[OSL | CCIE_Voice] ATA

2009-03-03 Thread hasan khalife

ATA MUSET BE IN THE VOICE AND DATA VLAN ?  

 

OR JUST VOICE VLAN 

 

 

SET VLAN 140 (X/X) ATA PORT

 

 

SET PORT AUXILIARY VLAN (X/X) 240 

 

 

 

THANK YOU

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Re: [OSL | CCIE_Voice] ATA

2009-03-03 Thread basant yadav
Seems like your ATA is connect on CAT6.

If your Voice Vlan is 140 then command on CAT 6 would be set vlan 140 X/X

- Basant

On Tue, Mar 3, 2009 at 9:28 AM, hasan khalife hasan_khal...@hotmail.comwrote:

  ATA MUSET BE IN THE VOICE AND DATA VLAN ?

 OR JUST VOICE VLAN


 SET VLAN 140 (X/X) ATA PORT


 SET PORT AUXILIARY VLAN (X/X) 240



 THANK YOU

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[OSL | CCIE_Voice] call group of DN number to conference at a time

2009-03-03 Thread Balamurugan Singaram
Hi,
 
IF I want to call group of DN number to conference at a time, I can dial one by 
one and call each one.
There is any simple way like hunt group to call all of them at once for 
conference, please light on this.
 
For ex in a company they have sales team, here manager want to call the whole 
team just by dialing a single number, any workaround for this ?
 
 
Thanks,
Bala.


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Re: [OSL | CCIE_Voice] call group of DN number to conference at a time

2009-03-03 Thread Alex
In CME it is called paging but it's one-way communication from the person who 
initiated the paging to the group of others who are paged. I guess you can join 
the paging-DN into the conference but I never tried this.
I don't know a way to do it on CCM.
Rgds
Alex
  - Original Message - 
  From: Balamurugan Singaram 
  To: ccie_voice@onlinestudylist.com 
  Sent: Tuesday, March 03, 2009 9:41 AM
  Subject: [OSL | CCIE_Voice] call group of DN number to conference at a time


Hi,

IF I want to call group of DN number to conference at a time, I can 
dial one by one and call each one.
There is any simple way like hunt group to call all of them at once for 
conference, please light on this.
 
For ex in a company they have sales team, here manager want to call the 
whole team just by dialing a single number, any workaround for this ?
 
 
Thanks,
Bala. 


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[OSL | CCIE_Voice] Unity Express

2009-03-03 Thread Tanmay Devare
Hi,

Integrated CUE with CME.

Q : User can press 2 (where  = actual DN of IP phone which is
registered to local CME) to reach user's voicemail box greeting.

This call routing should also apply to PSTN calls.

Any idea how to achieve this???


- Tanmay


Re: [OSL | CCIE_Voice] Unity Express

2009-03-03 Thread Alex Arseniev
ephone-dn 12
number 2
call-forward all voicemail DN
!
Additionally, either:
1/ In CUE, add 2 as E.164 number for each mailbox
2/ use translation-profile to translate redirect-number from 2 to 
on CUE dial-peer.
Rgds
Alex

2009/3/3 Tanmay Devare tanmaypdev...@gmail.com

 Hi,

 Integrated CUE with CME.

 Q : User can press 2 (where  = actual DN of IP phone which is
 registered to local CME) to reach user's voicemail box greeting.

 This call routing should also apply to PSTN calls.

 Any idea how to achieve this???


 - Tanmay


[OSL | CCIE_Voice] frame-relay

2009-03-03 Thread hasan khalife

 frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be 0
 frame-relay mincir 729600

 

 

 

where we should get the value 729600  from ?

 

there is a table or rules ?

 

 

thx 

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Re: [OSL | CCIE_Voice] frame-relay

2009-03-03 Thread marwa
from QoS SRND


  - Original Message - 
  From: hasan khalife 
  To: ccie_voice@onlinestudylist.com 
  Sent: Tuesday, March 03, 2009 2:41 PM
  Subject: [OSL | CCIE_Voice] frame-relay


   frame-relay cir 729600
   frame-relay bc 7296
   frame-relay be 0
   frame-relay mincir 729600
   
   
   
  where we should get the value 729600  from ?
   
  there is a table or rules ?
   
   
  thx 


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Re: [OSL | CCIE_Voice] frame-relay

2009-03-03 Thread James Key
It is 95% of the CIR.


James

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of hasan khalife
Sent: Tuesday, March 03, 2009 6:41 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] frame-relay

 frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be 0
 frame-relay mincir 729600



where we should get the value 729600  from ?

there is a table or rules ?


thx

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Re: [OSL | CCIE_Voice] call group of DN number to conference at a time

2009-03-03 Thread Cliff McGlamry
On CallManager by itself?  No.  But you could have a meet me conference number 
on CCM you could have them call. 

A better solution is MeetingPlace.


- Original Message - 
  From: Balamurugan Singaram 
  To: ccie_voice@onlinestudylist.com 
  Sent: Tuesday, March 03, 2009 4:41 AM
  Subject: [OSL | CCIE_Voice] call group of DN number to conference at a time


Hi,

IF I want to call group of DN number to conference at a time, I can 
dial one by one and call each one.
There is any simple way like hunt group to call all of them at once for 
conference, please light on this.
 
For ex in a company they have sales team, here manager want to call the 
whole team just by dialing a single number, any workaround for this ?
 
 
Thanks,
Bala. 


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Re: [OSL | CCIE_Voice] V3 Lab1A?

2009-03-03 Thread Mark Snow
You know what - I realized the mistake - and now all routers initial  
configs for Lab1A are loaded into the vRack interface.

There is simply no initial config to be loaded for UCM.

Thanks!

--
Mark Snow
CCIE #14073 (Voice, Security)

Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: ms...@ipexpert.com
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Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
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On Mar 3, 2009, at 1:49 PM, Bryan Brooks wrote:


Hi Mark,

Thank you for the response and taking care of the PSTN initial  
configuration.  I may have missed it but is the dlci information in  
the LAB1A?  I only remember seeing the tasks to setup VLAN, DHCP and  
NTP.


Thanks

Bryan Brooks

On Tue, Mar 3, 2009 at 12:41 AM, Mark Snow ms...@ipexpert.com wrote:
No, there shouldn't. :) Well - I change that - PSTN should be setup.  
I will create an Initial for Lab1A that takes care of the PSTN  
right now. (OK - done.)
It is the only lab there isn't any other initial config for other  
than the PSTN. You see - it is Lab 1A - Infrastructure - there is  
nothing to be put in for an Initial config (revised to say except  
for the PSTN :). You are meant to do all of the work :).


Have fun!
:D
(p.s. - it's easy infrastructure stuff ;-))


--
Mark Snow
CCIE #14073 (Voice, Security)

Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: ms...@ipexpert.com
--
Join our free online support and peer group communities: 
http://www.IPexpert.com/communities
--
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On-Demand and Audio Certification Training Tools for the Cisco CCIE  
RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice  
Lab and CCIE Storage Lab Certifications.

--




On Mar 1, 2009, at 10:31 AM, Bryan Brooks wrote:

I have been working on the V3 Labs for a couple of weeks and was  
just curious.  Shouldn't there be an initial configuration that we  
can load on the devices for Lab1a?  The only option for  
configurations we have to load are the final configs.  When first  
connecting to the rack there is nothing configured such as routing  
and the PSTN router does not even appear to be up.  I find that I  
have to load the final configs and remove the vlan, dhcp etc. to  
practice the first lab.  Am I  missing something?  Any insight  
would be appreciated.


Thanks

Bryan Brooks







[OSL | CCIE_Voice] IP PIM

2009-03-03 Thread hasan khalife

WHAT IS THE DIFFERENEC BTW IP PIM DENSE-MODE 

 

 

IP PIM SPARSE-DENSE-MODE 

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Re: [OSL | CCIE_Voice] ios conference

2009-03-03 Thread Cliff McGlamry
assuming it isn't registered with SCCP,


dspfarm profile 1 conference
shut
(you'l then have to acknowledge you want to do this)

no dspfarm profile 1 conference

  - Original Message - 
  From: hasan khalife 
  To: ccie_voice@onlinestudylist.com 
  Sent: Tuesday, March 03, 2009 2:15 PM
  Subject: [OSL | CCIE_Voice] ios conference


  Pod14-BR1-RTR(config)#dspfarm profile 1 ?
conference  Profile type Conference
mtp Profile type MTP
transcode   Profile type Transcoding
cr
  Pod14-BR1-RTR(config)#dspfarm profile 1 conference?
  conference  
  Pod14-BR1-RTR(config)#dspfarm profile 1 conference
  Profile id 1 is being used for service MTP 
   please select a different profile id

   
   
   
   
  how to disable the profile id for mtp ,i didnt create it !
   
  i want to associate profile 1 register mtp..


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[OSL | CCIE_Voice] unity voice mail

2009-03-03 Thread hasan khalife

where i can fid the file that record in unity ?

 

thx

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[OSL | CCIE_Voice] Lab attack order.

2009-03-03 Thread CCIELabRat
I'm curious how people have approached the order of completing a lab.
What order do you use to ensure gathering the most points and build the lab
correctly without doubling back to adjust things as you get further into the
lab.

I know Vik and Mark have their own opinions on this, but I wanted to throw
it out to the more general audience for feedback.


Re: [OSL | CCIE_Voice] frame-relay

2009-03-03 Thread SYED HUSSAIN
There is a formula to calculate this,

(BW * 95/100) * 1000

For instance BW=768

(768 * 0.95) * 1000 = 729600

Sheeraz





From: hasan khalife hasan_khal...@hotmail.com
To: ccie_voice@onlinestudylist.com
Sent: Tuesday, March 3, 2009 7:41:14 AM
Subject: [OSL | CCIE_Voice] frame-relay

  frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be 0
 frame-relay mincir 729600
 
 
 
where we should get the value 729600  from ?
 
there is a table or rules ?
 
 
thx 


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Re: [OSL | CCIE_Voice] frame-relay

2009-03-03 Thread SYED HUSSAIN
to add in my last reply,

If nothing specified then 5% is the default tolerance as per QoS SRND, but if 
it's specified in question then you should take that value.

Again, formula is

(BW * 95/100) * 1000


Sheeraz





From: hasan khalife hasan_khal...@hotmail.com
To: ccie_voice@onlinestudylist.com
Sent: Tuesday, March 3, 2009 7:41:14 AM
Subject: [OSL | CCIE_Voice] frame-relay

  frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be 0
 frame-relay mincir 729600
 
 
 
where we should get the value 729600  from ?
 
there is a table or rules ?
 
 
thx 


Invite your mail contacts to join your friends list with Windows Live Spaces. 
It's easy! Try it!

Re: [OSL | CCIE_Voice] Lab attack order.

2009-03-03 Thread Chris Parker
I think everyone has their own way of doing it, but here's how I go 
about it:


STEP ZERO - read the whole lab carefully. It's hard to do when you are 
nervous. Try and pay attention the the details and look for tricks. Then 
work things out like what partitions and css you'll need, how the 
gatekeeper will work. Do you need multicast. Try and get you head around 
the lab as much as possible.


1. Gather info

This is where I log into everything and look at CDP to get the MAC 
addresses of the phones and of the 6608 devices. I put all of this info 
into notepad for easy cut and paste. I hardly use any paper in the lab 
everything goes into notepad for easy access and no retyping.


2. 6500

Next I set up the 6500. I do all the vlans, aux vlans, voice ports, 
muulticast, and any QoS


3. HQ

Here I set up interfaces, NTP, timezone, DHCP if need be, multicast if 
needed, QoS and the basic Gatekeeper config. I like doing all the QoS at 
the beginning of the lab, and by doing it on HQ first it helps with time 
because you can just cut and paste what you do on HQ to BR1 and BR2. I 
set up the GK enough so that I know the CME and UCM will register in the 
correct zones. Also if I see I need VIA-GK I go ahead and set up IPIPGW, 
dial peers and transcoder (as needed) on HQ. I go back and tweak the GK 
config later after I have set up UCM and CME.


4. BR1

I paste in QoS config copied from HQ, create the vlans, create 
interfaces, set helper address and multicast if needed, set up switch 
ports, set up dhcp if needed, set up ntp and timezone, set isdn 
switchtype, configure PRI for MGCP or H323, tweak isdn settings on 
serial, set up mgcp if needed, set up translation rules, set up CoR if 
needed, configure dial peers if needed for H323 and/or SRST, config 
transcoder and conf bridge if needed, set up SRST.


5. BR2

I paste in QoS config copied from HQ, create interfaces, set helper 
address and multicast if needed, set up dhcp if needed, set up ntp and 
timezone, set isdn switchtype, configure PRI, tweak isdn settings on 
serial, set voice service to allow h323 to sip, set up translation 
rules, set up CoR if needed, configure dial peers (these can often be 
copied with minor tweaks from BR1), config transcoder and conf bridge if 
needed (another section copied fro BR1), run telephony service setup, 
paste mac address from step 1 to appropriate ephone, set up H323 gateway 
and confirm registered to HQ, set voip dial peers for HQ GK and CUE, add 
ephone-dns for MWI, set up BACD, add any hunt groups as needed.


6. CUE

I normally do this step while doing something else at the same time like 
setting up the 3550. There's alot of waiting in this step while CUE just 
does its thing. Once CUE is up and ready to be configured, I normally 
do the basic setup with the GUI (this will set up mwi, and your sip 
triggers and applications) and then do user creation from the CUE CLI.


7. 3550

Here I do vlans and set up ports, and do any QoS.

8. CME Testing

So at this point 90% of the CLI work is done and all the CME/CUE stuff 
is complete. So here I do some quick testing on CME dialing in / out to 
PSTN, CUE MWI, and BACD.


9. Callmanager Basic

So here Im doing all the CM stuff I can think of except call routing and 
phones. I go into serviceability first and turn on whatever services I 
need. Then I go back to admin. I use the Top Left to Right method. So 
System menu first and usually touch everything except Device Defaults. 
Then in Route Plan I do AAR, Partitions, CSS, and I try to do any  
translation patterns that I think I'll need. Then I jump over to the 
Device menu. I do any custom soft keys or button templates. I set up EM 
profiles if I need them. I set up all my gateways and make sure they 
register, I set up the GK and trunk I set up any CTI route points I 
need. I DO NOT set up the phones yet however. Then I move to the Feature 
Menu. I setup my phone services, voice mail, and park or pickup if need 
be. Then I go to the services menu. I setup all the media resources, 
lists and groups and make sure everything is registered and in the right 
device pool. After the media stuff is done I go back to the device pools 
and assign the MRGL. I always do IPMA or AC later after the phones are 
set up.


10. Eat Lunch

Yep all the stuff listed above needs to be done before lunch.

11. Callmanager routing and Phones

So this part is all about the Route Groups, Route Lists, and Route 
Patterns. I plan most of this out in step zero or at lunch. Once thats 
all done I start setting up the phones and users. If I did everything 
right in step 9 and I can do everything I need to on every phones 
without going back and add partitions or what not. When this is done I 
do some quick basic testing with the PSTN.


12. Gatekeeper

Now that everything is built on CCM and CME and everything is registered 
to the GK, I finish the GK config. This could mean adding zone prefixes, 
aliases, bandwidth commands etc. Then I test 

[OSL | CCIE_Voice] MOH problem - but I managed to figure it out....but useful info

2009-03-03 Thread Cliff McGlamry
Running into a strange issue this evening on the rack.  

Streaming multicast MOH from HQ to BR1.  Performace counters show the MOH 
source active on the server, but PSTN caller into BR1 doesn't get music.  

I've done the ip pim sparse-dense on the Loopback, Vlan, and on the 
virtual-template interface (this is MLPoFR) for BR1.  

On BR1, sh ip mroute returns the info you'd expect:


(*, 239.2.1.1), 00:05:37/stopped, RP 0.0.0.0, flags: D 
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
Virtual-Access3, Forward/Dense, 00:05:37/00:00:00

(10.21.201.1, 239.2.1.1), 00:05:36/00:00:24, flags: T
  Incoming interface: Vlan410, RPF nbr 0.0.0.0
  Outgoing interface list:
Virtual-Access3, Forward/Dense, 00:02:32/00:00:00

(172.21.101.1, 239.2.1.1), 00:05:37/00:00:32, flags: T
  Incoming interface: Loopback0, RPF nbr 0.0.0.0
  Outgoing interface list:
Virtual-Access3, Forward/Dense, 00:02:33/00:00:00

(*, 224.0.1.40), 01:19:01/00:02:24, RP 0.0.0.0, flags: DCL
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
Virtual-Access1, Forward/Dense, 01:19:01/00:00:00
Virtual-Access3, Forward/Dense, 01:19:12/00:00:00


So the router KNOWS about it.  But when the call goes on hold, the stream isn't 
coming across.  I'm really not seeing any packets across the WAN while the 
caller is on hold, so I don't think it's actually sending it across.  But 
callers into HQ DO get multicast MOH, but of course, they are on the same VLAN 
so it's easy.  

So I start wonderingwhere the heck is it getting stuck?  I've got the BR1 
router set up with the 

no mgcp timer receive-rtcp
ip pim-dense
no ip igmp snooping

I'm thinkingthis isn't the problem.  So, I back up and start looking at HQ 
router.  It also sees the PIM info...:

Pod21-HQ-RTR#sh ip mroute
IP Multicast Routing Table
Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C - Connected,
   L - Local, P - Pruned, R - RP-bit set, F - Register flag,
   T - SPT-bit set, J - Join SPT, M - MSDP created entry,
   X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,
   U - URD, I - Received Source Specific Host Report,
   Z - Multicast Tunnel, z - MDT-data group sender,
   Y - Joined MDT-data group, y - Sending to MDT-data group
Outgoing interface flags: H - Hardware switched, A - Assert winner
 Timers: Uptime/Expires
 Interface state: Interface, Next-Hop or VCD, State/Mode

(*, 239.2.1.1), 00:04:33/00:01:26, RP 0.0.0.0, flags: D
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
Virtual-Access3, Forward/Dense, 00:04:33/00:00:00

(*, 239.2.1.3), 00:00:16/stopped, RP 0.0.0.0, flags: D
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
Virtual-Access3, Forward/Dense, 00:00:16/00:00:00

(162.21.101.2, 239.2.1.3), 00:00:17/00:02:42, flags: PT
  Incoming interface: Virtual-Access3, RPF nbr 0.0.0.0
  Outgoing interface list: Null

(*, 224.0.1.40), 05:50:11/00:02:45, RP 0.0.0.0, flags: DCL
  Incoming interface: Null, RPF nbr 0.0.0.0

So...I wonderit couldn't be that simple

I go in and apply the following command onto the fas 0/0.410 interface:

ip pim dense

Make a test calland we have audio.  

While this allows it to flow correctly, I'm still not clear on how it could 
KNOW it was there but not flow...but at least I won't get killed by this 
situation again.  

Cliff



Re: [OSL | CCIE_Voice] Lab attack order.

2009-03-03 Thread CCIELabRat
Chris ,
Great layout and exactly what I was looking for.

Thanks


On Tue, Mar 3, 2009 at 10:27 PM, Chris Parker cpar...@cparker.us wrote:

 I think everyone has their own way of doing it, but here's how I go about
 it:

 STEP ZERO - read the whole lab carefully. It's hard to do when you are
 nervous. Try and pay attention the the details and look for tricks. Then
 work things out like what partitions and css you'll need, how the gatekeeper
 will work. Do you need multicast. Try and get you head around the lab as
 much as possible.

 1. Gather info

 This is where I log into everything and look at CDP to get the MAC
 addresses of the phones and of the 6608 devices. I put all of this info into
 notepad for easy cut and paste. I hardly use any paper in the lab everything
 goes into notepad for easy access and no retyping.

 2. 6500

 Next I set up the 6500. I do all the vlans, aux vlans, voice ports,
 muulticast, and any QoS

 3. HQ

 Here I set up interfaces, NTP, timezone, DHCP if need be, multicast if
 needed, QoS and the basic Gatekeeper config. I like doing all the QoS at the
 beginning of the lab, and by doing it on HQ first it helps with time because
 you can just cut and paste what you do on HQ to BR1 and BR2. I set up the GK
 enough so that I know the CME and UCM will register in the correct zones.
 Also if I see I need VIA-GK I go ahead and set up IPIPGW, dial peers and
 transcoder (as needed) on HQ. I go back and tweak the GK config later after
 I have set up UCM and CME.

 4. BR1

 I paste in QoS config copied from HQ, create the vlans, create interfaces,
 set helper address and multicast if needed, set up switch ports, set up dhcp
 if needed, set up ntp and timezone, set isdn switchtype, configure PRI for
 MGCP or H323, tweak isdn settings on serial, set up mgcp if needed, set up
 translation rules, set up CoR if needed, configure dial peers if needed for
 H323 and/or SRST, config transcoder and conf bridge if needed, set up SRST.

 5. BR2

 I paste in QoS config copied from HQ, create interfaces, set helper address
 and multicast if needed, set up dhcp if needed, set up ntp and timezone, set
 isdn switchtype, configure PRI, tweak isdn settings on serial, set voice
 service to allow h323 to sip, set up translation rules, set up CoR if
 needed, configure dial peers (these can often be copied with minor tweaks
 from BR1), config transcoder and conf bridge if needed (another section
 copied fro BR1), run telephony service setup, paste mac address from step 1
 to appropriate ephone, set up H323 gateway and confirm registered to HQ, set
 voip dial peers for HQ GK and CUE, add ephone-dns for MWI, set up BACD, add
 any hunt groups as needed.

 6. CUE

 I normally do this step while doing something else at the same time like
 setting up the 3550. There's alot of waiting in this step while CUE just
 does its thing. Once CUE is up and ready to be configured, I normally do
 the basic setup with the GUI (this will set up mwi, and your sip triggers
 and applications) and then do user creation from the CUE CLI.

 7. 3550

 Here I do vlans and set up ports, and do any QoS.

 8. CME Testing

 So at this point 90% of the CLI work is done and all the CME/CUE stuff is
 complete. So here I do some quick testing on CME dialing in / out to PSTN,
 CUE MWI, and BACD.

 9. Callmanager Basic

 So here Im doing all the CM stuff I can think of except call routing and
 phones. I go into serviceability first and turn on whatever services I need.
 Then I go back to admin. I use the Top Left to Right method. So System menu
 first and usually touch everything except Device Defaults. Then in Route
 Plan I do AAR, Partitions, CSS, and I try to do any  translation patterns
 that I think I'll need. Then I jump over to the Device menu. I do any custom
 soft keys or button templates. I set up EM profiles if I need them. I set up
 all my gateways and make sure they register, I set up the GK and trunk I set
 up any CTI route points I need. I DO NOT set up the phones yet however. Then
 I move to the Feature Menu. I setup my phone services, voice mail, and park
 or pickup if need be. Then I go to the services menu. I setup all the media
 resources, lists and groups and make sure everything is registered and in
 the right device pool. After the media stuff is done I go back to the device
 pools and assign the MRGL. I always do IPMA or AC later after the phones are
 set up.

 10. Eat Lunch

 Yep all the stuff listed above needs to be done before lunch.

 11. Callmanager routing and Phones

 So this part is all about the Route Groups, Route Lists, and Route
 Patterns. I plan most of this out in step zero or at lunch. Once thats all
 done I start setting up the phones and users. If I did everything right in
 step 9 and I can do everything I need to on every phones without going back
 and add partitions or what not. When this is done I do some quick basic
 testing with the PSTN.

 12. Gatekeeper

 Now that everything is built on CCM and CME 

Re: [OSL | CCIE_Voice] MOH problem - but I managed to figure it out....but useful info

2009-03-03 Thread anil batra
Add -
 
ccm-manager music-on-hold.au

--- On Wed, 3/4/09, Cliff McGlamry cl...@mcglamry.net wrote:

From: Cliff McGlamry cl...@mcglamry.net
Subject: [OSL | CCIE_Voice] MOH problem - but I managed to figure it outbut 
useful info
To: ccie_voice@onlinestudylist.com
Date: Wednesday, March 4, 2009, 10:36 AM





Running into a strange issue this evening on the rack.  
 
Streaming multicast MOH from HQ to BR1.  Performace counters show the MOH 
source active on the server, but PSTN caller into BR1 doesn't get music.  
 
I've done the ip pim sparse-dense on the Loopback, Vlan, and on the 
virtual-template interface (this is MLPoFR) for BR1.  
 
On BR1, sh ip mroute returns the info you'd expect:
 

(*, 239.2.1.1), 00:05:37/stopped, RP 0.0.0.0, flags: D 
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
    Virtual-Access3, Forward/Dense, 00:05:37/00:00:00
 
(10.21.201.1, 239.2.1.1), 00:05:36/00:00:24, flags: T
  Incoming interface: Vlan410, RPF nbr 0.0.0.0
  Outgoing interface list:
    Virtual-Access3, Forward/Dense, 00:02:32/00:00:00
 
(172.21.101.1, 239.2.1.1), 00:05:37/00:00:32, flags: T
  Incoming interface: Loopback0, RPF nbr 0.0.0.0
  Outgoing interface list:
    Virtual-Access3, Forward/Dense, 00:02:33/00:00:00
 
(*, 224.0.1.40), 01:19:01/00:02:24, RP 0.0.0.0, flags: DCL
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
    Virtual-Access1, Forward/Dense, 01:19:01/00:00:00
    Virtual-Access3, Forward/Dense, 01:19:12/00:00:00

 
So the router KNOWS about it.  But when the call goes on hold, the stream isn't 
coming across.  I'm really not seeing any packets across the WAN while the 
caller is on hold, so I don't think it's actually sending it across.  But 
callers into HQ DO get multicast MOH, but of course, they are on the same VLAN 
so it's easy.  
 
So I start wonderingwhere the heck is it getting stuck?  I've got the BR1 
router set up with the 
 
no mgcp timer receive-rtcp
ip pim-dense
no ip igmp snooping
 
I'm thinkingthis isn't the problem.  So, I back up and start looking at HQ 
router.  It also sees the PIM info...:
 
Pod21-HQ-RTR#sh ip mroute
IP Multicast Routing Table
Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C - Connected,
   L - Local, P - Pruned, R - RP-bit set, F - Register flag,
   T - SPT-bit set, J - Join SPT, M - MSDP created entry,
   X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,
   U - URD, I - Received Source Specific Host Report,
   Z - Multicast Tunnel, z - MDT-data group sender,
   Y - Joined MDT-data group, y - Sending to MDT-data group
Outgoing interface flags: H - Hardware switched, A - Assert winner
 Timers: Uptime/Expires
 Interface state: Interface, Next-Hop or VCD, State/Mode
 
(*, 239.2.1.1), 00:04:33/00:01:26, RP 0.0.0.0, flags: D
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
    Virtual-Access3, Forward/Dense, 00:04:33/00:00:00
 
(*, 239.2.1.3), 00:00:16/stopped, RP 0.0.0.0, flags: D
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
    Virtual-Access3, Forward/Dense, 00:00:16/00:00:00
 
(162.21.101.2, 239.2.1.3), 00:00:17/00:02:42, flags: PT
  Incoming interface: Virtual-Access3, RPF nbr 0.0.0.0
  Outgoing interface list: Null
 
(*, 224.0.1.40), 05:50:11/00:02:45, RP 0.0.0.0, flags: DCL
  Incoming interface: Null, RPF nbr 0.0.0.0
 
So...I wonderit couldn't be that simple
 
I go in and apply the following command onto the fas 0/0.410 interface:
 
ip pim dense
 
Make a test calland we have audio.  
 
While this allows it to flow correctly, I'm still not clear on how it could 
KNOW it was there but not flow...but at least I won't get killed by this 
situation again.  
 
Cliff