneys in courtrooms with Internet access through
 my network.  Of course, I would like to provide them access to what they
 need while blocking access to our internal network.
 My network is setup in the following manner:
 In the new courthouse, the MDF has a 3550-12G acting as the root switch
for
 the building, and has the layer 3 image.  It connects directly to my core,
 with a 6506 with Sup2 and MSFC2, which in turn connects to my PIX 515 for
 Internet access.  I plan on creating a separate VLAN for the public
Internet
 access, but beyond that I'm left a bit short.  Obviously I don't want to
 create a 300 line access-list that would deny them access to each internal
 VLAN, then each of our servers in turn.  Can someone give me some
 suggestions to get this done?  Thanks in advance.




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Re: Public Internet Access [7:55898]

2002-10-18 Thread Steven A. Ridder
I guess policy routing is what I'd recommend, or put a firewall in front of
the servers and set up the appropriate controls.   Policy routing is what
that type of application was inteded for, so you are along the right track,
although it's far from secure.  If security isn't an issue, then check out a
firewall.  If you got the cash, get the firewall blade for the 6500, and
implement the controls there.  Then you have optimal control over all
aspects of the network that pass through it.

Robert Edmonds  wrote in message
news:200210181926.TAA13264;groupstudy.com...
 First, the 300 line access-list was a bit of an exageration, more to make
 the point that I don't want an ungodly long access-list.
 Well, basically every floor in each building has its own /24 subnet.
 Unfortunately the real problem is that to get to the Internet, traffic
must
 traverse VLAN 1, which also houses all my servers.  That's the real
problem.
 Is it possible to force traffic from one VLAN to go only out through my
PIX
 and not be able to browse the servers on that subnet?
 Not being really familiar with the concept, I was thinking along the lines
 of policy routing.  Is this the type of application it is intended for?
I'm
 still trying to find good information on it.
 Steven A. Ridder  wrote in message
 news:200210181920.TAA12300;groupstudy.com...
  Not sure I understand how you are running your network, but if you deny
 the
  lawyers VLAN from accessing the other VLAN's in your network, you should
 be
  all set.  That way you only have one deny statement to add to each VLAN.
 I
  think what's throwing me is the 300 line access-list statement.  There's
a
  ton of solutions out there for you, but you need to be more clear in
terms
  of describing your internal network.
 
 
  Robert Edmonds  wrote in message
  news:200210181908.TAA09447;groupstudy.com...
   I work for a county government.  As part of building a new courthouse,
I
  am
   tasked with providing attorneys in courtrooms with Internet access
 through
   my network.  Of course, I would like to provide them access to what
they
   need while blocking access to our internal network.
   My network is setup in the following manner:
   In the new courthouse, the MDF has a 3550-12G acting as the root
switch
  for
   the building, and has the layer 3 image.  It connects directly to my
 core,
   with a 6506 with Sup2 and MSFC2, which in turn connects to my PIX 515
 for
   Internet access.  I plan on creating a separate VLAN for the public
  Internet
   access, but beyond that I'm left a bit short.  Obviously I don't want
to
   create a 300 line access-list that would deny them access to each
 internal
   VLAN, then each of our servers in turn.  Can someone give me some
   suggestions to get this done?  Thanks in advance.




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Re: run VoIP on a frame network at BIR instead of CIR rates [7:55833]

2002-10-17 Thread Steven A. Ridder
This was Cisco's old theory.  In theory, it would work, but in reality, if
the frame switch saw a packet come into it's ingress interface with the
packet already marked DE, it will drop it because it was unexpected.

I asked the telco's your question last year and that's the answer they gave
me.  Cisco seems to have abandoned that theory a while ago, which is
probably why you haven't seen it written anywhere.


dj  wrote in message
news:200210171534.PAA26762;groupstudy.com...
 Running a VoIP application over a frame-relay network with 256k CIR and
 512k BIR.  From the LLQ docs I reviewed, to guarantee good voice
 quality, traffic shaping all frame traffic to CIR is recommended along
 with LLQ of voice packets.

 Would like to take advantage of BIR bandwidth and still guarantee voice
 packets are not dropped by the frame relay switch network when
 congestion occurs.  Here are my thoughts:

 What if the router were to pre-mark all data packets as Discard
 Eligible (DE) on the outbound serial interface connected to the frame
 network.  Voice packets would NOT be marked DE.  Then run up to BIR
 rates with LLQ prioritization for voice. Would the carrier frame network
 switches drop only the pre-marked DE data packets (by the router) when
 congestion occurred and NOT drop any voice packets?  I haven't found any
 Cisco links that addressed QOS in this fashion.  Any links on this topic
 would be greatly appreciated.

 The objective is to squeeze more bandwidth (BIR vs CIR) out of your
 frame relay network without dropping any voice packets. Why would this
 not work and what are the caveats?

 regards,
 dj




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Re: CallManager query to Win2k Active Directory [7:55789]

2002-10-17 Thread Steven A. Ridder
ONe place is the corporate directory, which is usually in the DC direcrotry.
YOu get that by clicking on the directory button.  Is that what you are
talking about, or are you talking about personal directory, or the AD
plugin, or the Exchange PAB plug-in?

If it's what I think it is, the Active Directory, you probably have to run
the Active Directory Plug in again:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a0080094493.shtml




Jay Dunn  wrote in message
news:200210170828.IAA04931;groupstudy.com...
 I have inherited responsibility for our IP phone system and am using OJT
 to figure everything out. We are using CallManager 3.2 and receiving our
 directory user lists from our win2k AD. The tech that originally set
 this up created separate OUs in AD for onsite and offsite personnel.
 CallManager only queries the onsite OU for our user directory list. A
 user's phone extension is looked up in the telephone number field in
 the user's AD profile. I now have reason to change the OU hierarchy in
 AD. I would also like to change the field where CallManager looks up a
 user's extension. Could someone point me in the right direction for
 determining where these queries are configured? I've examined the system
 parameters and the ASP pages referenced in the directory URL as well as
 the registry on the CCM server. I've also run the AD plug-in, but I'm
 stumped.



 Thanks..



 Jay Dunn

 IPI*GrammTech, Ltd.

 www.ipi-gt.com

 Nunquam Facilis Est




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Re: NTP server again !!!!!!!!!!!!!! [7:55836]

2002-10-17 Thread Steven A. Ridder
I think you have to be an NTP server, as I don't think a workstation can
peer with you.


 wrote in message
news:200210172226.WAA27043;groupstudy.com...
 Hi,

 I am trying to configure my NTP server on the cisco 7505 router.

 The configuration which I did is as follows:

 router#ntp master 10
 router#ntp peer 192.168.0.72

 192.168.0.72 is the address of the Windows 2000 client which I am using.

 I am getting the following o/p for  sh ntp associations

 router#sh ntp associations
   address   ref clockst   when  poll reach delay  offset  disp
 *~127.127.7.1  127.127.7.1   9 21   64377   0.0   0.0016000
  ~192.168.0.72  0.0.0.0  16-1024   00.0   0.0016000

 I don't know why my client(windows 2000) is not getting synced?

 I also tried to connect a Solaris machine and the result is the same.It
 seems that ntp is not getting broadcasted from cisco router.

 I am not using authentication and access lists.Just two commands as shown
 above.Is that enough or something else is required at the router end.

 I am sure that something else is wrong in my config

 It will be greatful if anybody can throw some light into this.

 Thanks,

 Jay


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Re: VoIP Clarification. [7:55682]

2002-10-16 Thread Steven A. Ridder

The CM uses the MAC as a unique identifier in it's SQL database.  It's
actually a distorted version of the MAC, such as a phone's identifier -
SEP003094C26105
--

RFC 1149 Compliant.



Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Great answer. Finally an explanation that makes sense for the marketing
 babble about IP Telephony making Moves, Adds, and Changes easier. ;-)

 One quesiton though, does CallManager really care about MAC addresses?
 Unless the receiving phone is on the same network segment as the calling
 phone, the MAC address won't help matters. ARP would take care of getting
 the MAC when it's needed.

 Priscilla


 Bruce Enders wrote:
 
  B. J.
  The only trick here is to remember that the User phone number
   is
  mapped to the MAC address and IP address of the ethernet
  interface
  associated with the hard phone, or the laptop in the case of
  Softphone.
  (Both are PCs running specific applications software). Whenever
  either is
  disconnected from the network long enough for link to drop,
  they have to
  check in with DHCP when they are re-connected to the network.
  Both also
  have to check in with their CallManager. During that process,
  they
  identify themselves using their MAC address, and announce their
  current
  IP address. After that, the CM can simply forward based on the
  IP
  address. This capability is one of the primary reasons that
  Moves, Adds,
  and Changes in an IP Telephony system are far more simple than
  in a
  legacy PBX environment. (The logic behind your response sounds
  like it
  comes from the legacy telephone world, which is very used to
  working in a
  very static addressing environment).
  Bruce
 
  B.J. Wilson wrote:
 
Hi Vance -
 
I too am studying All Things VoIP, and I'm curious how
  this would work.
Say you have User A trying to call User B.  User B is
  currently in the
office.  So User A dials '' which is User B's phone
  number (or route
pattern if you want to be specific).  CallManager picks up
  the route
pattern, looks up User B's location, and forwards the call
  on.  All is good.
Now, say User B is telecommuting.  How does CallManager
  know this?  How
does your RAS (remote access) server notify CM that User B's
  geographical
location has moved?  Is there something in User B's RAS
  (Registration,
Admission and Status) setup that alerts CM to the fact that
  they're dialing
in from home?
 
Thanks,
 
BJ
 
- Original Message -
From: Vance Krier
To:
Sent: Wednesday, October 16, 2002 4:08 AM
Subject: Re: VoIP Clarification. [7:55682]
 
  Hey Stu,
 
  In simple terms, yes you are correct.  However, as I'm sure
  you know, you
  need to take this type of setup with a grain of salt.  If
  you have a
 
decent
 
  bandwidth, low latency, consistent connection between the
  phone and CM, it
  works fine.   There's absolutely no guarantees for QoS on
  the Internet.
  Now, FWIW, I use softphone on my laptop when I travel and
  I've gotten
  satisfactory results (IMO) better than 75% of the time.
 
  I always pitch this as being a *kewl* feature, but never as
  a selling
 
point.
 
  I'm
  very, very cautious with customers over this.  As long as
  the user
  using it is understanding and realizes there will be times
  when it doesn't
  work or the quality is really crappy, then typically they
  stay happy.  Not
  something I'd give to Internet/computer/technology
  illiterate executive.
 
  I love it, by the way.
 
  Good luck,
  Vance
 
  Stuart Pittwood  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 
Good Morning all,
 
I am just starting to look into VoIP as I have been asked
  by my manager
 
to
 
do some research and find out if there are any benifits
  from VoIP for
 
our
 
firm.
 
Am I right in saying that if we had a solution based on
  Cat 6000 (or
similar) switches, with a cisco VPN solution for the home
  workers, that
users who use their laptop at home with cisco softphone
  or hardware
 
phone
 
could have their telephone extenstion follow them?
 
Please forgive the simplicity of my question, just making
  sure I am
 
  thinking
 
along the right lines.
 
Thanks
 
Stu
  --
 
Bruce Enders   Email:
  [EMAIL PROTECTED]  Chesapeake NetCraftsmen
  o:(410)-280-6927, c:(443)-994-0678
1290 Bay Dale Drive, Suite 312 WWW:
  http://www.netcraftsmen.net  Arnold, MD 21012-2325
  Cisco CCSI# 96047
   Efax 443-331-0651




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Re: VoIP Clarification. [7:55682]

2002-10-16 Thread Steven A. Ridder

(Didn't come through in last one.)

SEP003094C26105

The SEP stands for Selius Ethernet Phone, and the numbers are the MAC
address.  A gateway has a different 3 letter code, can't rememner it though,
and it all depends on the protocol it uses, such as mgcp or h.323, as the
latter doesn't have identifiers.

--

RFC 1149 Compliant.



Steven A. Ridder  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 The CM uses the MAC as a unique identifier in it's SQL database.  It's
 actually a distorted version of the MAC, such as a phone's identifier -
 SEP003094C26105
 --

 RFC 1149 Compliant.



 Priscilla Oppenheimer  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Great answer. Finally an explanation that makes sense for the marketing
  babble about IP Telephony making Moves, Adds, and Changes easier. ;-)
 
  One quesiton though, does CallManager really care about MAC addresses?
  Unless the receiving phone is on the same network segment as the calling
  phone, the MAC address won't help matters. ARP would take care of
getting
  the MAC when it's needed.
 
  Priscilla
 
 
  Bruce Enders wrote:
  
   B. J.
   The only trick here is to remember that the User phone number
    is
   mapped to the MAC address and IP address of the ethernet
   interface
   associated with the hard phone, or the laptop in the case of
   Softphone.
   (Both are PCs running specific applications software). Whenever
   either is
   disconnected from the network long enough for link to drop,
   they have to
   check in with DHCP when they are re-connected to the network.
   Both also
   have to check in with their CallManager. During that process,
   they
   identify themselves using their MAC address, and announce their
   current
   IP address. After that, the CM can simply forward based on the
   IP
   address. This capability is one of the primary reasons that
   Moves, Adds,
   and Changes in an IP Telephony system are far more simple than
   in a
   legacy PBX environment. (The logic behind your response sounds
   like it
   comes from the legacy telephone world, which is very used to
   working in a
   very static addressing environment).
   Bruce
  
   B.J. Wilson wrote:
  
 Hi Vance -
  
 I too am studying All Things VoIP, and I'm curious how
   this would work.
 Say you have User A trying to call User B.  User B is
   currently in the
 office.  So User A dials '' which is User B's phone
   number (or route
 pattern if you want to be specific).  CallManager picks up
   the route
 pattern, looks up User B's location, and forwards the call
   on.  All is good.
 Now, say User B is telecommuting.  How does CallManager
   know this?  How
 does your RAS (remote access) server notify CM that User B's
   geographical
 location has moved?  Is there something in User B's RAS
   (Registration,
 Admission and Status) setup that alerts CM to the fact that
   they're dialing
 in from home?
  
 Thanks,
  
 BJ
  
 - Original Message -
 From: Vance Krier
 To:
 Sent: Wednesday, October 16, 2002 4:08 AM
 Subject: Re: VoIP Clarification. [7:55682]
  
   Hey Stu,
  
   In simple terms, yes you are correct.  However, as I'm sure
   you know, you
   need to take this type of setup with a grain of salt.  If
   you have a
  
 decent
  
   bandwidth, low latency, consistent connection between the
   phone and CM, it
   works fine.   There's absolutely no guarantees for QoS on
   the Internet.
   Now, FWIW, I use softphone on my laptop when I travel and
   I've gotten
   satisfactory results (IMO) better than 75% of the time.
  
   I always pitch this as being a *kewl* feature, but never as
   a selling
  
 point.
  
   I'm
   very, very cautious with customers over this.  As long as
   the user
   using it is understanding and realizes there will be times
   when it doesn't
   work or the quality is really crappy, then typically they
   stay happy.  Not
   something I'd give to Internet/computer/technology
   illiterate executive.
  
   I love it, by the way.
  
   Good luck,
   Vance
  
   Stuart Pittwood  wrote in message
   [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  
 Good Morning all,
  
 I am just starting to look into VoIP as I have been asked
   by my manager
  
 to
  
 do some research and find out if there are any benifits
   from VoIP for
  
 our
  
 firm.
  
 Am I right in saying that if we had a solution based on
   Cat 6000 (or
 similar) switches, with a cisco VPN solution for the home
   workers, that
 users who use their laptop at home with cisco softphone
   or hardware
  
 phone
  
 could have their telephone extenstion follow them?
  
 Please forgive the simplicity of my question, just making
   sure I am
  
   th

Re: Cisco ExecNet [7:55573]

2002-10-15 Thread Steven A. Ridder

I think wireless and converged data over high speed links wil co-exist, not
compete for same space n market.  I can't see high-speed wireless out in the
WAN of a cellular network anywhere down the road.  Without that speed over
wireless, we are stuck with being able to DL e-mails and web-pages at a
slow, but decent rate.  The high speed stuff will happed over wires for a
while, and although I don't see PC's being used as TV's, I do forsee the PC
being the digital gateway/servwer of the high-speed home where other devices
like a TIVO work off of the gateway and provide TV services to the family
and a phone will be a phone, just getting it's information form same gateway
and the phone will provide the phone services for a family.

Our consulting side does see wireless devices with two bands - 802.b/a/g for
use in hot spots and GSM/GRPS over the WAN, and this is going to be the
way of wireless for a while.  While your at a hotspot, maybe a hotel or
airport (or Starbuck now, which we helped developed for them) you can get
high speeds and DL video, maybe play a java game with a buddy.  Then you
have to leave the area, and now you rely on GSM.  You still have
connectivity, but in a limited fashion.  I work for a company that tests,
writes, and demos the latest devices from that carriers, and so I get to
play with them as well, and I have seen a lot of innovative devices, (right
now I get a T-Mobile Pocket PC Phone Edition as my cell phone) and I love
them, but what I'm seeing is not the devleopment of bandwidth over their
networks, but the 2.5G network development, and the standardization of the
network with 1 common signal.

--

RFC 1149 Compliant.



The Long and Winding Road  wrote in
message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Steven A. Ridder  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  I always thought that the PSTN was based off of that fact that not all
  phones would be calling at once, and if they did, then some would get
  through while others wouldn't.

 CL: yes. true. however, decades ago the Bell folks knew and practiced the
 optimum manner in which to provision such that you or I or any other
 individual would experience dial tone almost all of the time. We know this
 through the Ehrlang calcualtions.


 Then to ensure that important calls got
  through during these periods, there was the priority network that gov't
  officials have with their PINS, etc.  (Can't remember the name, but
 there's
  also an IETF working group working on the same thing.)

 CL good idea. having been through an earthquake or two, I'm quite familiar
 with fast busy's during emergencies. nice to know there is a means for the
 right people to be able to get through.


 
  I don't think that the Converged Network theory is reinventing the wheel
 and
  is a dead end.  I think the opposite is true.  The TDM/PSTN world is
dead
  (or dying) and that most calls are circuit-switched across ATM now.

 CL: different issue. the Bell network grew and matured because of
regulation
 that guaranteed return on capital. therefore it was in Bell's interest to
 invest in capital - switches, lines, CO's. Since deregulation in 1984 it
can
 be argued that the appropriate investment has not been made in the
 network - all that has happened is that the CLECs have cherry picked the
 most concentrated and profitable areas while underinvesting in not so
 profitable areas. I sometimes sign my messages TANSTAAFL - there ain't no
 such thing as a free lunch. Right now, for all intents and purposes, the
 internet is free. What happens when people have to start paying for
their
 proportional share of services? Assuming the internet becomes the
 replacement for the telco netowrk?

 CL: I'm not saying that there is room for improvement. There is no reason
 that a PBX has to be larger than a couple of IBM mainframes. But I gotta
 ask - is it really a good idea to make your PC into a telephone into a
 television?


  Now if
  someone could just solve the last mile
 

 CL: oh boy. video on demand. OC192 to the television set. I can hardly
wait.

 CL: much as I despise the idea, I go along with the school of thought that
 wireless is the future, not voice and data converged. It's another one of
 those trekkie tech things, but telcos continue to lose 10's of thousands
 of lines per year to wireless, and most people just want to yak on the
 phone, no matter where they are. Which is one more reason to telecommute.



  --
 
  RFC 1149 Compliant.
 
 
 
  The Long and Winding Road  wrote in
  message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   Steven A. Ridder  wrote in message
   [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
I understand the technology and stand by whoever said what IP
   telephony/VoIP
isn't a bandwidth hungry app.  It isn't.  G.729, which can use as
 little
   as
8k with proper compresion, has nearly the same MOS score as G.711,
 which
   is
toll quality. 

Re: Cisco ExecNet [7:55573]

2002-10-14 Thread Steven A. Ridder

I understand the technology and stand by whoever said what IP telephony/VoIP
isn't a bandwidth hungry app.  It isn't.  G.729, which can use as little as
8k with proper compresion, has nearly the same MOS score as G.711, which is
toll quality.  Even though it's not officially toll quality I consider it
toll quality, as I can't tell the difference, and most people couldn't
either.  Even if using G.711, I can still use compression and VAD to get
down to 25K or so, which isn't bandwidth hungry in my book either.

I think the apps that will be on a converged network in the future will be
bandwidth hungry, such as video. Voice isn't.



 -Original Message-
 From: Joe A
 To: 'Nathan Chessin'; 'Albert Lu'; [EMAIL PROTECTED]
 Cc: [EMAIL PROTECTED]
 Sent: 10/14/02 11:52 AM
 Subject: RE: Cisco ExecNet

 Maybe I should say IP Telephony, not VoIP.  How many uncompressed,
 toll-quality calls can you push out simultaneously over a T1???  Have
 you done the math? 24?   Maybe 23 on a good day.  Sure, if you use
 compression you can squeeze in quite a bit more, but you can't deny that
 IPT is bandwidth-hungry, with streaming MOH, voicemail audio streams,
 the calls themselves.  Believe me, VoIP is absolutely a bandwidth-hungry
 app.  No one who understands the technology would deny that.

 Joe

 -Original Message-
 From: Nathan Chessin [mailto:[EMAIL PROTECTED]]
 Sent: Friday, October 11, 2002 1:56 AM
 To: 'Joe'; 'Albert Lu'; [EMAIL PROTECTED]
 Cc: [EMAIL PROTECTED]
 Subject: RE: Cisco ExecNet


 1) Since when is VoIP a bandwidth-hungry app

 Nate

  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of

  Joe
  Sent: Thursday, October 10, 2002 8:42 PM
  To: 'Albert Lu'; [EMAIL PROTECTED]
  Cc: [EMAIL PROTECTED]
  Subject: RE: Cisco ExecNet
 
 
  Technology isn't necessarily heading in that direction - Cisco is
  driving it there.  Bottom line is this: Cisco is traditionally a
  router and switch manufacturer, and no one buys routers and switches
  these days, at least not enough to provide continued growth for Cisco.
  Company infrastructures are already built, have been for
  years, and are
  running for the most part nowhere near capacity.  These technology
  applications, besides generating hardware sales directly, will also
  increase bandwidth consumption, thereby causing indirect
  hardware sales
  when customers upgrade their routers and switches to support the new
  bandwidth-hungry apps like VoIP.  If Cisco can drive the customers'
  purchases in that direction, they win.
 
  My two cents.
 
  Joe
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On
  Behalf Of
  Albert Lu
  Sent: Thursday, October 10, 2002 8:16 AM
  To: [EMAIL PROTECTED]
  Cc: [EMAIL PROTECTED]
  Subject: OT: Cisco ExecNet
 
 
  Hello Group,
 
  Has anyone checked out the Cisco ExecNet, which is basically thoughts
  about where technology is heading in the future from the VPs at Cisco.
 
  http://newsroom.cisco.com/dlls/tln/execnet/
 
  From what they are saying (specifically Mike Volpi), the
  direction for
  technology is heading towards: CDN, Security, Wireless, IP Telephony,
  VPN. Reegineering business processes to best utilise these
  technologies in order to improve productivity and reduce cost for
  enterprises.
 
  Does anyone have any comments about this, and where money
  will be spent
  in the future for technologies?
 
  Regards,
 
  Albert Lu
  CCIE #8705




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Re: VoIP QoS [7:55597]

2002-10-14 Thread Steven A. Ridder

LLQ would be his best option, not WFQ.  If he is using it, that's probably
his issue.

--

RFC 1149 Compliant.



lamb stephen  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Group,
Hoping that someone can help me out with a VoIP QoS issue that I am
 currently dealing with. I work for a service provider, and I am currently
 troubleshooting a VoIP over frame relay quality complaint. My end user has
a
 768K host with four 256K drops dedicated solely to VoIP traffic. My
customer
 states that he experiences intermittent jitter on his calls, but they
follow
 no real pattern. We have had his vendor place test calls, and sometimes 7
 simultaneous calls can
 go through fine while 3 simultaneous calls will experience poor call
quality
 and excessive jitter. The end user's vendor is of no real help with this
 issue stating that his configurations are fine and the trouble must be
with
 the WAN link.
I have verified that the entire network is clean, no T1 performance
 monitor errors , no input errors on the customer's serial interfaces, and
no
 input errors to my frame switch. No apparent utilization issues, the host
 averaged 50% port utilization during a 24 hour sniff. We have also
verified
 the drops are not receiving any FECNs or BECNs. I have a copy of the
 customer's router
 configurations and his map-class statements appear to be correct as well.
 His CIR and MINCIR are set to match the frame relay PVC CIR in my network
 (which I believe means that he has configured the statements to prevent
any
 bursting, please correct me if I am wrong).
On to my question. The only discrepancy I find with this customer's
 configuration is his queuing. On all four of his drop routers he has
 configured WFQ, on his host he has no queuing specified. Could this be the
 cause of all of his problems? Would WFQ be the most desirable method? What
I
 have read in the past led me to believe that a fragment statement in the
 map-class was the most
 desirable because it activated the dual-FIFO feature on the physical
 interface. I do not have a great deal of experience with VoIP so all I
have
 to go on right now are theories. Any direction is greatly appreciated.

 Thanks,
 Steve Lamb
 CCDA, CCNA




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Re: suppress-map with summary-only?? [7:55599]

2002-10-14 Thread Steven A. Ridder

I think the aggregate address has to be in your routing table first.
Someone please correct me if i'm wrong, as I'm trying to get it right from
memory.

--

RFC 1149 Compliant.



Cisco Nuts  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hello,

 Does the suppress-map work along with the summary-only keyword?

 I would only like to see the summary 13.0.0.0/8 but I keep seeing the rest
 of the networks.

 Here is the config:

 R7-FR(config)#aggregate-address 13.0.0.0 255.0.0.0 suppress-map CHECK
 summary-only

 route-map CHECK permit 10
 match ip address 21

 access-list 21 permit 13.4.0.0 0.0.255.255
 access-list 21 deny   any

 This works as it should.denies netw 13.4.0.0/16 and permits the rest,
 13.1.0.0/24, 13.2.1.0/24, 13.3.0.0/16 and 13.0.0.0 BUT I would only like
to
 see the aggregate 13.0.0.0/8

 Am I even asking the right thing here? :-)
 Just checking.

 Thank you.

 Sincerely.











 _
 MSN Photos is the easiest way to share and print your photos:
 http://photos.msn.com/support/worldwide.aspx




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Re: QoS and CBWFQ [7:55546]

2002-10-14 Thread Steven A. Ridder

THe only reason I can see using QoS is to limit traffic to certain amouts of
BW.  Even then it's tricky becasuse in CBWFQ, you are guaranteing a minimum,
not a maximum amout of BW for a class.  You could police certain classes of
traffic to never exceed a BW, but that can be crummy as well, espcially if
there isn't congestion.

--

RFC 1149 Compliant.



Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 John Neiberger wrote:
 
  JM  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   I have Internet connection to IPS and I don't know what QoS
  politic to
   choose.
   I have now  4 classes and CBWFQ:
   gold ( SMTP , POP 3) etc.
   silver HTTP
   bronse ( FTP)
   default ( fail-queue)
   and service-policy out.
   Is there any sense to use CBWFQ on a serial interface like
  service-policy
  in
   ?
   My ISP can't mark or shape  my traffic.
   What is it the best QoS solusion for input traffic ?
   Thanks a lot.
   JM
 
  Others might disagree but I see no advantage whatsoever to
  using QoS
  mechanisms on the link going to your ISP.

 Would it depend on the bandwidth of his circuit? WFQ is on by default for
 speeds of E1 and less. Perhaps that's all he needs if he has a low-speed
 circuit. He probably doesn't need anything special if the circuit is
higher
 speed. For low-speed, he could at least prioritize the order of packets
sent
 (and possibly dropped) by his own router.

 He should check the circuit speed and load to see if he needs to do
anything.

 Also, it would be silly to make SMTP and POP3 highest priority in many
 environments. Is there a local e-mail server for SMTP and POP3? If yes,
the
 clients are sending and receiving locally. The server also sends SMTP
 traffic to servers on the Internet probably, but if that gets congested,
the
 server will simply try again. There's no user waiting around for this. In
 most cases, server-to-server delays aren't noticeable by users.

 But if the e-mail server is offsite, then maybe it makes sense to
prioritize
 SMTP and POP3.

 ___

 Priscilla Oppenheimer
 www.troubleshootingnetworks.com
 www.priscilla.com

  Once you hand off
  traffic to them
  you're completely at their disposal.  You have no control over
  traffic
  within their network so why even bother adding queueing to your
  outgoing
  interface?  If your link is congested often enough that you
  feel it's
  necessary I'd suggest getting another circuit installed, if
  that's possible.
 
  Incoming I'd think that CAR would be useful depending on what
  you're really
  trying to accomplish. It would at least allow you to classify
  traffic based
  on your own criteria and then mark it for special handling
  within your
  network.  All of this really depends on your specific situation
  and your
  goals.
 
  John




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Re: VoIP Interoperability [7:55523]

2002-10-14 Thread Steven A. Ridder

I think you need to look at analog/digital gateways, and how they interact
with a trunk connection on the different systems.  At least thats how you
tackle the problem from the cisco side.  Or maybe the other VoIP solutions
are H.323 compliant, and maybe you can get lucky and try that approach. I'm
going to guess that other than H.323, the boxes don;t communicate very well.
I'd recommend looking at the gateway solution.

--

RFC 1149 Compliant.



Steve Watson  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 We are beginning deployment of a multi-vendor VoIP solution. I could go
 on and on about WHY we are pursuing multi-vendor but then we would spend
 a day or so answering that question (which isn't even in question).

 What I need to know is there any sites or good books that talk about
 multi-vendor VoIP integration, specifically the call processing boxes
 themselves? I need to know what VoIP PBX's (if you will) will talk to
 others. Specifically looking at Alcatel, Cisco and Inter-tel.

 Thanks for any info,

 Steve




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Re: Cisco ExecNet [7:55573]

2002-10-14 Thread Steven A. Ridder

I always thought that the PSTN was based off of that fact that not all
phones would be calling at once, and if they did, then some would get
through while others wouldn't.  Then to ensure that important calls got
through during these periods, there was the priority network that gov't
officials have with their PINS, etc.  (Can't remember the name, but there's
also an IETF working group working on the same thing.)

I don't think that the Converged Network theory is reinventing the wheel and
is a dead end.  I think the opposite is true.  The TDM/PSTN world is dead
(or dying) and that most calls are circuit-switched across ATM now.  Now if
someone could just solve the last mile

--

RFC 1149 Compliant.



The Long and Winding Road  wrote in
message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Steven A. Ridder  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  I understand the technology and stand by whoever said what IP
 telephony/VoIP
  isn't a bandwidth hungry app.  It isn't.  G.729, which can use as little
 as
  8k with proper compresion, has nearly the same MOS score as G.711, which
 is
  toll quality.  Even though it's not officially toll quality I consider
 it
  toll quality, as I can't tell the difference, and most people couldn't
  either.  Even if using G.711, I can still use compression and VAD to get
  down to 25K or so, which isn't bandwidth hungry in my book either.
 
  I think the apps that will be on a converged network in the future will
be
  bandwidth hungry, such as video. Voice isn't.
 

 CL: I don't think the issue is the bandwidth taken by one compressed call.
 The issue is poisson 99. I think that's how the telco guys call it. What
 happens when a significant number of calls must go through - say during
an
 emergency?

 CL: current telco networks are engineered such that you get dial tone
99.5%
 of the time you go off hook, day or night, busy hour or not. the VoIP
 netowork must not only operate at that kind of reliability, but must
 tramsmit data simultaneously.

 CL: This rush to converged networks means not only reinventing what the
 telcos have already done, but building out a whole new infrastructure as
 well. There is at least one school of thought that calls this a dead end.

 CL: one of the bad things that has come out of Microsoft is the attitude
 that Mainframe computers are just PC's with a little bit more horsepower
and
 that the internet is just a bigger version of the Microsoft campus
network,
 with a few more hubs involved. I see one of the bad things about Cisco's
 vision of converged networks is the attitude that the Telephone Network is
 nothing more than just the Cisco campus telephone network with a few more
 phones attached.




 
  
   -Original Message-
   From: Joe A
   To: 'Nathan Chessin'; 'Albert Lu'; [EMAIL PROTECTED]
   Cc: [EMAIL PROTECTED]
   Sent: 10/14/02 11:52 AM
   Subject: RE: Cisco ExecNet
  
   Maybe I should say IP Telephony, not VoIP.  How many uncompressed,
   toll-quality calls can you push out simultaneously over a T1???  Have
   you done the math? 24?   Maybe 23 on a good day.  Sure, if you use
   compression you can squeeze in quite a bit more, but you can't deny
that
   IPT is bandwidth-hungry, with streaming MOH, voicemail audio streams,
   the calls themselves.  Believe me, VoIP is absolutely a
bandwidth-hungry
   app.  No one who understands the technology would deny that.
  
   Joe
  
   -Original Message-
   From: Nathan Chessin [mailto:[EMAIL PROTECTED]]
   Sent: Friday, October 11, 2002 1:56 AM
   To: 'Joe'; 'Albert Lu'; [EMAIL PROTECTED]
   Cc: [EMAIL PROTECTED]
   Subject: RE: Cisco ExecNet
  
  
   1) Since when is VoIP a bandwidth-hungry app
  
   Nate
  
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf
Of
  
Joe
Sent: Thursday, October 10, 2002 8:42 PM
To: 'Albert Lu'; [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: RE: Cisco ExecNet
   
   
Technology isn't necessarily heading in that direction - Cisco is
driving it there.  Bottom line is this: Cisco is traditionally a
router and switch manufacturer, and no one buys routers and switches
these days, at least not enough to provide continued growth for
Cisco.
Company infrastructures are already built, have been for
years, and are
running for the most part nowhere near capacity.  These technology
applications, besides generating hardware sales directly, will also
increase bandwidth consumption, thereby causing indirect
hardware sales
when customers upgrade their routers and switches to support the new
bandwidth-hungry apps like VoIP.  If Cisco can drive the customers'
purchases in that direction, they win.
   
My two cents.
   
Joe
   
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On
Behalf Of
Albert Lu
Sent: Thursday, October 10, 2002 8:16 AM
To: 

Re: Cisco ExecNet [7:55573]

2002-10-14 Thread Steven A. Ridder

I remeber what that emergency network is:  GETS.

--

RFC 1149 Compliant.



Steven A. Ridder  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I always thought that the PSTN was based off of that fact that not all
 phones would be calling at once, and if they did, then some would get
 through while others wouldn't.  Then to ensure that important calls got
 through during these periods, there was the priority network that gov't
 officials have with their PINS, etc.  (Can't remember the name, but
there's
 also an IETF working group working on the same thing.)

 I don't think that the Converged Network theory is reinventing the wheel
and
 is a dead end.  I think the opposite is true.  The TDM/PSTN world is dead
 (or dying) and that most calls are circuit-switched across ATM now.  Now
if
 someone could just solve the last mile

 --

 RFC 1149 Compliant.



 The Long and Winding Road  wrote in
 message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Steven A. Ridder  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   I understand the technology and stand by whoever said what IP
  telephony/VoIP
   isn't a bandwidth hungry app.  It isn't.  G.729, which can use as
little
  as
   8k with proper compresion, has nearly the same MOS score as G.711,
which
  is
   toll quality.  Even though it's not officially toll quality I
consider
  it
   toll quality, as I can't tell the difference, and most people couldn't
   either.  Even if using G.711, I can still use compression and VAD to
get
   down to 25K or so, which isn't bandwidth hungry in my book either.
  
   I think the apps that will be on a converged network in the future
will
 be
   bandwidth hungry, such as video. Voice isn't.
  
 
  CL: I don't think the issue is the bandwidth taken by one compressed
call.
  The issue is poisson 99. I think that's how the telco guys call it. What
  happens when a significant number of calls must go through - say
during
 an
  emergency?
 
  CL: current telco networks are engineered such that you get dial tone
 99.5%
  of the time you go off hook, day or night, busy hour or not. the VoIP
  netowork must not only operate at that kind of reliability, but must
  tramsmit data simultaneously.
 
  CL: This rush to converged networks means not only reinventing what the
  telcos have already done, but building out a whole new infrastructure as
  well. There is at least one school of thought that calls this a dead
end.
 
  CL: one of the bad things that has come out of Microsoft is the attitude
  that Mainframe computers are just PC's with a little bit more horsepower
 and
  that the internet is just a bigger version of the Microsoft campus
 network,
  with a few more hubs involved. I see one of the bad things about Cisco's
  vision of converged networks is the attitude that the Telephone Network
is
  nothing more than just the Cisco campus telephone network with a few
more
  phones attached.
 
 
 
 
  
   
-Original Message-
From: Joe A
To: 'Nathan Chessin'; 'Albert Lu'; [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: 10/14/02 11:52 AM
Subject: RE: Cisco ExecNet
   
Maybe I should say IP Telephony, not VoIP.  How many uncompressed,
toll-quality calls can you push out simultaneously over a T1???
Have
you done the math? 24?   Maybe 23 on a good day.  Sure, if you use
compression you can squeeze in quite a bit more, but you can't deny
 that
IPT is bandwidth-hungry, with streaming MOH, voicemail audio
streams,
the calls themselves.  Believe me, VoIP is absolutely a
 bandwidth-hungry
app.  No one who understands the technology would deny that.
   
Joe
   
-Original Message-
From: Nathan Chessin [mailto:[EMAIL PROTECTED]]
Sent: Friday, October 11, 2002 1:56 AM
To: 'Joe'; 'Albert Lu'; [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: RE: Cisco ExecNet
   
   
1) Since when is VoIP a bandwidth-hungry app
   
Nate
   
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On
Behalf
 Of
   
 Joe
 Sent: Thursday, October 10, 2002 8:42 PM
 To: 'Albert Lu'; [EMAIL PROTECTED]
 Cc: [EMAIL PROTECTED]
 Subject: RE: Cisco ExecNet


 Technology isn't necessarily heading in that direction - Cisco is
 driving it there.  Bottom line is this: Cisco is traditionally a
 router and switch manufacturer, and no one buys routers and
switches
 these days, at least not enough to provide continued growth for
 Cisco.
 Company infrastructures are already built, have been for
 years, and are
 running for the most part nowhere near capacity.  These technology
 applications, besides generating hardware sales directly, will
also
 increase bandwidth consumption, thereby causing indirect
 hardware sales
 when customers upgrade their routers and switches to support the
new
 bandwidth-hungry apps like V

Re: frame-relay traffic shaping [7:55432]

2002-10-12 Thread Steven A. Ridder
In cisco terms, mincir is the cir, and cir is the port speed.

--

RFC 1149 Compliant.



neil K.  wrote in message
news:200210112334.XAA14349;groupstudy.com...
 Hi Group,

 Can someone please explain to me the difference between cir and mincir.Any
 help is highly appreciated.

 Regards,

 neil




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Re: Filtering NT domain listings at the router [7:54668]

2002-10-01 Thread Steven A. Ridder

I don't think its possible to filter NetBIOS domain lists specifically, but
you can filter UDP 135, 137-139, which should cover it.  Still not sure how
you are receiving rouge domains inside your network, though, as it could be
somthing else that is causing the domain lists to bleed in, such as rouge
PC's on your network or something.


Roberts, Larry  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hey all,

 Just curious if anyone has any links on filtering the domains on their
 network at the router.
 We are having a large amount of NT domains that are showing up internally,
 and I would like to start blocking these advertisements at the remote
 routers.
 Is this possible ? I can't figure out how, but I suspect that if it can be
 done, someone on this list has done it.



 Thanks

 Larry




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Re: CEF w/ NAT - Per packet or per destination? [7:54659]

2002-10-01 Thread Steven A. Ridder

per destination.  Especially if you are dual homed, and NAT two different IP
blocks, then the per packet load balancing would be a disaster.


Jarmoc, Jeff  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 When using CEF on a border internet router with a NAT device behind it, is
 it best to use per packet or per destination switching?

 Per Packet would obviously yield better balancing, but out or order packet
 delivery might be a problem.  Per-destination has the potential to flood
one
 circuit while the others sit idle, and also reduces maximum throughput to
 one circuit.  Does anyone have any other ideas or is BGP really the best
 solution at this point?

 Jeff Jarmoc - CCSA, CCNA, MCSE
 Network Analyst - Grubb  Ellis
 [EMAIL PROTECTED]




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Re: EM [7:54475]

2002-09-29 Thread Steven A. Ridder

If the analog flavor, 1 call per port.  But I think you can do emulated EM
on a PRI so would be 23 (one per channel).  EM is just a supervisory
signaling protocol, doesn't control the number of calls across the line.



Ismail M Saeed  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 All,
 Does anyone know how many voice channels the EM interface carry ?

 Thanks and best regards




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Re: Router Config [7:54479]

2002-09-29 Thread Steven A. Ridder

There is a command printer under global config that allows you to send
jobs to a printer.  I just looked it up and you need to understand the LPD
Unix command to understand printer on a router.


Robert Edmonds  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Copy it to a text file, then print it.  There is no way to print it
directly
 from the router.

 Hamed Sedighi  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Hi,
  How can I print my router configuration?
 
  Thanks,
  Hamed Sedighi




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Re: IDS [7:54493]

2002-09-29 Thread Steven A. Ridder

It can put a temp acl in place or do a TCP rst.  Be careful when deploying
either though.


Bruno Fernandes  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hi !!!

 In the case a sensor detects a match in one of its signatures is it
 possible for this sensor to shutdown a router interface ??

 Thank's in advance
 BF




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Re: OT: FXO FXS terminology - comments? [7:54331]

2002-09-28 Thread Steven A. Ridder

I should have specified.  I meant using an IP phone regularly, by pluging it
into a switch. But a crossover cable would work I guess.  FXO/FXS to IP
Phones = no.

--

RFC 1149 Compliant.



Chuck's Long Road  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 You can take a Cisco IP phone and connect it into a router?  Well OK, I
 didn't consider an ethernet crossover cable, which I suppose should work.
 You aren't saying you could plug an IP phone into an FXS or FXO port, are
 you?

 one other comment below:

 --

 www.chuckslongroad.info
 like my web site?
 take the survey!



 Steven A. Ridder  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Yes we have settled the question.  Most PBX's will probably use analog
EM
  if small, or Digital PRI/QSIG if larger.

 CL: or you plug a router FXO port into a PBX analogue port, correct? same
as
 you would plug a telco 1mb into a router FXO port?


 
  You can run an IP phone off of a router with ITS or SRST, but I probably
  shouldn't be telling you that without the caveat that you need a license
 for
  either service.  Contact your local cisco account rep, blah, blah,
blah...
 
 
  Chuck's Long Road  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   so far as I know, this will not work. Cisco's IP phones are ethernet
   devices, and must connect to a switch port. Well, you could use a hub
if
   you're looking for trouble. ;-
  
   IP phones are more akin to PC's, servers, etc, and you can't plug a PC
  into
   either an FXO or FXS port either. at least not and get it to do
anything
   useful.
  
   FXS and FXO are for telco connections only. FXS for analogue phone or
 fax.
   FXO for connection to PBX or telco CO.
  
   Have we settled this question - that an FXS port provides telco
 signaling
  to
   an FXO device?
  
   Chuck
  
   --
  
   www.chuckslongroad.info
   like my web site?
   take the survey!
  
  
  
   Daniel Lafraia  wrote in message
   [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
I'm wondering about buying a couple of Cisco Phones 7960 and a FXS
 card
   for
2600 and play with it. Will I be able to have a good voice lab only
 with
that? Maybe a FXO card and connect it in a regular phone line, is it
possible?
   
- Original Message -
From: Priscilla Oppenheimer
To:
Sent: Friday, September 27, 2002 3:25 PM
Subject: Re: OT: FXO FXS terminology - comments? [7:54331]
   
   
[...]
   
 Yes, you connect a phone to a router's FXS port. That's not
because
  the
 phone is a station, however. (That's what the NO referred to.)
It's
   becaue
 the phone is an FXO device.

 FXS goes to FXO and vice versa.

 Yes a PBX connects to a router's FXO port. The PBX uses an FXS
port
 in
this
 case. From the PBX point of view, it's connecting a phone. Makes
 sense
 right? What do PBXes connect? Phones. From the router's point of
 view,
   the
 router is getting dial tone, etc. from the PBX. The router is an
FXO
  in
this
 case. The router interface is labeled with what it is, as
mentioned.

 OK, I will stop writing messages on this topic. I should just turn
 my
 computer off. ;-)
   
[...]




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Re: Lookee Lookie - new certifications!!!! [7:54435]

2002-09-28 Thread Steven A. Ridder

Not sure I saw any new certifications, just a new org. to certify
Chuck's Long Road  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Check this out


 http://www.cisco.com/warp/public/625/ccie/ccie_program/fcpa.html

 AND

 http://www.fieldcertification.org

 AND


 http://www.fieldcertification.org/Membership/Pre_Qualification.htm

 Now you can get a certification that certifies that you are certified!!!

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Re: Lookee Lookie - new certifications!!!! [7:54435]

2002-09-28 Thread Steven A. Ridder

I guess you are correct.  I wonder if it will ever take off.


Chuck's Long Road  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 check further into the site:

 http://www.fieldcertification.org/Field_Certification.htm

 read all about field certification

 also

 http://www.fieldcertification.org/How_It_Works.htm

 sure looks like a whole new level of certification to me.

 not that I disagree with the principal here. But the home page ( and
Cisco's
 site ) does talk about this

 Get the Field Certified Professional (FCPT) credential to assert yourself
 as the real IT professional with actual skills and set your credential
apart
 from the paper ones! 

 Like I said - a whole new certification to certify that your certification
 is better than some paper certification.

 I can hardly wait.

 Chuck

 --

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 Steven A. Ridder  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Not sure I saw any new certifications, just a new org. to certify
  Chuck's Long Road  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   Check this out
  
  
   http://www.cisco.com/warp/public/625/ccie/ccie_program/fcpa.html
  
   AND
  
   http://www.fieldcertification.org
  
   AND
  
  
   http://www.fieldcertification.org/Membership/Pre_Qualification.htm
  
   Now you can get a certification that certifies that you are
certified!!!
  
   --
  
   www.chuckslongroad.info
   like my web site?
   take the survey!




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Re: Confused about Catalyst part numbers [7:54437]

2002-09-28 Thread Steven A. Ridder

Were all the servers on the same card and CEF on?  I had issues with that,
so we re-engineered the traffic to keep as much as possible on individual
cards, as the bus on the 4006 is only 2GB, as opposed to the 64 the
marketing department claims.


Erick B.  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Comments inline...

 --- Chuck's Long Road  wrote:
  Lupi, Guy  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   I would think about going with a 6509, the 5500
  series has been eol'd, but
   the last support dates are a while away yet.
  
 
  CL: Lorda mercy!!! you sound like almost all of the
  Cisco sales guys I
  know ;- Mention the word core and the only
  thing they can say is
  6509. Let's see -  one slot for the sup, one for
  the 16 port gig blade,
  one for the 48 port ethernet blade - the rest of the
  slots for baking pizzas
  :-
 
  CL: 12 copper gig ports and 48x10/100 ports fits
  nicely into a 4006, which
  conveniently now sells with an L3 blade.. Use the
  10/100/1000 blade, or use
  the copper gig GBICs, depending on other
  consideration.

 But the 4006 is a wiring closet switch. I recently ran
 into a company trying to use a 4006 w/sup2 with 12
 GBIC ports attached to servers w/gigabit NICs and
 their performance and throughput suffered. (Ie:
 In-lost errors, rx-errors, and txmt-errors which all
 point to excessive traffic and full buffers). I've
 only seen this w/sup2s however so maybe sup3 or sup4
 would help. I've seen other companys also have
 problems when using 4006 as a core/data-center device
 with a good amount of servers attached.

  CL: OR... I gotta keep brining this up - depending
  on the applications and
  traffic flows, a 3550-12G and a cou-ple of 3550-48's
  might just do the
  trick. The 12G is L3 out of the box.

 Agreed, or some other vendors box that isn't as pricy
 as the 6500 series (Extreme, Foundry).

   -Original Message-
   From: Stuart Pittwood
  [mailto:[EMAIL PROTECTED]]
   Sent: Saturday, September 28, 2002 2:12 PM
   To: [EMAIL PROTECTED]
   Subject: Confused about Catalyst part numbers
  [7:54437]
  
  
   I am looking into buying a Catalyst 5509 for the
  core of our network, I
   am however confused by the part numbers I will
  need.
  
   I need about 12 + Gigabit Ethernet (Copper) ports,
  48 10/100BaseT ports,
   a GBIC uplink to some 2950G-EIs we have, and an
  RSM to provide intervlan
   routing.
  
   Can anyone advise of of the part numbers I would
  need to get the
   required ports?
  
   Am I correct in thinking the the Supervisor Engine
  III would provide the
   layer 3 functions?
  
   Thanks in advance
  
   Stu


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Re: WAN Monthy Report [7:54362]

2002-09-27 Thread Steven A. Ridder

I'd just use MRTG, and cut and paste the files into a Power Point slide or
2.  Plus some numbers with the pictures.  That should suffice for both of
you.


Moises Luy  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Sorry, I don't have a report. Just in case, you might want to include SNMP
 information like traps and MIB poll stats.

 Can you send out a template of the final WAN report?

 -Moises

 -Original Message-
 From: Azhar Teza [mailto:[EMAIL PROTECTED]]
 Sent: Fri 9/27/2002 12:22 PM
 To: [EMAIL PROTECTED]
 Cc:
 Subject: WAN Monthy Report [7:54362]



 My boss has asked me to provide a monthly WAN reports regarding the
 UpTime/Downtime, Data Throughput etc.  Does someone has a template in
 regards of what other fields can be included in the report?  He would like
 to have a professional report. Thanks, Teza

 
 Changed your e-mail?  Keep your contacts!  Use this free e-mail change of
 address service from Return Path.  Register now!




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Re: ISDN Question [7:54356]

2002-09-27 Thread Steven A. Ridder

You can use dialer-load threshold for incoming traffic as well.


Christopher Dumais  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hello all,
 I have an ISDN line set up for vendor support. It is dedicated to one
 vendor. They are always the one to initiate the call, their system does
not
 allow incoming calls. They are telling me that I have to set the
thresholds
 on my router(Cisco 2620). I have never heard of the receiving side asking
 for the second connection and can't find any commands either. The only
 command I see is the dialer load-threshold command and that's for dial
out.
 Am I missing something? Any thought? Thanks in advance!

 Chris Dumais, CCNP, CNA
 Sr. Network Administrator
 NSS Customer and Desktop Services Team
 Maine Medical Center
 (207)871-6940
 [EMAIL PROTECTED]




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Re: 3500 GigatStack Module [7:54360]

2002-09-27 Thread Steven A. Ridder

When you combine the switches via gigastack, you are usually doing it
because you can't afford a chassis based switch in the closet (or don't need
one).  If you do gigastack the switches, they are treated as separate hops
in spanning tree.  I guess it's cheaper to run small gigastack cables then
it is to run the switches via fiber back to a core/disro 6500 both on the
wiring and the ports on the core/distro switch.  Those gbic blades cost more
than the chassis itself.




Azhar Teza  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 If I take 4 Catalyst 3500 Series Switches and configure in a GigabitStack
 Module then I would assume that I am creating a one virtual switch and all
 the backplanes of the switches should combine the total speed of switching
 backplane.  Am I correct or it is a samething you are connecting two
swiches
 through crossover and dividing the bandwidth.   If my assumptions are
 correct then the STP run only on those ports which will be uplink to (2)
 6509 layer 3 switches.  One in forwarding mode and the other one in
blocking
 mode.  The GigabitStack ports between the four switches should not be in
 either a forwarding or blocking port since they are just being used creat
a
 big one virtual switch from the 4 seperate physical switches. If my
 assumptions are incorrect then what is the benefit of using stacking
modules
 and diving the bandwidth instead of  combining them.  I would then rather
 connect each 3500 directly to 6509 switch.   Thanks

 
 Changed your e-mail?  Keep your contacts!  Use this free e-mail change of
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Re: OT: FXO FXS terminology - comments? [7:54331]

2002-09-27 Thread Steven A. Ridder

Yes we have settled the question.  Most PBX's will probably use analog EM
if small, or Digital PRI/QSIG if larger.

You can run an IP phone off of a router with ITS or SRST, but I probably
shouldn't be telling you that without the caveat that you need a license for
either service.  Contact your local cisco account rep, blah, blah, blah...


Chuck's Long Road  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 so far as I know, this will not work. Cisco's IP phones are ethernet
 devices, and must connect to a switch port. Well, you could use a hub if
 you're looking for trouble. ;-

 IP phones are more akin to PC's, servers, etc, and you can't plug a PC
into
 either an FXO or FXS port either. at least not and get it to do anything
 useful.

 FXS and FXO are for telco connections only. FXS for analogue phone or fax.
 FXO for connection to PBX or telco CO.

 Have we settled this question - that an FXS port provides telco signaling
to
 an FXO device?

 Chuck

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 Daniel Lafraia  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  I'm wondering about buying a couple of Cisco Phones 7960 and a FXS card
 for
  2600 and play with it. Will I be able to have a good voice lab only with
  that? Maybe a FXO card and connect it in a regular phone line, is it
  possible?
 
  - Original Message -
  From: Priscilla Oppenheimer
  To:
  Sent: Friday, September 27, 2002 3:25 PM
  Subject: Re: OT: FXO FXS terminology - comments? [7:54331]
 
 
  [...]
 
   Yes, you connect a phone to a router's FXS port. That's not because
the
   phone is a station, however. (That's what the NO referred to.) It's
 becaue
   the phone is an FXO device.
  
   FXS goes to FXO and vice versa.
  
   Yes a PBX connects to a router's FXO port. The PBX uses an FXS port in
  this
   case. From the PBX point of view, it's connecting a phone. Makes sense
   right? What do PBXes connect? Phones. From the router's point of view,
 the
   router is getting dial tone, etc. from the PBX. The router is an FXO
in
  this
   case. The router interface is labeled with what it is, as mentioned.
  
   OK, I will stop writing messages on this topic. I should just turn my
   computer off. ;-)
 
  [...]




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Re: 802.1Q support [7:54323]

2002-09-27 Thread Steven A. Ridder

I'd preface it with it depends on the traffic going across the VLAN's.  I've
seen 3600's get killed (actually die) with NFS traffic crossing an ethernet
port via inter-vlan traffic.  A 1750 can only handle so much pps.



Chris Headings  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 We use 1751's in a couple of environments and are easily using around
25-30
 VLANS's...with no problems.  The router gets some decent usage and
 everything has worked fine for us...

 HTH

 Chris




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Re: OT: rate-limiting proofs [7:54134]

2002-09-25 Thread Steven A. Ridder

that's the best command to show the output

--

RFC 1149 Compliant.



Chuck's Long Road  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Priscilla Oppenheimer  wrote in message
 [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
  Chuck's Long Road wrote:
  
   I'm putting in some rack time to review certain QoS features.
   Configuration
   is not really a problem. MQC makes this really easy :-
  
   However, I am attempting to observe results, and I am finding
   that I am
   unable to make bad things happen, such as packet drops.
  
   I am pinging from three different routers on a token ring to 3
   other routers
   via a 64K frame relay. The router that bridges the token ring
   and frame
   networks has the policy configured.
 
  You would have to exceed 64 Kbps for drops to occur, wouldn't you? Do
you
  have any idea how much bandwidth you're using on the Token Ring side?
What
  does show int show for load?
 
  I'm thinking you'll need to do more than ping. The problem with Cisco's
 ping
  is that it doesn't let you specify how much time between pings,
sometimes
  called an interval. The timeout value is for unsucessful pings. But what
 you
  need is a configurable interval  between the sending of pings,
successful
 or
  not. A real operating system or real ping tool would let you do this.
;-)


 CL: I finally was able to get some bad things to happen.

 token ring domain border router - frame relay domain

 I just started pinging from both sides, over an extended period of time.
To
 judget from the result, given the rudimentary configurations, it takes a
 minute or two for the rate limits to apply. There is an average traffic
 rate.

 three routers from each domain pinging the other side, packet sizes 1500
 bytes,  and I lowered the timeout value to 1 second from the default two
 seconds. By the time I added the sixth router's traffic, everybody started
 timing out. It took a minute or two for traffic to start going through
again
 after I stopped traffic from a router or two. I'll have to look into the
 defaults more closely.

 There has got to be a better show command than the show policy-map
 interface etc for this.

 Back to the docs.




 
  Ping in the MS-DOS prompt on Windows doesn't have this either, at least
 not
  the version I'm using. But ping under UNIX does, although it may not let
 you
  set the interval low enough. Some UNIXes have a -f (flood) option that
 will
  let you really whip the pings out. And a ping utility would let you do
 that
  too. For example, I use iNetTools from WildPackets.
 
  Are you trying to consume bandwidth just by using router tools or could
 you
  use a host also? Then there are many more options, of course.
 
  Hmm, what are some other ways to consume bandwidth by just configuring
  router options. Gazillions of SAPs? G


Re: SIP vs H323 [7:53852]

2002-09-23 Thread Steven A. Ridder

The signalling protocol used to set-up/tear-down a converstation wouldn't
affect the quality of the voice.  Maybe the equipment you used was superior,
or was on a low latency network.


Gunjan Mathur  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I tested one SIP equipement of vonage, and that was
 far far better then any device using H323...that's the
 reason I want to know the diff in between these two.
 What I understand is SIP model works on www/internet
 and h323 model is telephony, I believe this is the
 main reason for the quality difference.

 What you suggest...
 TIA

 --- Steven A. Ridder  wrote:
  I agree that SIP is the future, it just isn't there
  yet.  There is some SIP
  being built into Unity and CM, but until everything
  is SIP (as opposed to
  MGCP/H.323 and Skinny), it just isn't useful yet.
 
   I know that SIP is being deployed in SP networks,
  and I have implemented it
  in a Telco, but for enterprise, it's useless.  I
  can't wait til it is
  developed and more mature.
 
 
  Jason Weden  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   Ok, so SIP is nowhere near useless.  It is being
  used all over the place
  and
   will eventually replace H.323.  Telcos like Vonage
  (which uses Cisco SIP
   equipment), deltathree, and Denwa are using it for
  last mile telephony
   connectivity for residences and enterprises, and
  WorldCom, after surfacing
   from its financial issues, will be using it on its
  global network as well.
   Microsoft has built a SIP client into Windows XP
  (Microsoft Messenger) and
   SIP is very flexible and extensible and the best
  place to start is
   http://www.sipcenter.com.  PBX manufacturers like
  Mitel and Siemens have
   developed their PBX completely around SIP.
  
   To get back to Cisco (as this is a Cisco
  newsgroup), Cisco has taken the
   time and $$ to start to develop SIP functionality
  in its products despite
   the fact that it isn't need for AVVID at all.
  Though their initial SIP
   focus is on carrier-class products (since that is
  the logical choice --
  see
   my list of companies above), my bet is that SIP
  will surface as a more
   central part of the AVVID architecture for the
  enterprise.  A good Cisco
   link is here:
  
 
 http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/prodlit/index.shtml
  
or here (which displays more enterprise
  scenarios):
  
  
 

http://cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/index.htm
   Regards,
  
   Jason
 [EMAIL PROTECTED]


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bgp no-sync [7:53920]

2002-09-23 Thread Steven A. Ridder

Is it me, or is no-sync the default in BGP in 12.2.11T?

--
RFC 1149 Compliant




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Re: SIP vs H323 [7:53852]

2002-09-22 Thread Steven A. Ridder

h.323 is more robust, but more complicated (both to develop for and learn).
SIP is new and easy, but useless right now.  There's more security in h,323
right now.

--

RFC 1149 Compliant.



Gunjan Mathur  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hi,

 Is Cisco 2600 series support SIP? if yes, Please
 forward the web link if any...

 What is the main differance in between SIP  H323,
 which gives better quality of output?

 TIA



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Re: SIP vs H323 [7:53852]

2002-09-22 Thread Steven A. Ridder

I agree that SIP is the future, it just isn't there yet.  There is some SIP
being built into Unity and CM, but until everything is SIP (as opposed to
MGCP/H.323 and Skinny), it just isn't useful yet.

 I know that SIP is being deployed in SP networks, and I have implemented it
in a Telco, but for enterprise, it's useless.  I can't wait til it is
developed and more mature.


Jason Weden  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Ok, so SIP is nowhere near useless.  It is being used all over the place
and
 will eventually replace H.323.  Telcos like Vonage (which uses Cisco SIP
 equipment), deltathree, and Denwa are using it for last mile telephony
 connectivity for residences and enterprises, and WorldCom, after surfacing
 from its financial issues, will be using it on its global network as well.
 Microsoft has built a SIP client into Windows XP (Microsoft Messenger) and
 SIP is very flexible and extensible and the best place to start is
 http://www.sipcenter.com.  PBX manufacturers like Mitel and Siemens have
 developed their PBX completely around SIP.

 To get back to Cisco (as this is a Cisco newsgroup), Cisco has taken the
 time and $$ to start to develop SIP functionality in its products despite
 the fact that it isn't need for AVVID at all.  Though their initial SIP
 focus is on carrier-class products (since that is the logical choice --
see
 my list of companies above), my bet is that SIP will surface as a more
 central part of the AVVID architecture for the enterprise.  A good Cisco
 link is here:
 http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/prodlit/index.shtml

  or here (which displays more enterprise scenarios):


http://cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/index.htm
 Regards,

 Jason




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Re: Off Topic - Quietest Cisco Switch [7:53800]

2002-09-21 Thread Steven A. Ridder

I have to imagine a 1548 is quiet.


Charlie Wehner  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I'm looking to buy a switch for my apartment.  (Right now, the 2950T
24port
 10/100/1000Base-T looks promising.)

 However, the amount of noise this thing produces is a concern.  I want to
 put it in my living room (Actually, it's the only room... I live in a
 studio.) so I can't have this thing cranking away while I'm trying to
watch
 a movie, have a date over (Ya, it does happen sometimes... it's a
miracle.)
 or when I'm trying to go to sleep.  Does anyone know which switches are
the
 quietest?  I would like it to support the enhanced image.  Anyone else run
 into this problem?

 Thanks,
 Charlie




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Re: QOS on Sub-interfaces [7:53706]

2002-09-20 Thread Steven A. Ridder

you can do priority dlci's and you can apply class maps to map classes I
think.


dayo olabisi  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hi listers,

 I want to do some form of queueing on sub-interfaces
 on some of my routers. Each subif maps to a frame
 dlci.

 does any one know of a feature in IOS that can help?
 most queueing mechanisms I've come across work only on
 interfaces (not subifs.)

 thx,
 dayo

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Re: Cisco Serials and Theft [7:53574]

2002-09-18 Thread Steven A. Ridder

no, not that I have ever heard of.

--

RFC 1149 Compliant.



John Wright  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hi guys,

 Just found this group and it looks like a great resource for Cisco
 certification misc. questions. I'm CCNP and have passed CCIE written.

 My question is this: an aquaintance has offered me some really good
 equipment that I could really use to study for the CCIE, at really low
prices.

 I don't know the guy very well, he seems legit, but is there any way to
 check the serials on the equipment to see if its stolen? I don't want to
 possess stolen equipment, and I definitely don't want to find that out
 when/if I sell it after I pass the Lab. Thanks for any and all advice--

 John




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Re: CVOICE book: VoATM and VoFR [7:53567]

2002-09-18 Thread Steven A. Ridder

the router handles signaling.

--

RFC 1149 Compliant.



Tom Scott  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Having asked about VoMPLS transcoding from analog voice to MPLS
 frames without intermediate IP packets, my lab partner noticed
 that the CVOICE book (edited by Steve McQuerry etal) discusses
 VoFR and VoATM (chapters 8 and 9):


 analog+---+ +---+  analog
 phone A1  |   |   ATM   |   |  phone B1
... | rtr A |  or FR  | rtr B |...
 analog    |   |  cloud  |   |  analog
 phone Ai  +---+ +---+  phone Bj


 Are we reading this correctly, that the analog phones plug into
 the cisco routers and the analog voice traffic is transformed
 into FR frames or ATM cells, with no IP packets in between?
 It makes sense to do it that way in some applications. For
 example, if you have a call center in a distant suburb across
 a LATA line or two, that services a metropolitan area, then
 you'd want to bypass long-distance charges if at all possible.

 This seems like an easy way to do it. But what handles the
 call control? Does the router do that? Some of the diagrams
 in the CVOICE book have no PBX (or CCM) in them. Does the
 router translate the call-control signaling from the analog
 phone into corresponding pass-through signaling in the ATM/FR
 packets (sort of like user-to-user signaling that could be
 passed through SS7, in this case the users are the routers
 and the network is the ATM/FR switches)?

 -- TT




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Re: Cisco Proprietary? [7:53556]

2002-09-18 Thread Steven A. Ridder

I believe IGRP is still proprietary.

From the IETF page:

http://www.ietf.org/ietf/IPR/igrp



--

RFC 1149 Compliant.



hktco  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 When I learned my CCNA and CCNP, I read that IGRP is Cisco
 proprietary.  Recently I was told that IGRP is no longer proprietary
 and became an open standard.

 I would like to verify on this.  Any URL would be nice. Thanks.

 hktco




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Re: Prefix-list VS Access-list [7:53582]

2002-09-18 Thread Steven A. Ridder

I believe that it's the same.

--

RFC 1149 Compliant.



JohnZ  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Can I use access-list to produce the same effect as prefix-list ? Any
 thoughts on which is a better way to use in redistribution over other. I
am
 just trying to find which one I should stick with.
 Thanks

 router rip
  redistribute ospf 1
   network 135.11.0.0
   default-metric 5
  distribute-list prefix test out ospf 1

 ip prefix-list test seq 5 deny 199.172.4.0/24
 ip prefix-list test seq 10 deny 199.172.6.0/24
 ip prefix-list test seq 15 deny 199.172.8.0/24
 ip prefix-list test 20 permit 0.0.0.0/0 le 32

 or

 router rip
  redistribute ospf 1
   network 135.11.0.0
   default-metric 5
  distribute-list 10 out ospf 1

 access-list  10  deny 199.172.4.0/24
 access-list 10 deny 199.172.6.0/24
 access-list 10 deny 199.172.8.0/24
 access-list 10 permit any




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Re: CVOICE book: VoATM and VoFR [7:53567]

2002-09-18 Thread Steven A. Ridder

I don't think much has changed from the old days.  I know there's a ton of
new h.323 features, but those aren't in CVOICE.  And most aren't used in
simple networks.  The VoFR stuff will probably go away.

SIP's the new thing, but not there yet

--

RFC 1149 Compliant.



Priscilla Oppenheimer  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Tom Scott wrote:
 
  Having asked about VoMPLS transcoding from analog voice to MPLS
  frames without intermediate IP packets, my lab partner noticed
  that the CVOICE book (edited by Steve McQuerry etal) discusses
  VoFR and VoATM (chapters 8 and 9):
 
 
  analog+---+ +---+  analog
  phone A1  |   |   ATM   |   |  phone B1
 ... | rtr A |  or FR  | rtr B |...
  analog    |   |  cloud  |   |  analog
  phone Ai  +---+ +---+  phone Bj
 
 
  Are we reading this correctly, that the analog phones plug into
  the cisco routers and the analog voice traffic is transformed
  into FR frames or ATM cells, with no IP packets in between?
  It makes sense to do it that way in some applications. For
  example, if you have a call center in a distant suburb across
  a LATA line or two, that services a metropolitan area, then
  you'd want to bypass long-distance charges if at all possible.
 
  This seems like an easy way to do it. But what handles the
  call control? Does the router do that? Some of the diagrams
  in the CVOICE book have no PBX (or CCM) in them. Does the
  router translate the call-control signaling from the analog
  phone into corresponding pass-through signaling in the ATM/FR
  packets (sort of like user-to-user signaling that could be
  passed through SS7, in this case the users are the routers
  and the network is the ATM/FR switches)?

 Yup, you got it, although it may be even simpler than you imagine.

 Before AVVID, Cisco did VoIP, VoFR, and VoATM, as discussed in the CVOICE
 class. With these solutions, you simply connected analog phones to FXS
ports
 on routers. The routers digitized and compressed the dialed digits and the
 voice itself and packetized it. If it was VoATM or VoFR, there was no IP.
 The data was simply put into data-link-layer frames (or cells with ATM).

 You asked about the call-control signaling from the analog phone, but how
 much would there be? These phones would be your basic $5.99 KMart special
 with no bells and whistles, so to speak. The router provides dial tone and
 picks up the dialed digits and forwards them to the other router.

 As you can probably tell, I'm not a telepony expert, but I have gotten
quite
 a few of these simple voice/data networks up and running. It's very easy.
 There is no Call Manager! And, as you mentioned, the major benefit is that
 you bypass long-distance charges because you simply use the existing data
 network. You may need to prioritize voice, and break up big data packets
to
 get the low level of delay required for voice, but other than that,
there's
 not much to it. The original CVOICE class covered only these types of
 solutions and I'm sure the book still has a lot of this flavor, although
 both the book and the newer version of CVOICE also cover newer solutions
too
 these days probably.

 ___

 Priscilla Oppenheimer
 www.troubleshootingnetworks.com
 www.priscilla.com

 
  -- TT




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Re: ISDN load-interval [7:52851]

2002-09-07 Thread Steven A. Ridder

tells isdn card at what point to bring up second channel.

--

RFC 1149 Compliant.



hagedorn  wrote in message
news:[EMAIL PROTECTED]...
 Hy can someone exlpain me for what the load-interval in isdn is.

 any comments are welcome.

 Regards Philipp




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Re: Access Server Problem !!! Help!!! [7:52849]

2002-09-07 Thread Steven A. Ridder


That line is the reason, but it got cut off.  Resend the entire line.

--

RFC 1149 Compliant.



FAhmed  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hi all
 I ve got a problem

 The user is getting disconnect after 2-3 minutes, I ve changed the
 line at both side,
 Doesn't help, it was working b4 a month no configuration has been
 changed
 Anyone knows about this error?
 8589934592d8589934592h: Call Handle failed for Modem 2/1
 Also this one
 2002-08-31 14:20:45 Local7.Debug 192.168.10.13 1924: 2d06h: TTY66:
 Async Int reset: Dropping DTR 2002-08-31 14:20:45 Local7.Debug
 192.168.10.13 1925: 2d06h: Modem 2/1 Mcom: in modem state
 'Disconnecting' 2002-08-31 14:20:45 Local7.Debug 192.168.10.13 1926:
 2d06h: Modem 2/1 Mcom: DISCONNECT, duration = 00:02:01, reason (0x9)
 DTR Drop



  2d07h: Modem 2/1 Mcom: in modem state 'Dialing/Answering'
  2d07h: Modem 2/1 Mcom: in modem state 'Incoming ring'
  2d07h: %LINK-3-UPDOWN: Interface BRI1/2:1, changed state to up
  2d07h: Modem 2/1 Mcom: in modem state 'Waiting for Carrier'
  2d07h: %LINEPROTO-5-UPDOWN: Line protocol on Interface BRI1/2:1, o up
  2d07h: %ISDN-6-CONNECT: Interface BRI1/2:1 is now connected to 0
  2d07h: Modem 2/1 Mcom: in modem state 'Connected'
  8589934592d8589934592h: Call Handle failed for Modem 2/1
  2d07h: Modem 2/1 Mcom: CONNECT at 31200/31200(Tx/Rx), V34, LAPM,
  2d07h: TTY66: DSR came up
  2d07h: Modem 2/1 Mcom: switching to PPP mode
  2d07h: TTY66: no timer type 1 to destroy
  2d07h: TTY66: no timer type 0 to destroy
  2d07h: tty66: Modem: IDLE-(unknown)
  2d07h: %LINK-3-UPDOWN: Interface Async66, changed state to up
  2d07h: Modem 2/1 Mcom: PPP escape map: Tx map = , Rx map
  2d07h: Modem 2/1 Mcom: PPP escape map: Tx map = , Rx map
  2d07h: TTY66: Async Int reset: Dropping DTR
  2d07h: Modem 2/1 Mcom: in modem state 'Disconnecting'
  2d07h: Modem 2/1 Mcom: DISCONNECT, duration = 00:00:26, reason (
  2d07h: TTY66: DSR was dropped
  2d07h: tty66: Modem: READY-(unknown)
  2d07h: TTY66: dropping DTR, hanging up
  2d07h: tty66: Modem: HANGUP-(unknown)
  2d07h: Modem 2/1 Mcom: in modem state 'Idle'
  2d07h: %LINK-5-CHANGED: Interface Async66, changed state to rese
  2d07h: TTY66: cleanup pending. Delaying DTR
  2d07h: TTY66: cleanup pending. Delaying DTR
  2d07h: TTY66: cleanup pending. Delaying DTR
  2d07h: Modem 2/1 Mcom: switching to character mode
  2d07h: TTY66: no timer type 0 to destroy
  2d07h: TTY66: no timer type 1 to destroy
  2d07h: TTY66: no timer type 3 to destroy
  2d07h: TTY66: no timer type 4 to destroy
  2d07h: TTY66: no timer type 2 to destroy
  2d07h: Async66: allowing modem_process to continue hangup
  2d07h: TTY66: restoring DTR
  2d07h: TTY66: autoconfigure probe started
  2d07h: %LINK-3-UPDOWN: Interface Async66, changed state to down

 Best Regards
 Have A Good Day!!
 ++
 Farhan Ahmed
 MCSE+I, MCP Win2k, CCA, CCDA, CCNA, CSE , CCNP
 Network Engineer
 Mideast Data Systems Abu Dhabi Uae. www.mdsemirates.com
 Tel: 97126274000Cellular: 971507903578
 ++

 Be a builder, not a destroyer!!!

 Disclaimer:
 Privileged/Confidential Information may be contained in this message or
 Attachments hereto. Please advise immediately if you or your employer do
 not consent to Internet email for messages of this kind. Errors and
 Omissions may occur in the contents of this e-mail arising out of or in
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 or software error, malfunction, or  by the person who is sending the
email.
 Mideast Data Systems accepts no responsibility for any such errors or
 omissions  Opinions, Conclusions and other information in this message
that
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Re: IDS Appliance [7:52308]

2002-08-29 Thread Steven A. Ridder

so far so good.  I installed one for a client, and it worked awesome.  I
even dropped it 10 feet, and it still worked!



Brian Wilkins  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I was wondering if anyone else has been experiencing problems with Cisco's
 IDS sensor appliance (formerly Netranger).  Almost every time I load a
 service pack or new signature file I end up rebuilding the device from
 scratch using the install CD's.  I've filed multiple cases with TAC, with
 little help recieved.  I've even spoken to the product manager for the
 devices and still can't seem to stabilize these things.

 Anyone else using Cisco IDS appliances?  If so, how's your luck with them?

 Thanks,

 Brian




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Re: Blocking directed broadcast [7:52205]

2002-08-28 Thread Steven A. Ridder

sh ip int

--

RFC 1149 Compliant.



YASSER ALY  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hi All,  Is there a way to verify that  no ip directed-broadcast  is
 enabled on an interface.I was trying to enable it on a router but typing
 the command and then doing sh run nothing appears, trying to type the
 command without no at the begining it shows in the sh run which gives the
 impression that  no ip directed-broadcast  is the default for this
 version.  Any ideas to verify that it is really working and blocking the
 directed-broadcasts ? Regards,Yasser

 

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Re: 128Kbps instead 64kbpes [7:52190]

2002-08-28 Thread Steven A. Ridder

dialer-load threshold 1either

--

RFC 1149 Compliant.



Eng. ABDALLAH QUQAS  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Dear ALL,

 how i can make ISDN  BRI to connect at 128kbps instead of 64kbps as a
bundle
 channel of Cisco router 3600 using ppp encapsulation.

 Kind Regards
 abd quqas




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Re: VoIP over Qwest VPN WAN backbone [7:51837]

2002-08-21 Thread Steven A. Ridder

Larry hit it right on the head.  WFQ is a poor choice for voice, and you'll
probably notice a huge difference when moving to LLQ.

--

RFC 1149 Compliant.



Roberts, Larry  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I have 6 sites that have active queuing for VoIP traffic on the Qwest VPN
 network. ( and 100 or so others that don't )

 I have heard good things about the quality , to the point that they are
 starting to request other sites have this as well.

 I configured it for LLQ on the interface, and required that they send
their
 voice traffic with the DSCP already set.
 I guess I could have done it for them, but if they have the ability to
mark
 the traffic, I let the Voice Switches do it.

 I believe that I used a priority of 128k for the Voice Queue, and matched
 that via a bandwidth statement on the class default.
 It depends on the amount of data that you are going to have going between
 sites, and the compression that you use.
 I'm being told that they are using 729a, but that's just what I'm told.

 If jitter is the problem , LLQ should fix it as the variance is most
likely
 caused by your routers. I say this knowing that with the loss of
customers,
 the VPN has plenty of spare bandwidth to use :)

 Thanks

 Larry


 -Original Message-
 From: dj [mailto:[EMAIL PROTECTED]]
 Sent: Wednesday, August 21, 2002 10:18 AM
 To: [EMAIL PROTECTED]
 Subject: VoIP over Qwest VPN WAN backbone [7:51837]


 Has anyone deployed VoIP successfully across Qwest's WAN backbone?  I have
2
 sites connected over Qwest's VPN service (consists of running
 IPSec over Qwest's MPLS backbone).   Qwest says they currently have no
 QOS support in backbone, but the backbone has huge capacity (OC-192
pipes),
 which Qwest asserts should support VoIP apps.  Round trip ping times
across
 the WAN are typically 50 msec.

 I have the traditional VoIP problem of excessive jitter causing poor voice
 quality.  Currently have 2621 routers at each site running WFQ wth Voice
 packets marked with IP precedence 5 for voice data, and IP precedence 3
for
 voice control.  I am considering re-configuring the routers to run
 PQ-CBWFQ.

 If anyone has successfully got VoIP running successfully across this
Qwest's
 backbone, I would be interested in hearing your approach and which config
 worked.  Even though the Qwest WAN has huge capacity, but no QOS support,
 can it really support real-time apps like VoIP???

 thanks,
 dj

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Re: voip [7:51729]

2002-08-20 Thread Steven A. Ridder

h.323 can do it with RAI, or you could use SA Agents. THose are your two
best options.

--

RFC 1149 Compliant.



Jake  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Is there a way to tell a router (3810) , which is running voip, to reroute
a
 voip call if the destination router is down.  This is how I see it.  The
 call is made from a typical digital phone.  The pbx sends the digits to
the
 router. The router processes the digits and sends them to the destination
 router.  What happens if the destination router is down.  The PBX does not
 know if the destination router is down , so it will send the digits to the
 local router.  But,  how do I tell the local router to reroute the phone
 call?? If you need a more info please specify..

 Thanks
 Jake




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Re: PIX Port-Scanning Q [7:51503]

2002-08-16 Thread Steven A. Ridder

Is there an IDS signature that can help you recognize it on the PIX?  That
maybe the easiest way to do it.


Richard Tufaro  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hey all, maybe silly question, but how do I log who is port scanning me?
Iv
 got the logging on my PIX to informational, and still don't see anything
 like port scanning logs...




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Re: PIX Failover [7:51491]

2002-08-16 Thread Steven A. Ridder

Speaking of stateful PIX's, if I make a change on 1 PIX, and it has failover
on, will it automatically make a change on the other PIX?


Gaz  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 In article , [EMAIL PROTECTED]
 says...
  Hi,
 
  In a Stataful configuration, and two PIX are interconnected via a
  dedicated Failover Fastethernet, in case of the Active unit's Internal
  interface fails, is there any method to shift traffic to the Standby
  unit's Internal interface to maintain connectivity, thanks.
 
  Leo
  Best Regards.
 Not sure what you mean there. That's what failover does unless I'm
 misunderstanding your question.

 You configure the main IP address for the interface and you configure a
 failover address. If the Pix's decide that the active one has a problem
 (power,interface down etc) the secondary pix takes over the main IP
 address.
 If the primary is still contactable it will have the failover IP address
 on its inside interface.

 That's why it's safe to telnet to the main IP address and you know that
 you're on the active Pix, but by console you need to do a show fail to
 make sure the device you're on is primary active or secondary active
 before you make changes.

 Regards,

 Gaz




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Re: Local Cisco office and CCIE [7:51282]

2002-08-13 Thread Steven A. Ridder

I never had a cisco engineer ask if we were silver or up.  Just call your SE
and schedule some lab time, that's how it works in MA.  The only requirement
is you have to have passed the CCIE written.


cebuano  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I believe this is only true for Silver and up if the local Cisco CAM
 will sponsor you to the ASET program, which has been on, off, on, off...
 You can contact your local Cisco office to see if the in-house lab
 engineer will let you practice on their equipment. Last time I checked,
 the SE was real friendly, as long as he thinks you won't damage anything
 on the racks.

 Elmer
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of
 NetEng
 Sent: Monday, August 12, 2002 8:00 PM
 To: [EMAIL PROTECTED]
 Subject: Local Cisco office and CCIE [7:51282]

 I thought I read once on cisco.com (can not find now) that once you pass
 the
 CCIE written your local cisco office will help you prepare for the lab
 portion with local lab/resources. Was this wishful thinking or do they
 help?




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Re: callmanager dial plan question [7:48300]

2002-07-31 Thread Steven A. Ridder

H.323 gatekeepers are the grand dial-plan for CM.


[EMAIL PROTECTED] (John Nemeth)  wrote in
message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 On Nov 28,  1:50pm, Chris Charlebois wrote:
 }
 } And Chuck is right, dial plans are not shared between clusters within
the
 } software, so a Grand Dial Plan Scheme should be developed before
starting
 } and implemented within each cluster.

  At this point, the astute will note that dial plans are simply
 another form of addressing.  They will then run out and get Howard's
 book, Designing Addressing Architectures.  It is a very good book.  It
 doesn't cover dial plans specifically, since it is geared towards data
 networks.  However, the basic concepts still applies.  It does spend a
 fair bit of time on the basics which is good, since once you know the
 basics, you can work with any kind of addressing system.

 }-- End of excerpt from Chris Charlebois




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Re: Supervisor Engines [7:50279]

2002-07-31 Thread Steven A. Ridder

Sup 3 is IOS based, Sup 2 is CatOS based.  Sup 3 is new, and may not have
all the features you desire right away.

I think Sup 1 is L2 only.


Stuart Pittwood  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 We're looking into replacing some of our old hubs/switches with a single
 4000 series switch.

 My question is what is the difference between supervisor engines I, II,
 III?

 Any help appreciated
 Thanks

 Stu




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Re: IPSec or CBAC etc.?? [7:50177]

2002-07-30 Thread Steven A. Ridder

I don't think so.  I have never seen any of those requirements in the
blueprint or anywhere else.

--

RFC 1149 Compliant.



Cisco Nuts  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hello,

 Does anyone know if IPSec or CBAC or TCP intercept topics are REQUIRED for
 the RS Lab?

 Are candidates required to know how to configure IPSec etc. in the Lab?

 I see Reflexive, Time-based and Dynamic ACL which I think is OK but these
 other technologies?? :-(

 Please advise.

 Thank you.



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Re: Hardware requirement for Cisco CallManager [7:50142]

2002-07-30 Thread Steven A. Ridder

I'm not sure there are any certified HP servers for CM.  Last I checked,
there weren't.


Brian Zeitz  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Yea, I think there is 2 different versions, Compaq and HP. I think you
 need to pick the version up front. I have Compaq DL380s and DL360s,
 these are the fastest servers I have ever seen. Esp for the size, 1U,
 and its great that it comes with insight manager for free. I don't like
 IBM, I think there products are junk and they can never get a concept
 off the ground. You wont be sorry if you go with Compaq, most people I
 talk to who use CallManger say use Compaq hands down. Someone offered me
 this software, but having Compaq servers in my living room would be a
 bit too much ;) The DL360 must be certified, because that is what most
 people use. I couldn't find the info on the site.


 -Original Message-
 From: Chris Charlebois [mailto:[EMAIL PROTECTED]]
 Sent: Tuesday, July 30, 2002 2:57 PM
 To: [EMAIL PROTECTED]
 Subject: RE: Hardware requirement for Cisco CallManager [7:50142]

 CallManager 3.1 and higher is certified on Compaq DL320, DL380, and IBM
 series 340, for sure.  I assume that DL360, also, although I have no
 first-hand knowledge of that, and I beleive some HP server (I think even
 a
 Dell).  These are just the servers that are supported using the Sperion
 Installation Utility for the OS.  In actualality, you can run an OS
 patch on
 any server running W2K Server and then install CallManager itself on
 top.
 The manufacturer isn't nearly as important as the performance.  That
 being
 said, I wouldn't install even a lab CallManager on anything less than
 P3-700
 with 512 memory.  Production *should* be over a gigahertz with a GB of
 memory.  And I would recommend installing any other apps on the
 CallManger
 server, either.




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Re: Queuing.. [7:49992]

2002-07-29 Thread Steven A. Ridder

CBWFQ should work.  You can guarantee a minimum that each class of traffic
will receive during times of congestion.


DW  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Can anyone help,

 I am setting up a WAN link between two sites that will be running Citrix.
 There will be quite a lot of TCP/IP print traffic running across the link
 and I want to ensure that this does not interfere with the ICA traffic.
What
 would be the best queuing method to use (If this is the way I should
 proceed) ???

 Any help appreciated..




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Re: OSPF and auto-cost refernce-bandwidth question, value [7:49973]

2002-07-28 Thread Steven A. Ridder

oc192.

--

RFC 1149 Compliant.



bergenpeak  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Suppose you have a network with a mix of FE, GE, OC-3/12/48 POS
 links.

 With the standard OSPF link costing mechanism, all of these links
 turn out to have a link cost of 1.

 Are there reasons to not go ahead and change the link cost calculation
 via the auto-cost reference-bandwidth command to better reference the
 link capacities?  (I'm assuming that changing this value on all
 routers will cause a cascade of LSAs and SPF recalcs, but that one is
 willing to take that hit)

 Suppose the decision is to go ahead and make the auto-cost change. Is is
 possible that the calculated OSPF paths will be different if one uses
 OC-48
 or OC-192 as the reference bandwidth?

 Thanks




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Re: access-list for steaming audio [7:49817]

2002-07-27 Thread Steven A. Ridder

I haven't been keeping up with NBAR, but they may have some pdm's to block
the streaming audio apps.

NBAR was built for stuff like that, but I don't feel there's a need to block
this type of stuff.  Same with IM.  Let the users have some use of their PC
and increase productivity.


Spencer Plantier  wrote in message
news:[EMAIL PROTECTED]...
 Which ports need to be blocked for streaming video and
 audio.

 Thanks

 =
 Spencer Plantier
 Internet Solutions Engineer
 Cell 919-696-8848

 __
 Do You Yahoo!?
 Yahoo! Health - Feel better, live better
 http://health.yahoo.com




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Re: CCIE Study: CBWFQ / CQ [7:49816]

2002-07-27 Thread Steven A. Ridder

CBWFQ is easier to configure and the default q is a WFQ.


Jay Greenberg  wrote in message
news:[EMAIL PROTECTED]...
 Can anyone please explain the difference between CBWFQ and CQ?  It seems
 to me that they both allow you to class traffic in a custom manner, so
 whats the main difference?

 Jay Greenberg




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Re: CLI vs PDM [7:49774]

2002-07-27 Thread Steven A. Ridder

PDM.  It's just too easy.  Or use CiscoWorks VMS or CSPM to manage a bunch
of them.

I can't stand the cli and it's archaic language.  I'd rather make sure my FW
is easy to configure so I don't miss anything.  If something is too
complicated, you don't even know if you have problems because you are stuck
fighting with configs instead of hackers.

Although the a basic PIX config is easy to set up, once you start doing
complicated large-scale projects and need to have complex configurations,
the cli on the PIX gets ridiculous.


Just my opinion.


Georgescu, Aurelian  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 If you can live with the fact that PDM is very slow, then use PDM, it's
far
 more intuitive for access control (which is what PIX is made for...)

 Aurelian


 -Original Message-
 From: Juan Blanco [mailto:[EMAIL PROTECTED]]
 Sent: Friday, July 26, 2002 10:46 AM
 To: [EMAIL PROTECTED]
 Subject: CLI vs PDM [7:49774]

 Team,
 For those security people on the Cisco World, Normally which interface do
 you use the most, the CLI or the PDM. I am in the process of setting up
 standards and we would like to define one for having access to our Pix.

 Thanks,


 Juan Blanco
 
 The greatest glory in living lies not in never falling,
  but in rising every time we fall .
  -- Nelson Mandela
 




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Re: ISP QoS Architecture Question [7:49767]

2002-07-27 Thread Steven A. Ridder

There are a lot of Cisco Powered networks doing QoS, but more for
multi-service type stuff for voice and video.

If you look on Cisco's web-site, there's a whole program your ISP can join
to become Cisco-powered in multiservice, although I don't know the exact
search terms you'd need to find it.  If you want to join, Cisco will tell
you in technical detail what you need to do for QoS.  I imagine it's just
stuff under the MQC, but I never bothered to ask.

In any case, I don't think giving lowly http priority is the best business
decision you can make.  It's TCP based so it can handle delay, plus you have
no control of the http traffic once the user leaves your network onto
another, so all the time and effort you place into it, may be all for
naught.  Even if you give http A+ traffic rating, as soon as the user goes
to another network peer, it may be congested.

Just my .02 cents


--
RFC 1149 Compliant






Jay Greenberg  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I am considering deploying QoS features in our ISP.   The ISP has about
 60 thousand users in total, and I was thinking of setting a general
 traffic policy.E.g., I would like to set HTTP traffic down to a very
 low delay, to make the network seem faster to end users.   I suppose
 what I am asking is - has anyone done this for an ISP, and if so, how
 did it turn out?


 Jay Greenberg




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Re: access-list for steaming audio [7:49817]

2002-07-27 Thread Steven A. Ridder

As log as you can nat the IM (or tunes) behind the FW, it's pretty safe, but
I'll never say there aren't any vulnerabilities.  At leas tif it's antted,
there are no holes comming in.

--

RFC 1149 Compliant.



Maccubbin, Duncan  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Be careful with this kind of thinking. More and more holes in IM are
 showing up everyday. If you let IRC on your network then you are asking
for
 trouble. As for streaming audio, have you looked at the % of bandwidth
they
 use? If you have a fairly utilized pipe or (like most companies) are
paying
 for bandwidth then that is a consideration.

 Just my $0.02.

 Duncan

 -Original Message-
 From: Steven A. Ridder [mailto:[EMAIL PROTECTED]]
 Sent: Saturday, July 27, 2002 10:59 AM
 To: [EMAIL PROTECTED]
 Subject: Re: access-list for steaming audio [7:49817]

 I haven't been keeping up with NBAR, but they may have some pdm's to block
 the streaming audio apps.

 NBAR was built for stuff like that, but I don't feel there's a need to
block
 this type of stuff.  Same with IM.  Let the users have some use of their
PC
 and increase productivity.


 Spencer Plantier  wrote in message
 news:[EMAIL PROTECTED]...
  Which ports need to be blocked for streaming video and
  audio.
 
  Thanks
 
  =
  Spencer Plantier
  Internet Solutions Engineer
  Cell 919-696-8848
 
  __
  Do You Yahoo!?
  Yahoo! Health - Feel better, live better
  http://health.yahoo.com




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polycom Video Unit [7:49882]

2002-07-27 Thread Steven A. Ridder

Does anyone know what type of traffic a typical Polycom Video Con. unit
creates?  It it multicast?  What ports does it use?  Is it standard h.323?

I can sniff it, but if anyone has already done their homework on it, it will
save me some time.

--

RFC 1149 Compliant.




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Re: access-list for steaming audio [7:49817]

2002-07-27 Thread Steven A. Ridder

I talk to people outside the company all the time with IM.  I use it for
remote tech support.  It's great.




Dan Penn  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 What I really like is some corporations embracing instant messaging for
 internal use.  I think that having a private corporate IM server for the
 corporate users to connect to would be a great way to increase
 productivity.  However, on the same hand, I would fear the end-users
 being able to connect to AIM, ICQ, MSN, etc.  That would open up way to
 many holes.  I mean really...what good is going to come of user being
 able to connect to an instant messenger.  Who do they need to talk to
 outside of the corporation during working hours?

 Dan

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of
 Maccubbin, Duncan
 Sent: Saturday, July 27, 2002 9:34 AM
 To: [EMAIL PROTECTED]
 Subject: RE: access-list for steaming audio [7:49817]

 Be careful with this kind of thinking. More and more holes in IM are
 showing up everyday. If you let IRC on your network then you are asking
 for
 trouble. As for streaming audio, have you looked at the % of bandwidth
 they
 use? If you have a fairly utilized pipe or (like most companies) are
 paying
 for bandwidth then that is a consideration.

 Just my $0.02.

 Duncan

 -Original Message-
 From: Steven A. Ridder [mailto:[EMAIL PROTECTED]]
 Sent: Saturday, July 27, 2002 10:59 AM
 To: [EMAIL PROTECTED]
 Subject: Re: access-list for steaming audio [7:49817]

 I haven't been keeping up with NBAR, but they may have some pdm's to
 block
 the streaming audio apps.

 NBAR was built for stuff like that, but I don't feel there's a need to
 block
 this type of stuff.  Same with IM.  Let the users have some use of their
 PC
 and increase productivity.




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Re: CCIE Study: CBWFQ / CQ [7:49816]

2002-07-27 Thread Steven A. Ridder

I haven't taken the CCIE lab yet, but first I'd think they'd want you to use
CBWFQ or LLQ, as the rest are just ancient.   Cisco is emphasizing AVVID, so
you'd probably be tested on avvid technologies.  I'd also study NBAR and
multicast, among others.   Probably dial-peers as well.

But I guess if they did ask about old qing methods, it would be someting
like this:

Use a qing method that gives priority to one q with no regard to protocol
starvation.  Then you'd use PQ

Use a qing method with no PQ and has 5 classes for you to chose from.  Then
you'd use CQ

Use a qing method that is fair to all protocols and gives preference to
higer ToS.  Then you'd use WFQ which is the default for e1 and below.

Use a qing method that meets modern requirements for voice and AVVID.  Then
it's LLQ

Use a qing methid that gives classes to traffic and wfq's the rest.  Then
that's CBWFQ.

Jason Greenberg  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Ok fair enough, but from a CCIE lab exam question perspective, I'm
 trying to determine when to use which technique for what type of
 question.  Are there certain things that each can do that the other
 cannot?

 On Sat, 2002-07-27 at 10:59, Steven A. Ridder wrote:
  CBWFQ is easier to configure and the default q is a WFQ.
 
 
  Jay Greenberg  wrote in message
  news:[EMAIL PROTECTED]...
   Can anyone please explain the difference between CBWFQ and CQ?  It
seems
   to me that they both allow you to class traffic in a custom manner, so
   whats the main difference?
  
   Jay Greenberg
 --
 Jason Greenberg, CCNP
 Network Administrator
 Execulink, Inc.
 [EMAIL PROTECTED]




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Re: Eigrp Summarizing [7:49730]

2002-07-26 Thread Steven A. Ridder

Yes, I didn't read the requirements.  If this had been an actual CCIE test
objective, I would have received 0 points.

--

RFC 1149 Compliant.



David j  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I would added eigrp stub connected on spoke routers (I think you need ios
 12.0) if you don't do that, based in my experience, you're going to have
 SIAs on your backbone. See the following link:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120
limit/120s/120s15/eigrpstb.htm

 Steven, correct me if I'm wrong, but if you use ip default network, all
 other routes are being announced anyway?

 Steven A. Ridder wrote:
 
  on the interface to the rest of the routers, do a ip eigrp
  summary address
  00.0.00 0.0.0.0
 
  or
 
  IP default network xx.x.x.
 
 
  JohnZ  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   I have a 3640 as hub and 20 1604s as spokes. Eigrp is the
  routing protocol
   in use. Internet access is through the 3640. How can summrize
  in Eigrp so
   all the spokes have a single route to the Hub router.
   Thanks,




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Re: Eigrp Summarizing [7:49730]

2002-07-26 Thread Steven A. Ridder

I don't think stub generate summary routes like it does in OSPF.  You still
need to summarize.

--

RFC 1149 Compliant.



David j  wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 I would added eigrp stub connected on spoke routers (I think you need ios
 12.0) if you don't do that, based in my experience, you're going to have
 SIAs on your backbone. See the following link:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120
limit/120s/120s15/eigrpstb.htm

 Steven, correct me if I'm wrong, but if you use ip default network, all
 other routes are being announced anyway?

 Steven A. Ridder wrote:
 
  on the interface to the rest of the routers, do a ip eigrp
  summary address
  00.0.00 0.0.0.0
 
  or
 
  IP default network xx.x.x.
 
 
  JohnZ  wrote in message
  [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
   I have a 3640 as hub and 20 1604s as spokes. Eigrp is the
  routing protocol
   in use. Internet access is through the 3640. How can summrize
  in Eigrp so
   all the spokes have a single route to the Hub router.
   Thanks,




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Re: Which Voice Class [7:49564]

2002-07-24 Thread Steven A. Ridder

VoIP or Call Manager/IP Tel?

If I had to take a class, I guess it would be CVOICE, although I've never
taken it.  I've read the book, and I'd recommend that.

I would also recommend a QoS class.  That may be the most useful, and there
aren't a ton of good books to help understand that.

my $0.02


 wrote in message
[EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
 Hello Guys

 I've zero voice experience and I would like to take one Cisco's voice
 classes, can you guys recommend which of the classes I should take?


 Thanks...Nabil

 I have never let my schooling interfere with my education.




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