Re: [OSL | CCIE_Voice] RE : Mobile voice access a sking userid rather then pin

[kapil atrish]

There is a CCM Service parameter to check calling no for MVA. You may set it to 
partial match to avoid issues with not having exact digit by digit match. See 
if it helps.
 


--- On Thu, 6/24/10, naoufal.kerboute  wrote:


From: naoufal.kerboute 
Subject: [OSL | CCIE_Voice] RE : Mobile voice access asking userid rather then 
pin
To: laurent.bourm...@orange-ftgroup.com, "." 
Date: Thursday, June 24, 2010, 10:04 PM



check the caller id coming from pstn to UCM and youe remote destination number.
May be you need to apply any transformation in the calling number to match the 
remote destination number


 Message d'origine
De: ccie_voice-boun...@onlinestudylist.com de la part de 
laurent.bourm...@orange-ftgroup.com
Date: jeu. 24/06/2010 18:30
À: .
Objet : [OSL | CCIE_Voice] Mobile voice access asking userid rather then pin

Hi,

It seems that I have an issue getting the MVA oerational.

Actually I use a H323 gatway connected to the CUCM, so when I call from my 
remote destination I have the prompt saying " Please enter your userid ..." 
rather than getting the pin prompt.

I did a trace on the CUCM then it seems that my remote destination is properly 
recognized :

2010-06-24 18:21:01,269 DEBUG [http-8080-Processor24] 
controller.IVRGetAudioFile - [CCM_IVR]:: IVR get Filename
2010-06-24 18:21:01,270 DEBUG [http-8080-Processor24] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile()
2010-06-24 18:21:01,271 DEBUG [http-8080-Processor24] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 1.au and  
locale = en_US
2010-06-24 18:21:01,271 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::execute(): action start
2010-06-24 18:21:01,272 DEBUG [http-8080-Processor24] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 1.au and  
locale = en_US
2010-06-24 18:21:01,273 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::srcdir: en_US
2010-06-24 18:21:01,274 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::-new-code-got-callerid-as-remotedest: 
+447976852817
2010-06-24 18:21:01,275 DEBUG [http-8080-Processor24] 
controller.IVRGetAudioFile - [CCM_IVR]:: getting the file now
2010-06-24 18:21:01,275 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::srcdir: en_US   
sessionid:1472FAADA95E59CFE334AD38AFA395FF
2010-06-24 18:21:01,277 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::remotedest: +447976852817   
sessionid:1472FAADA95E59CFE334AD38AFA395FF
2010-06-24 18:21:01,278 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::Host: 10.100.210.11:8080
2010-06-24 18:21:01,279 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::Content-Type 
application/x-www-form-urlencoded
2010-06-24 18:21:01,279 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::Connection: close
2010-06-24 18:21:01,280 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::Accept: text/vxml, text/x-vxml, 
application/vxml, application/x-vxml, application/voicexml, 
application/x-voicexml, text/plain, text/html, audio/basic, audio/wav, 
multipart/form-data, application/octet-stream
2010-06-24 18:21:01,280 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::User-Agent: Cisco-IOS-C3845/12.4
2010-06-24 18:21:01,281 DEBUG [http-8080-Processor23] util.IVRDBInterface - 
[CCM_IVR]<--- getUserIdFromRemoteDestination- DB Query -->   remoteDest  = 
+447976852817
2010-06-24 18:21:01,285 DEBUG [http-8080-Processor23] util.IVRDBInterface - 
[CCM_IVR]<--- isPartialMatchEnabled --DB Query -->   sql:  = select paramValue 
from processconfig where paramName ='RemDestCallerIDMatchType' and tkService 
='0'
2010-06-24 18:21:01,325 DEBUG [http-8080-Processor23] util.IVRDBInterface - 
[CCM_IVR]<--- isPartialMatchEnabled --DB Query -->   RemDestCallerIDMatchType:  
= 1
2010-06-24 18:21:01,327 DEBUG [http-8080-Processor23] util.IVRDBInterface - 
[CCM_IVR]<--- getUserIdFromRemoteDestination()  -- partial match enabled
2010-06-24 18:21:01,328 DEBUG [http-8080-Processor23] util.IVRDBInterface - 
[CCM_IVR]<--- getNumDigitsPartialMatch --DB Query -->   sql:  = select 
paramValue from processconfig where paramName ='RemDestCallerIDMatchDigits' and 
tkService ='0'
2010-06-24 18:21:01,329 DEBUG [http-8080-Processor23] util.IVRDBInterface - 
[CCM_IVR]IVRDBInterface.getUserIdFromRemoteDestination() 
-->java.lang.NullPointerException
2010-06-24 18:21:01,329 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::ccmusername: null
2010-06-24 18:21:01,330 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]::ccmusername:is Null ?? null
2010-06-24 18:21:01,331 DEBUG [http-8080-Processor23] 
controller.IVRCalleridLookup - [CCM_IVR]IVRCallerIdLookup::execute(): 
ccmusername is Null and action forwarding IVRUserid 
2010-06-24 18:21:04,631 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile -

[OSL | CCIE_Voice] vouchers

[kapil atrish]

Hi List,
 
I've discussed this with PL team and taken their permission before posting this.
 
I've around 20 odd vouchers available at minimal price. Those are left 
overs after passing my lab. If anyone interested pl PM me. All vouchers are 
valid for V3.
 
Thanks


  ___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] V3 Attempt Two

[kapil atrish]

I cracked mine on 3rd go in V2. Be consistent...good luck

--- On Wed, 9/2/09, Ravindra Lakpriya  wrote:


From: Ravindra Lakpriya 
Subject: Re: [OSL | CCIE_Voice] V3 Attempt Two
To: "Tanner Ezell" 
Cc: ccie_voice@onlinestudylist.com
Date: Wednesday, September 2, 2009, 11:00 AM


Let's hope for the best man. U ll nail it. All the best dude.

On Wednesday, September 2, 2009, Tanner Ezell  wrote:
> Good luck Jonathan, look forward to hearing the results!
>
> On Tue, Sep 1, 2009 at 9:53 PM, Jonathan Charles  wrote:
> OK, took v3 again in RTP today... finished 30 minutes early...
>
> Well, not really... what happened was that I was doing some last
> minute tweaking (just retesting stuff, cleaning up some config) and
> some key huge point items stopped working... I undid what I did to
> break stuff, got up and walked away... yes, there was 30 minutes on
> the table, but it could have been the death of me...
>
> Anyway, waiting on results I would like to claim optimism, as I
> studied the crap out of my shortcomings last time, but I have done
> this before where I walked out of a lab pretty confident to see zero
> on sections I thought I aced... to be honest, I am like 85% sure I
> failed again.
>
> As they all say, the test is fair, nothing out of left field, some
> surprises on what was on there and what wasn't... there are some
> sleazy traps, but if you have a clue, you will work around em pretty
> quick...
>
>
> Took the first one in SJ, took this one in RTP... so, I can compare...
>
> In SJ, the phones are nailed to the walls in the cubicle... in RTP,
> they are on the desk (so you can flip em over and look at em...)...
> not sure which I prefer... I kinda like throwing them at the wall...
> But then again, in SJ, Ben Ng is sitting 4 feet from you, so, no
> intimidation there...
>
> I saw the remnants of the old v2 labs sitting in RTP, still had phones
> and fax machines... looked abandoned...
>
> Everything else I could say would be NDA... so, guys, do what you
> always do, look for the flurry of questions on 'how do I" in this
> group or as veiled customer issues on Puck
>
> As a joke, here are the four questions I would ask:
>
> Why on this day are we limited on how we can dial? When on all other
> days we can dial however we want?
> Why on this day must we use frame-relay, when on all other days, all
> of our customers have MPLS?
> Why on this day are we running unpatched, basically beta-versions of
> CUCM, CUPS, CUCCX, when on other days we can install patches to get
> around bugs?
> Why on this day do I have to fly all the way to Raleigh and start the
> test at 7:15AM, when the guys who go to San Jose get to sleep in and
> take their test at 9:00?
>
>
>
>
>
> Jonathan
> If you are Jewish, those are funny.
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>

-- 
Ravindra Lakpriya
+94 773 532 094
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com



  ___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Thanks

[kapil atrish]

Hi List,
 
The journey I started last year with a CCIE Voice boot-camp from a freelancer. 
The boot camp was little frustrating and the only good thing was that the 
instructor introduced me to the PL and OSL. 
 
During my earlier attempt I practiced on a small Dynamips/VM Ware simulated 
home-lab in addition to PL Remote racks. I also got hold of some hardware and 
built a rack for practice.
 
Covering the technology was not a problem in the lab, but there was no defined 
strategy to attack. In the lab exam, I started covering section by section. By 
the end of the lab it was whole mess in my mind and fixing even smallest of 
issues were taking longer then expected. I didn't have the testing strategy as 
well. I couldn't pass that time, not even close to passing.
 
This time I worked a lot on the test taking and testing strategy both. My 
recommendation in addition to covering the technology is to work on the attack 
order you are going to follow in the lab, Atleast that was the problem for me. 
Also don't forget to have a workable testing strategy in place, that could be 
the difference between pass/fail.
 
I would also recommend to go through the Online Voice VLectures available at 
http://www.ipexpert.com/index.cfm/a/p/vlectures. There is lot of relevant 
information available in those vlectures.
 
 
Wish good luck to all,
 
Thanks,
Kapil Atrish



--- On Mon, 6/29/09, Cristobal Priego  wrote:


From: Cristobal Priego 
Subject: Re: [OSL | CCIE_Voice] Thanks
To: "kapil atrish" 
Cc: ccie_voice@onlinestudylist.com
Date: Monday, June 29, 2009, 6:04 AM



congratulations
 
could you share your expirience with us
 
what tips could you provide?


2009/6/28 kapil atrish 







Hi list,
 
I've passed the lab exam and would like to thanks each one of you for your 
contributions to this list. 
 
I would also like to extend  my thanks to PL After-hour support team for 
providing the instant support on various issues I faced during practice 
sessions.
 
Thanks,
Kapil Atrish
CCIE Voice# 24706




  

[OSL | CCIE_Voice] Thanks

[kapil atrish]

Hi list,
 
I've passed the lab exam and would like to thanks each one of you for your 
contributions to this list. 
 
I would also like to extend  my thanks to PL After-hour support team for 
providing the instant support on various issues I faced during practice 
sessions.
 
Thanks,
Kapil Atrish
CCIE Voice# 24706


  

Re: [OSL | CCIE_Voice] GDM configuration with notification on2phones

[kapil atrish]

xlate the mwi to say .

Create two ephone-dns:

ephone-dn 1
number 
label 3101
!
ephone-dn 2
number 
label 3102
!
Apply ephone-dn's using regular overlay option, keep in mind to put dn 1 and 2 
as first option against respective buttons. That way, you'll meet the phone 
display requirement and still get the mwi envelope...Don't have lab to test it, 
but not sure why it won't work.


--- On Tue, 6/23/09, Michael Ciarfello  wrote:

From: Michael Ciarfello 
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on2phones
To: "kapil atrish" , "Cristi Radescu" 

Cc: "ccie_voice@onlinestudylist.com" 
Date: Tuesday, June 23, 2009, 7:50 AM




 
 







What about 3101 requirement? 

   

   



From: kapil atrish
[mailto:nice_cha...@yahoo.com] 

Sent: Friday, June 12, 2009 4:50 AM

To: Cristi Radescu; Michael Ciarfello

Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification
on2phones 



   


 
  
  Yes, its possible. I've tried it and it works.

  

  Assume u want the envelope on 3102 which is a secondary line on the phone and
  mwi requests are coming for 3100.

  

  voice translation-rule 1

  rule 1 /^\(80003100\)/ /80003102/

  rule 2 /^\(80013100\)/ /80013102/

  ! 

  voice translation-profile mwi

  translate called 1

  !

  dial-peer voice 123 voip

  session protocol sipv2

  session target ipv4:CUE_IP_ADD

  destination-patt VM_PILOT

  incoming called-nu 800[0-1]...

  translation-prof outgoing mwi

  codec g711

  no vad

  dtmf-relay sip-notify

  !

  

  The MWI will get Xlated to 3102, and since its on line 2, u'll get only the
  envelope and not the Light.

  

  

  

  --- On Thu, 6/11/09, Michael Ciarfello 
  wrote: 
  

  From: Michael Ciarfello 

  Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on2phones

  To: "Cliff McGlamry" , "Cristi
  Radescu" , "ccie_voice@onlinestudylist.com"
  

  Date: Thursday, June 11, 2009, 8:03 AM 
  
  
  Well, we are thinking and using our experience to try
  things.  The CCIE is also a journey.  I didn't pass back in Feb,
  but I came out a lot sharper on the skills and basic building blocks. 
  Even these no solution scenarios are valuable because you never know what a
  customer will ask for and sometimes the answer is NO. 
  
  
    
  
  
  Wouldn't
  that be something to put no win scenarios on the lab and you have to explain
  why it doesn't work? 
  
  
    
  
  
  I'll
  shut up now before someone thinks it's a good idea.  lol. 
  
  
    
  
  
    
  
  
  
  
  
  
  
  From: Cliff McGlamry
  [cl...@mcglamry.net]

  Sent: Wednesday, June 10, 2009 11:26 AM

  To: Michael Ciarfello; Cristi Radescu; ccie_voice@onlinestudylist.com

  Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification
  on2phones 
  
  
  
  My
  understanding is that you are correct.  The answer thus far is no. 
  
  
    
  
  
  
  
  From: Michael Ciarfello
   
  
  
  Sent: Tuesday, June 09,
  2009 10:44 PM 
  
  
  To: Cliff McGlamry
  ; Cristi
  Radescu ; ccie_voice@onlinestudylist.com
   
  
  
  Subject: RE: [OSL |
  CCIE_Voice] GDM configuration with notification on2phones 
  
  
  
     
  
  
  Did this ever get solved?  Am I correct in saying the
  answer so far is NO? 
  
  
    
  
  
  
  
  
  From: ccie_voice-boun...@onlinestudylist.com
  [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry
  [cl...@mcglamry.net]

  Sent: Thursday, May 28, 2009 3:47 PM

  To: Cristi Radescu; ccie_voice@onlinestudylist.com

  Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on
  2phones 
  
  
  
  Actually,
  you might be able to do it if it is on CME. 
  
  
    
  
  
  Assign
  button 2 as an overlay, and put the mailbox number DN on the overlay
  ephone-dn.  It should be hard forwarded so it will never ring, but I bet
  that would make the envelope appear the way being discussed.   
  
  
    
  
  
  
  
  From: Cristi
  Radescu  
  
  
  Sent: Thursday, May 28,
  2009 5:22 AM 
  
  
  To: 'ccie_voice@onlinestudylist.com'
   
  
  
  Subject: Re: [OSL |
  CCIE_Voice] GDM configuration with notification on 2phones 
  
  
  
     
  
  
    
  I think
  this is not possible. With „secondary number” or „overlay” it will not work. 
    
  
  
  
  
  From: ccie_voice-boun...@onlinestudylist.com
  [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David
  Corbeil

  Sent: 27 May, 2009 8:42 PM

  To: 'ccie_voice@onlinestudylist.com'

  Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2
  phones 
  
    
  Hi, 
    
  I want
  to know if it’s possible to have the voicemail letter on the second line of 2
  phones without changing the phone facing. 
    
  Example: 
    
  Phone 1 
  Line1 :
  3001 
  Line2 :
  3101 
    
  Phone 2 
  Line1 :
  3002 
  Line2 :
  3102 
    
  Each
  line need to be access to GDM 

Re: [OSL | CCIE_Voice] Problem with outgoing calls from Branch via T1 Pri to simulated PSTN router

[kapil atrish]

That's a common issue. Create one DP on CME as following:

dial-peer voice xxx pots
incoming called-nu .
direct-inward-dial
!



--- On Thu, 5/28/09, ccieid1ot  wrote:

From: ccieid1ot 
Subject: Re: [OSL | CCIE_Voice] Problem with outgoing calls from Branch via T1 
Pri to simulated PSTN router
To: "Padmanabhan, Padhu" 
Cc: "ccie_voice@onlinestudylist.com" 
Date: Thursday, May 28, 2009, 10:20 AM

Just add another dial-peer for 911 with forward-digt all.


On Wed, May 27, 2009 at 5:09 PM, Padmanabhan, Padhu  wrote:
> Hi,
>
>
>
> I am just getting started with my voice lab preps…I have a branch router
> connected via WAN as well via T-1(crossover) to PSTN simulator running CME.
>
> I am unable to call simulated phones lines on the CME using the T-1 pri from
> branch. As soon as I dial 911 it goes into connected state from Branch-1.
>
>
>
> However from the CME /PSTN I can call inbound to the BR1 phones.
>
>
>
> Pasted below is relevant config.  BR1 router configured as h323 gw and added
> to RG-RL-RP. Pattern 911 added to partition and included in the CSS for the
>
> BR1 phone.
>
>
>
> Any ideas?.
>
>
>
> Thanks,Padhu
>
>
>
> Branch-1:
>
> isdn switch-type primary-ni
>
>
>
> voice class h323 1
>
>   h225 timeout tcp establish 3
>
>
>
> controller T1 1/0/0
>
>  pri-group timeslots 1-24
>
>  description connection from BR1 to PSTN
>
>
>
> interface Serial1/0/0:23
>
>  no ip address
>
>  encapsulation hdlc
>
>  isdn switch-type primary-ni
>
>  isdn incoming-voice voice
>
>  no cdp enable
>
>
>
> dial-peer voice 999 pots
>
>  description Calls to PSTN using Local gateway
>
>  destination-pattern .T
>
>  incoming called-number .
>
>  direct-inward-dial
>
>  port 1/0/0:23
>
> !
>
> dial-peer voice 1 voip
>
>  voice-class h323 1
>
>  incoming called-number .
>
>  dtmf-relay h245-alphanumeric
>
> !
>
> dial-peer voice 775 voip
>
>  destination-pattern 775255
>
>  voice-class h323 1
>
>  session target ipv4:10.1.40.10
>
>
>
> pstn-sim:
>
> isdn switch-type primary-ni
>
> isdn gateway-max-interworking
>
>
>
> voice class h323 1
>
>  h225 timeout tcp establish 3
>
>
>
> controller T1 0/0
>
>  framing esf
>
>  clock source internal
>
>  linecode b8zs
>
>  cablelength short 133
>
>  pri-group timeslots 1-24
>
>  description Connection to BR1
>
>
>
> interface Serial0/0:23
>
>  no ip address
>
>  encapsulation hdlc
>
>  isdn switch-type primary-ni
>
>  isdn protocol-emulate network
>
>  isdn incoming-voice voice
>
>  no cdp enable
>
>
>
> dial-peer voice 775 pots
>
>  description calls to BR1
>
>  destination-pattern 775255
>
>  incoming called-number .
>
>  port 0/0:23
>
>  forward-digits all
>
>
>
> telephony-service
>
>  max-ephones 8
>
>  max-dn 30
>
>  ip source-address 10.1.99.1 port 2000
>
>  system message CME as PSTN Simulator
>
>  max-conferences 8 gain -6
>
>  transfer-system full-consult
>
>  create cnf-files version-stamp Jan 01 2002 00:00:00
>
> !
>
> !
>
> ephone-dn  1
>
>  number 911
>
>  label Emergency_911
>
>
>
> ephone  1
>
>  device-security-mode none
>
>  mac-address 0002.FD65.9D5B
>
>  button  1:1 2:2 3:3 4:4
>
>  button  5:5 6:6



  

Re: [OSL | CCIE_Voice] ATA IVR not responding

[kapil atrish]

Hi Michael,

Can u give more inputs reg turning IVR off via config-register? Haven't heard 
abt this before, sounds interesting.

Yes the Red light does come on and goes off perfectly fine. I was able to 
make/receive calls to/from ata phone.

ATA is not reachable now can't change the software image...:(

I did shake it, opened the box and tried pressing the little button but no 
good.:(


--- On Sun, 5/3/09, Michael Ciarfello  wrote:

From: Michael Ciarfello 
Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding
To: "kapil atrish" , "ccie_voice@onlinestudylist.com" 

Date: Sunday, May 3, 2009, 9:00 AM


 
 
#yiv1591233982 P {
MARGIN-TOP:0px;MARGIN-BOTTOM:0px;}

Can you turn the IVR off via a config register?  Might be disabled.
 
When you pick up the phone does the red light come on?
 
Try the factoryreset thingie.
Try re-loading the software image.
Try shaking it.  (lol. just kidding)


From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish 
[nice_cha...@yahoo.com]

Sent: Saturday, May 02, 2009 11:31 PM

To: ccie_voice@onlinestudylist.com; Cliff McGlamry

Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding









Yes, that's what I am trying to invoke the IVR but no response. I've changed 
the cable, Phone, Phone port, removed lan cable and tried whole thing again but 
no good.



--- On Sat, 5/2/09, Cliff McGlamry  wrote:




From: Cliff McGlamry 

Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding

To: "kapil atrish" , ccie_voice@onlinestudylist.com

Date: Saturday, May 2, 2009, 11:30 PM




Did you push the button on top of the ATA after picking up the phone on port 
1?  That's how you activate the IVR menu. 

 
What exactly have you done?
 

- Original Message - 
From: 
kapil atrish 
To: 
ccie_voice@onlinestudylist.com 
Sent: Saturday, May 02, 2009 8:25 AM
Subject: [OSL | CCIE_Voice] ATA IVR not responding






Hi list,



M not able to access the IVR menu on ATA.I know it should work on Phone 1 port, 
no success on any of the port. Reboot didn't help. Removed lan cable and tried, 
no success. Checked physical layer stuff. Has anyone had this issue earlier and 
any troubleshooting
 I can do??? M running SCCP image on it.





I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA 
unreachable. Now I don't have a switch to configure in the specific vlan and 
access my ATA. Since IVR is not working I am not even able to revert the ATA 
settings.




Thanks in advance...





















 



  

Re: [OSL | CCIE_Voice] IPPA Host not found in UCM

[kapil atrish]

OK, I did point it to CCM. Let me point it to CCX and try again. Thanks for 
your reply

--- On Sat, 5/2/09, Cliff McGlamry  wrote:

From: Cliff McGlamry 
Subject: Re: [OSL | CCIE_Voice] IPPA Host not found in UCM
To: "kapil atrish" , ccie_voice@onlinestudylist.com
Date: Saturday, May 2, 2009, 11:32 PM



 
 

Did you point it to the IP address of the UCCX 
server or to the CCM server?  The IPPA URL needs to point to the UCCX 
server.  In the v2 labs, these were collocated and on the same 
server.  With the new blueprint, they will be two separate servers.  

 

  - Original Message - 
  From: 
  kapil 
  atrish 
  To: ccie_voice@onlinestudylist.com 
  
  Sent: Saturday, May 02, 2009 9:00 
AM
  Subject: [OSL | CCIE_Voice] IPPA Host not 
  found in UCM
  

  


  Hi list,

This is not for lab requirement. I've UCM 
6 with CCX 7.0. I configured IPPA Phone service as we do in ccm 4.X 
scenario, but getting "Host not found". I am in DNS less environment. 
Enterprise parameter points to IP Address. Other IP Phone service 
(Extension mobility login, single button logout) is working fine. Only 
with IPPA I get host not found,

Has anybody had this issue, any 
suggestion where should I look 
  at?

thanks...




  

Re: [OSL | CCIE_Voice] ATA IVR not responding

[kapil atrish]

Yes, that's what I am trying to invoke the IVR but no response. I've changed 
the cable, Phone, Phone port, removed lan cable and tried whole thing again but 
no good.

--- On Sat, 5/2/09, Cliff McGlamry  wrote:

From: Cliff McGlamry 
Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding
To: "kapil atrish" , ccie_voice@onlinestudylist.com
Date: Saturday, May 2, 2009, 11:30 PM



 
 

Did you push the button on top of the ATA after 
picking up the phone on port 1?  That's how you activate the IVR 
menu.  
 
What exactly have you done?
 

  - Original Message ----- 
  From: 
  kapil 
  atrish 
  To: ccie_voice@onlinestudylist.com 
  
  Sent: Saturday, May 02, 2009 8:25 
AM
  Subject: [OSL | CCIE_Voice] ATA IVR not 
  responding
  

  


  Hi list,

M not able to access the IVR menu on 
ATA.I know it should work on Phone 1 port, no success on any of the 
port. Reboot didn't help. Removed lan cable and tried, no success. 
Checked physical layer stuff. Has anyone had this issue earlier and any 
troubleshooting I can do??? M running SCCP image on it.


I was 
testing the ATA vlan stuff and put in a Vlan and OpFlag which made my 
ATA unreachable. Now I don't have a switch to configure in the specific 
vlan and access my ATA. Since IVR is not working I am not even able to 
revert the ATA settings. 

Thanks in 
advance...






  

[OSL | CCIE_Voice] IPPA Host not found in UCM

[kapil atrish]

Hi list,

This is not for lab requirement. I've UCM 6 with CCX 7.0. I configured IPPA 
Phone service as we do in ccm 4.X scenario, but getting "Host not found". I am 
in DNS less environment. Enterprise parameter points to IP Address. Other IP 
Phone service (Extension mobility login, single button logout) is working fine. 
Only with IPPA I get host not found,

Has anybody had this issue, any suggestion where should I look at?

thanks...



  

[OSL | CCIE_Voice] ATA IVR not responding

[kapil atrish]

Hi list,

M not able to access the IVR menu on ATA.I know it should work on Phone 1 port, 
no success on any of the port. Reboot didn't help. Removed lan cable and tried, 
no success. Checked physical layer stuff. Has anyone had this issue earlier and 
any troubleshooting I can do??? M running SCCP image on it.


I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA 
unreachable. Now I don't have a switch to configure in the specific vlan and 
access my ATA. Since IVR is not working I am not even able to revert the ATA 
settings. 

Thanks in advance...





  

[OSL | CCIE_Voice] CCM 7 on VMWare

[kapil atrish]

Hi list,

I want to know if CCM 7 is supported on VMWare workstation 5.0? The hardware 
I've is AMD quad-core, 4gig ram. Will that work, if someone who has tested it 
can comment please?

Thanks..







  

Re: [OSL | CCIE_Voice] Dial-peer overlapping

[kapil atrish]

I've tested in on CCM and CME but over the PSTN.

On CCM side:
Create a Voice-Mail profile "to_vm", select the Voice-mail pilot and put the 
"external phone number mask" as .
Create a Route Point with DN 22xxx, do call-forward all to VM, select the VM 
profile "to_vm". 

To make it work over the WAN, configure Translation Pattern to see the 1#.2 
(saying Tech-prefix for both sides is 1#) or Called number mask to 2.



On CME side:
Create a Xlation-rule (say rule no 1) to Xlate 24xxx to 4xxx. Create ephone-dn 
24xxx, do call-forward all to CUE.

Create a Xlation proflile (say to_vm),
translate called 1
translate redirected-called 1
!
on the CUE dial-peer, apply the Xlation profile "to_vm" in outbound direction.

Additional Xlation rule to make it work over the WAN.



--- On Tue, 3/31/09, Chris Parker  wrote:

From: Chris Parker 
Subject: Re: [OSL | CCIE_Voice] Dial-peer overlapping
To: "Duy Nguyen" 
Cc: "ccie_voice@onlinestudylist.com" 
Date: Tuesday, March 31, 2009, 4:24 PM

Duy,

OK I forgot that it is going to be an ephone dialing this number so the 
ITS is processing each digit in real time and will never let you hit the 
5th digit once its made a match on the first 4.

Chris

Duy Nguyen wrote:
> Chris,
>
> It actually keeps hitting the "Dial-peer voice 2300 voip" because of a 
> shorter pattern when I dial 2400.  I was able to achieve this by 
> making it more explicit.
>
> Allow to dial HQ phones
> !
> dial-peer voice 2300 voip
>  preference 1
>  destination-pattern [23][0-1]..
>  session target ras
>  tech-prefix 1
>  dtmf-relay h245-alphanumeric
>  ip qos dscp cs3 signaling
>  no vad
> !
>
> Allow to go to HQ Phone's VM directly
> dial-peer voice 22000 voip
>  preference 1
>  destination-pattern 22...
>  session target ras
>  tech-prefix 1
>  dtmf-relay h245-alphanumeric
>  ip qos dscp cs3 signaling
>  no vad
> !
>
> Allow to hit the CME ph's VM directly.  Also put in 24001 pattern into 
> E.164 in CUE.
> ephone-dn 20
> number 2400.
> preference 0 secondary 9
> huntstop
> call-forward all 4111
> call-waiting beep
>
> On Mon, Mar 30, 2009 at 10:31 PM, Chris Parker  > wrote:
>
>     Duy,
>
>     Remember, the digit analyzer in IOS will always match the peer
>     with the most matching digits first.
>
>     So for the first issue of forwarding to VM. The config you have is
>     exactly right. All of the extensions at that site are 4001 - 4003.
>     So when you dial 24001, it will always match that ephone first and
>     go where you want tit to go.
>
>     The same goes with the WAN dial peer (2300). The extension rages
>     at the other locations are 2001-4 and 3001-4 respectively. The
>     only time you would get into hot water is if you had an extension
>     at HQ that was 2400. Then you would have a problem.
>
>     For the third requirement the answer is simple make a peer with
>     the destination patten of: 2[23]...
>
>     Chris
>
>
>
>     Duy Nguyen wrote:
>
>         How would I achieve this?
>
>         User at Site C should press 24XXX, then it will forward to
>         user's voice mailbox greeting.
>
>         E.g. when user dial 24001, then it will be forwarding directly
>         to 4001 VM and leave a message. This call routing should work
>         over the WAN also.
>
>         My solution:
>         ephone-dn 20
>         number 24...
>         call-forward all 4111
>
>         Problem is:
>         dial-peer voice 2300 voip
>         destination-pattern [23]...
>         session target ras
>         tech-prefix 1
>         dtmf-relay h245-alphanumeric
>         ip qos dscp cs3 signaling
>         no vad
>         !
>
>         Another issue, it is also asking for the same on CCM side when
>         user dials 22XXX should go to VM directly and should also work
>         over the WAN. From CME could not dial 22XXX since it keeps
>         hitting Dial-peer voice 2300 voip.
>
>         If I change the
>         dial-peer voice 2300
>         destination-pattern [23][0128]..
>
>         Still won't achieve calling to Main site phone's VM.
>
>
>




  

Re: [OSL | CCIE_Voice] gatekeeper question

[kapil atrish]

Hi,

I've not tested this since I don't have lab access yet. I can use "endpoint 
max'conn" on GK since now I've two trunks towards CCM. But below I described 
using CCM locations.  Both should work depending on what is allowed in GK 
config snap-shot. 
gw-priority config would be straight forward:

zone-prefix GK 1* gw-priority 10 Sub_Trunk_1
zone-prefix GK 1* gw-priority 9 Pub_Trunk_2



--- On Sun, 3/29/09, CCIE OSL  wrote:

From: CCIE OSL 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "kapil atrish" 
Cc: "ccie Me" , ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 11:43 AM

kapil,
Actually, I was thinking of using AAR for the first part of the requirement. - 
As I said before, I have not tried this, I am scheculed for a proctorslab 
Monday.
For the requirement of "from HQ to SiteC via GK it will be rejected by GK and 
rerouted to pstn via 6608 on HQ".
I should be able to create AAR group for BR2 and apply it to the trunk. this 
way I can reserve the 4 digit HQ ANI as well.
I may have to use "Location" but I think GK will send out a call reject to CCM.

Have you confirm that your method works. If you got this working,
Can you send me GK end and GK gw-prefix output for this?

Thanks...

/Jin Jung...

kapil atrish wrote:
> You can try this:
> 
> Create two trunks between CCM and GK having only one CCM in each trunk i.e 
> one with Pub and another one with Sub. Create two set of regions (codec 
> G729), two locations (24kbps to allow only single call over the trunk), and 
> two DPs. Bind all this with respective trunks.
> 
> On the GK use gw-priority as regular, primary Sub and secondary Pub.
> 
> For HQ to Site C calling: Create two RGs having Sub and HQGW. Create RL 
> having both these RGs. Create a RP for Site C>>>Point to this RL. Now if any 
> call is already active over this H.323 Trunk any subsequent call from HQ side 
> will fall back (Location on the GK trunk will reject this call). You might 
> have to turn on the CCM Service parameters (Continue routing on unallocated 
> number).
> 
> 
> For Site C to HQ Calling: Since 1 call is already active on Sub, any 
> subsequent call from Site C will not be allowed due to b/w limitation over 
> that trunk (Location). Next call should fall back to Pub trunk which is 
> having gw-priority 9.
> 
> 
> Thanks...
> 
> 
> --- On *Sun, 3/29/09, CCIE OSL //* wrote:
> 
> 
>     From: CCIE OSL 
>     Subject: Re: [OSL | CCIE_Voice] gatekeeper question
>     To: "ccie Me" 
>     Cc: ccie_voice@onlinestudylist.com
>     Date: Sunday, March 29, 2009, 9:52 AM
> 
>     I have some questions for you.
> 
>     1. Are you running on 1 tech-prefix or two with on the gatekeeper?
>     2. Does entire BR2 has to able to call HQ or just a single phone?
>     3. Does it required to only use 1#, are you allow to use other
>     prefixes?
> 
>     Your first requirement for HQ to BR2 is fairly easy,
>     However, second requirement, is bit confusing.
> 
>     I think in order to make that work, I will have to use Hop-off
>     prefix  and statically map a another prefix to PUB address.
>     But since the CAC requirement of single call, and If I were to use
>     bandwidth interzone, I almost need another zone just for PUB,
>     Which means I may have to use different CAC method. /? or
>     somehow allow calls to PUB work using hop-off prefix not affected
>     by GK CAC.???
> 
>     I have proctor lab coming up on Monday night, I may have to lab
>     this up.
> 
>     If you can provide answer to my questions, It may help me to get
>     this done.
> 
>     Thanks...
> 
>     /Jin Jung...
> 
> 
> 
>     ccie Me wrote:
>     > Gents,
>     >
>     > i'm working on this case on gatekeeper:
>     >
>     > i need to only allow ONE active call that should be going
>     through SUB
>     > now for  any other new call
>     > if:
>     >
>     > -  it is from HQ to SiteC via GK it will be rejected by GK and
>     rerouted to pstn via 6608 on HQ
>     >
>     > -  if it is from SitC  to HQ via GK it will go through PUB
>     instead of SUB
>     >
>     > i tried to play with regions and CAC on gatekepper and CCM. but
>     i don't think that will lead to solve this case,
>     >
>     > does any body have idea about this
>     >
>     > thanks
>     >
> 
> 




  

Re: [OSL | CCIE_Voice] gatekeeper question

[kapil atrish]

You are absolutely righttwo ccm groups required having one ccm in each.

--- On Sun, 3/29/09, anil batra  wrote:

From: anil batra 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "ccie Me" , "CCIE OSL" , "kapil 
atrish" 
Cc: ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 11:06 AM

Do you mean we will be creating two CCM groups, two DP and then assign DP -Pub 
to one GK trunk and DP-Sub to another GK trunk...if you assgign a DP with CCM 
group having two CCM's it will be registering the trunk - gk-tunk_1 and 
gk-trunk_2 bu defualt ...

--- On Sun, 3/29/09, kapil atrish  wrote:

From: kapil atrish 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "ccie Me" , "CCIE OSL" 
Cc: ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 10:55 AM

You can try this:

Create two trunks between CCM and GK having only one CCM in each trunk i.e one 
with Pub and another one with Sub. Create two set of regions (codec G729), two 
locations (24kbps to allow only single call over the trunk), and two DPs. Bind 
all this with respective trunks.

On the GK use gw-priority as regular, primary Sub and secondary Pub. 

For HQ to Site C calling: Create two RGs having Sub and HQGW. Create RL having 
both these RGs. Create a RP for Site C>>>Point to this RL. Now if any call is 
already active over this H.323 Trunk any subsequent call from HQ side will fall 
back (Location on the GK trunk will reject this call). You might have to turn 
on the CCM Service parameters (Continue routing on unallocated number).


For Site
 C to HQ Calling: Since 1 call is already active on Sub, any subsequent call
 from Site C will not be allowed due to b/w limitation over that trunk 
(Location). Next call should fall back to Pub trunk which is having gw-priority 
9.


Thanks...


--- On Sun, 3/29/09, CCIE OSL  wrote:

From: CCIE OSL 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "ccie Me" 
Cc: ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 9:52 AM

I have some questions for you.

1. Are you running on 1 tech-prefix or two with on the gatekeeper?
2. Does entire BR2 has to able to call HQ or just a single phone?
3. Does it required to only use 1#, are you allow to use other prefixes?

Your first requirement for HQ to BR2 is fairly easy,
However, second requirement, is bit
 confusing.

I think in order to make that work, I will have to use Hop-off prefix  and 
statically map a another prefix to PUB address.
But since the CAC requirement of single call, and If I were to use bandwidth 
interzone, I almost need another zone just for PUB,
Which means I may have to use different CAC method. /? or somehow allow 
calls to PUB work using hop-off prefix not affected by GK CAC.???

I have proctor lab coming up on Monday night, I may have to lab this up.

If you can provide answer to my questions, It may help me to get this done.

Thanks...

/Jin Jung...



ccie Me wrote:
> Gents,
> 
> i'm working on this case on gatekeeper:
> 
> i need to only allow ONE active call that should be going through SUB
> now for  any other new call
> if:
> 
> -  it is from HQ to SiteC via GK it will be rejected by GK and rerouted to 
> pstn
 via 6608 on HQ
> 
> -  if it is from SitC  to HQ via GK it will go through PUB instead of SUB
> 
> i tried to play with regions and CAC on gatekepper and CCM. but i don't think 
> that will lead to solve this case,
> 
> does any body have idea about this
> 
> thanks
> 






  


  


  

Re: [OSL | CCIE_Voice] gatekeeper question

[kapil atrish]

You can try this:

Create two trunks between CCM and GK having only one CCM in each trunk i.e one 
with Pub and another one with Sub. Create two set of regions (codec G729), two 
locations (24kbps to allow only single call over the trunk), and two DPs. Bind 
all this with respective trunks.

On the GK use gw-priority as regular, primary Sub and secondary Pub. 

For HQ to Site C calling: Create two RGs having Sub and HQGW. Create RL having 
both these RGs. Create a RP for Site C>>>Point to this RL. Now if any call is 
already active over this H.323 Trunk any subsequent call from HQ side will fall 
back (Location on the GK trunk will reject this call). You might have to turn 
on the CCM Service parameters (Continue routing on unallocated number).


For Site C to HQ Calling: Since 1 call is already active on Sub, any subsequent 
call from Site C will not be allowed due to b/w limitation over that trunk 
(Location). Next call should fall back to Pub trunk which is having gw-priority 
9.


Thanks...


--- On Sun, 3/29/09, CCIE OSL  wrote:

From: CCIE OSL 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "ccie Me" 
Cc: ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 9:52 AM

I have some questions for you.

1. Are you running on 1 tech-prefix or two with on the gatekeeper?
2. Does entire BR2 has to able to call HQ or just a single phone?
3. Does it required to only use 1#, are you allow to use other prefixes?

Your first requirement for HQ to BR2 is fairly easy,
However, second requirement, is bit confusing.

I think in order to make that work, I will have to use Hop-off prefix  and 
statically map a another prefix to PUB address.
But since the CAC requirement of single call, and If I were to use bandwidth 
interzone, I almost need another zone just for PUB,
Which means I may have to use different CAC method. /? or somehow allow 
calls to PUB work using hop-off prefix not affected by GK CAC.???

I have proctor lab coming up on Monday night, I may have to lab this up.

If you can provide answer to my questions, It may help me to get this done.

Thanks...

/Jin Jung...



ccie Me wrote:
> Gents,
> 
> i'm working on this case on gatekeeper:
> 
> i need to only allow ONE active call that should be going through SUB
> now for  any other new call
> if:
> 
> -  it is from HQ to SiteC via GK it will be rejected by GK and rerouted to 
> pstn via 6608 on HQ
> 
> -  if it is from SitC  to HQ via GK it will go through PUB instead of SUB
> 
> i tried to play with regions and CAC on gatekepper and CCM. but i don't think 
> that will lead to solve this case,
> 
> does any body have idea about this
> 
> thanks
> 




  

Re: [OSL | CCIE_Voice] 6500 SERVER ACL

[kapil atrish]

following statement should also be added for H245 traffic:

set qos acl ip POD15_SERVER dscp 24 tcp any any ran 11000 65535

--- On Sun, 3/22/09, Christian Hennrich  
wrote:

From: Christian Hennrich 
Subject: Re: [OSL | CCIE_Voice] 6500 SERVER ACL
To: "Chris Parker" 
Cc: "OSL Group" 
Date: Sunday, March 22, 2009, 2:13 AM

Hi Chris,

I would mark the RTP also with the ACL:

set qos acl ip POD15_SERVER dscp 46 udp any range 16384 32767 any
set qos acl ip POD15_SERVER dscp 46 udp any any range 16384 32767

I think, you will not see a question, where you should mark and trust at the 
same time. And as far as I have tested,  does marking only work, when I had set 
the ports to untrusted.

HTH

Chris Parker schrieb:
> Here is my basic 6500 ACL for marking signaling to CS3:
> 
> set qos acl ip POD15_SERVER dscp 24 tcp    any eq 2000 any
> set qos acl ip POD15_SERVER dscp 24 udp    any eq 2427 any
> set qos acl ip POD15_SERVER dscp 24 tcp    any eq 2428 any
> set qos acl ip POD15_SERVER dscp 24 udp    any any eq 5060
> set qos acl ip POD15_SERVER dscp 24 tcp    any any eq 5060
> set qos acl ip POD15_SERVER dscp 24 udp    any ran 1718 1720 any
> set qos acl ip POD15_SERVER dscp 24 tcp    any ran 1718 1720 any
> set qos acl ip POD15_SERVER dscp 24 udp    any any ran 1718 1720
> set qos acl ip POD15_SERVER dscp 24 tcp    any any ran 1718 1720
> 
> Should I go ahead and  add 
> 
> set qos acl ip POD15_SERVER trust-dscp ip any any
> 
> to the end of the ACL so that we trust the DSCP marking of any media (MOH, 
> announcements, MTP) that originates from the UCM / Unity servers?
> 
> Chris
> 
> __
> This email has been scanned by the MessageLabs Email Security System.
> For more information please visit http://www.messagelabs.com/email 
> __
> 
> __
> This email has been scanned by the MessageLabs Email Security System.
> For more information please visit http://www.messagelabs.com/email 
> __



  

Re: [OSL | CCIE_Voice] Unity - "Wait while I transfer" option not available

[kapil atrish]

OK. Strangely I've see on all unity servers in PL labs the check box is 
available however Cisco documentation says its not available before Unity 4.2.

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_qanda_item09186a0080093c13.shtml



I've found a workaround to disable it by replacing the prompt with a blank one.

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=Unified%20Communications%20Applications&topicID=.ee835d2&fromOutline=&CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.2cbee5b8

--- On Tue, 3/24/09, Cristobal Priego  wrote:

From: Cristobal Priego 
Subject: Re: [OSL | CCIE_Voice] Unity - "Wait while I transfer" option not 
available
To: "kapil atrish" 
Cc: "ccie_voice@onlinestudylist.com" 
Date: Tuesday, March 24, 2009, 7:32 PM

What is the unity version?

On Mar 24, 2009, at 6:59 AM, kapil atrish  wrote:

Hi List,

Inside Unity Call Handler or Subscriber>>Call Transfer options you have the 
"Checkbox" to enable/disable the prompt "Wait while I transfer your call". I 
noted in a unity system this option is not there at all.

Has anybody seen this? Does anyone know if there is a way in Unity to get the 
option appear or how to make it disappear from Unity GUI interface?









  


  

[OSL | CCIE_Voice] Unity - "Wait while I transfer" option not available

[kapil atrish]

Hi List,

Inside Unity Call Handler or Subscriber>>Call Transfer options you have the 
"Checkbox" to enable/disable the prompt "Wait while I transfer your call". I 
noted in a unity system this option is not there at all.

Has anybody seen this? Does anyone know if there is a way in Unity to get the 
option appear or how to make it disappear from Unity GUI interface?









  

Re: [OSL | CCIE_Voice] Policer on Cat6k - Aggregate or Microflow

[kapil atrish]

Hi list,

Has anyone any thoughts on this?

--- On Fri, 3/13/09, kapil atrish  wrote:

From: kapil atrish 
Subject: Policer on Cat6k - Aggregate or Microflow
To: ccie_voice@onlinestudylist.com
Date: Friday, March 13, 2009, 1:46 AM

HI,

I am looking for a clarification on the policer to be used on Cat 6k.

Question says limit sccp traffic from phones to 32k, I've see few posts where 
an aggregate policer has been configured and implemented on voice vlan.

Shouldn't there be a microflow policer since I want to limit sccp traffic to 
32k per phone and not in total?

And if it is assumed that we need to limit cumulative sccp traffic to 32k, I've 
seen when same section asks to limit sccp traffic at BR1 and BR2 also, policing 
has been implemented inside policy-map which is an individual policer itself. 
There should've been a aggregate policer created and called inside policy-map??





  


  

[OSL | CCIE_Voice] Policer on Cat6k - Aggregate or Microflow

[kapil atrish]

HI,

I am looking for a clarification on the policer to be used on Cat 6k.

Question says limit sccp traffic from phones to 32k, I've see few posts where 
an aggregate policer has been configured and implemented on voice vlan.

Shouldn't there be a microflow policer since I want to limit sccp traffic to 
32k per phone and not in total?

And if it is assumed that we need to limit cumulative sccp traffic to 32k, I've 
seen when same section asks to limit sccp traffic at BR1 and BR2 also, policing 
has been implemented inside policy-map which is an individual policer itself. 
There should've been a aggregate policer created and called inside policy-map??





  

Re: [OSL | CCIE_Voice] How to only allow one international or LD call?

[kapil atrish]

Put max-conn under dial-peer.

--- On Fri, 3/13/09, CCIE OSL  wrote:

From: CCIE OSL 
Subject: [OSL | CCIE_Voice] How to only allow one international or LD call?
To: "OSL Group" 
Date: Friday, March 13, 2009, 12:56 AM



Question
How do you only allow one international or LD all any givin time?
This is CAC question or COR,, or both??





  

Re: [OSL | CCIE_Voice] how to make CME phone to have same number on button 1 and 2?

[kapil atrish]

You are not ref to the label, correct? 

I know of following two ways:

1) You can simply apply the same ephone-dn to button 1 and 2 on same phone.

2) You can create multiple DNs with same number and apply to two different 
buttons of same ephone.

--- On Fri, 3/13/09, CCIE OSL  wrote:

From: CCIE OSL 
Subject: [OSL | CCIE_Voice] how to make CME phone to have same number on button 
1 and 2?
To: "OSL Group" 
Date: Friday, March 13, 2009, 12:10 AM

how to make CME phone to have same number on button 1 and 2?

Is there a way to make 3001 appear on both button 1 and 2?
I am not talking about description field.

So it will show up

6727653001  <--description
           3001  <-- button 1
           3001  <-- button 2

/Jin Jung...






  

[OSL | CCIE_Voice] CNAME with PSTN

[Kapil Atrish]

Hi,

I've a query regarding CNAME display when calling PSTN.

1) CCM Phone calls PSTN Phone
Pstn phone can see the CNAME of CCM Phone but CCM Phone can't see the name of 
called party.

2) PSTN Phone calls CCM Phone
CCM phone can see the CNAME of PSTN phone but PSTN Phone can't see the name of 
called party. 

I've "display-IE delivery" check-box checked on CCM and "isdn outgoing 
display-ie" on PSTN side. CLID gets displayed correctly but not the CNAME, is 
this the correct behaviour?





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Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module

[Kapil Atrish]

Yes, the policy-map works on Vlan interface and since its an individual policer 
it serves the purpose of limiting SCCP traffic per phone.

Thanks...
From: spoli...@hotmail.com
To: kapilatr...@hotmail.com; vma...@ipexpert.com; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] DSCP marking on NM-ESW module
Date: Tue, 10 Mar 2009 20:59:28 -0300








 

If you apply the service policy input to the Interface Vlan of the Br1 could 
answer the question?
 


From: kapilatr...@hotmail.com
To: vma...@ipexpert.com; ccie_voice@onlinestudylist.com
Date: Wed, 11 Mar 2009 01:24:01 +0530
Subject: Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module



I tried "match protocol" but no good. See below error:


Pod28-BR1-RTR(config-if)#service-policy input EF
%Error: FastEthernet1/0 Service Policy Configuration Failed.Only Match with 
access group is supported




Date: Mon, 9 Mar 2009 16:46:00 -0800
Subject: Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module
From: vma...@ipexpert.com
To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com

Can you use “match protocol skinny” and “match protocol rtp”?
-- 
Vik Malhi – CCIE #13890, CCSI #31584 
Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444 
Fax: +1.810.454.0130 
Mailto: vma...@ipexpert.com


Join our free online support and peer group communities: 
http://www.IPexpert.com/communities
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.








From: Kapil Atrish 
Date: Tue, 10 Mar 2009 02:23:28 +0530
To: OSL Group 
Subject: [OSL | CCIE_Voice] DSCP marking on NM-ESW module

Any other workaround to mark all SCCP and RTP packets to respective DSCP values 
on this module? I could successfully do it on Cat 3550 with access-list + port 
range though.



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Re: [OSL | CCIE_Voice] Any Solution for: G729 call failed when CCM -> CME BACD via Gatekeeper??

[kapil atrish]

You need Xcoder. BACD supports only G711 and since you are doing G729, you'll 
face this problem. To verify, make end to end G711 and you should be able to 
reach BACD successfully.

--- On Wed, 3/11/09, Jiahong - tobeccie Fang  wrote:

From: Jiahong - tobeccie Fang 
Subject: [OSL | CCIE_Voice] Any Solution for: G729 call failed when CCM -> CME 
BACD via Gatekeeper??
To: ccie_voice@onlinestudylist.com
Date: Wednesday, March 11, 2009, 12:09 AM




#yiv237988227 .hmmessage P
{
margin:0px;padding:0px;}
#yiv237988227 {
font-size:10pt;font-family:Verdana;}

CCM and CME all register to GK, call between CCM and CME is G729. Call ip 
phones between each site work well.

Only problem is: when call from CCM to CME BACD say: DN:8800, it failed. When 
debug script in CME, can see 'aa' script did play prompt.

VoIP_CME#
Jan 29 22:09:33.670: //214//TCL :/tcl_PutsObjCmd: 
proc init_perCallvars
Jan 29 22:09:33.670: 
Jan 29 22:09:33.674: //214//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome 
Prompt and options menu ++

For gk trunk in CCM site, turn on or off 'wait for far-end h245 terminial 
capability set' - same result.

I know this is known issue. 

Does anyone have quick solution for it?

Thanks

James


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Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve

[kapil atrish]

I just checked on one of the POD. This parameter it there.



Allow Peer to Preserve H.323 Calls : This parameter determines whether Cisco 
CallManager allows the peer H.323 endpoint to try to preserve active H.323 
calls by not sending a Release Complete message to the peer H.323 endpoint when 
the TCP connection is lost between Cisco CallManager and the far endpoint, such 
as a Skinny Client Control Protocol [SCCP] IP phone. When connectivity is lost, 
the H.225 and H.245 TCP connection is torn down but Cisco CallManager does not 
send the Release Complete message so that the peer H.323 endpoint can attempt 
to preserve the media path. Valid values specify True (do not send the Release 
Complete message to the peer H.323 endpoint when the far endpoint’s 
connectivity with Cisco CallManager is lost) or False (send the Release 
Complete message to the peer H.323 endpoint when the far endpoint’s 
connectivity with Cisco CallManager is lost, which will resulting the call 
being terminated). 
This is a required field.
Default: false.
 
H225 Block Setup Destination : This parameter determines whether  


--- On Wed, 3/11/09, Christopher Clouse  wrote:

From: Christopher Clouse 
Subject: Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve
To: "Yung Hung" , mmailb...@yahoo.com, "Cliff 
McGlamry" , "Scott ODonnell" 
Cc: ccie_voice@onlinestudylist.com
Date: Wednesday, March 11, 2009, 1:24 AM



 
 

#yiv1653033877 v\:* {}
#yiv1653033877 o\:* {}
#yiv1653033877 w\:* {}
#yiv1653033877 .shape {}





Configuration of H.323 VoIP call preservation 
enhancements for WAN link failures involves configuring the call 
preserve command. If you are using Cisco Unified 
Communications Manager, you must enable the Allow Peer to Preserve H.323 Calls 
parameter from the Service Parameters window.
 
I pulled this out of the CM6 SRND, but I do know that it 
exists in CM4.  I'm running sr8a in my lab right now.
 
~Chris




From: Yung Hung 
Sent: Tuesday, March 10, 2009 2:39 PM
To: Christopher Clouse ; mmailb...@yahoo.com ; Cliff McGlamry ; 
Scott ODonnell 
Cc: ccie_voice@onlinestudylist.com 

Subject: RE: [OSL | CCIE_Voice] h.323 gateway config & call 
preserve



I’m looking for this 
parameter and I cannot find it. Which version are you running? 
   


From: Christopher Clouse 
[mailto:christopherc_56...@hotmail.com] 
Sent: Tuesday, March 10, 2009 
12:41 PM
To: Yung Hung; mmailb...@yahoo.com; Cliff McGlamry; Scott 
ODonnell
Cc: ccie_voice@onlinestudylist.com
Subject: 
Re: [OSL | CCIE_Voice] h.323 gateway config & call 
preserve 
   

I believe the service 
parameter that we're looking for is 'Allow Peer to Preserve Call'.  
 

  

~Chris 


   


From: Yung 
Hung  

Sent: Tuesday, March 10, 
2009 12:04 PM 

To: mmailb...@yahoo.com ; Cliff McGlamry ; 
Scott 
ODonnell  

Cc: ccie_voice@onlinestudylist.com 
 

Subject: Re: [OSL | 
CCIE_Voice] h.323 gateway config & call 
preserve 

   
What do you have 
setup on your service parameter in CCM “Allow TCP KeepAlives For 
H323*” 
   

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Balamurugan 
Singaram
Sent: Tuesday, March 10, 2009 1:20 AM
To: Cliff 
McGlamry; Scott ODonnell
Cc: ccie_voice@onlinestudylist.com
Subject: 
Re: [OSL | CCIE_Voice] h.323 gateway config & call 
preserve 
   

  
  

  
   IPExpert V3 Mock lab workbook, related to 
  h.323 gateway I am able to see below command for call 
  preservation: 
  
    
  
  voice service voip
h323
no h225 timeout 
  keepalive 
  
  even after update this command, my call are not 
  preserved when the both ccm SUB and PUB goes down, I am missing 
  something, could please some one light me. 
  
    
  
  Thanks, 
  
  
--- On Tue, 10/3/09, Scott ODonnell 
   wrote: 
  
From: Scott ODonnell 

Subject: Re: [OSL | CCIE_Voice] 
h.323 gateway config & call preserve
To: "Cliff McGlamry" 

Cc: "Kumar, Narinder" 
, mmailb...@yahoo.com, 
ccie_voice@onlinestudylist.com
Date: Tuesday, 10 March, 2009, 12:09 
PM 

I think you 
need the "no h225 timeout keepalive" configured under the "voice 
service 
voip / h323". 
This will 
allow the call to stay up after the GW stops receiving keepalives from 
the CM. 
   

http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080558061.html
 
H.323 
Gateways and SRST 
On H.323 gateways, when the 
WAN link fails, active calls from Cisco IP phones to the PSTN are not 
maintained by default. Call preservation may work with the no 
h225 timeout keepalive command, but call preservation using 
the no h225 timeout keepalive command is not officially 
supported by Cisco Technical Support. 
Under default configuration, 
th

Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve

[kapil atrish]

Its a CCM service parameter. I just checked it on one of the POD. Following is 
what it says:


Allow Peer to Preserve H.323 Calls : This parameter determines whether Cisco 
CallManager allows the peer H.323 endpoint to try to preserve active H.323 
calls by not sending a Release Complete message to the peer H.323 endpoint when 
the TCP connection is lost between Cisco CallManager and the far endpoint, such 
as a Skinny Client Control Protocol [SCCP] IP phone. When connectivity is lost, 
the H.225 and H.245 TCP connection is torn down but Cisco CallManager does not 
send the Release Complete message so that the peer H.323 endpoint can attempt 
to preserve the media path. Valid values specify True (do not send the Release 
Complete message to the peer H.323 endpoint when the far endpoint’s 
connectivity with Cisco CallManager is lost) or False (send the Release 
Complete message to the peer H.323 endpoint when the far endpoint’s 
connectivity with Cisco CallManager is lost, which will resulting the call 
being terminated). 
This is a required field.
Default: false.
 
H225 Block Setup Destination : This parameter determines whether  


--- On Wed, 3/11/09, Yung Hung  wrote:

From: Yung Hung 
Subject: Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve
To: "Christopher Clouse" , 
"mmailb...@yahoo.com" , "Cliff McGlamry" 
, "Scott ODonnell" 
Cc: "ccie_voice@onlinestudylist.com" 
Date: Wednesday, March 11, 2009, 1:09 AM




 
 







I’m
looking for this parameter and I cannot find it. Which version are you running? 

   





From: Christopher Clouse
[mailto:christopherc_56...@hotmail.com] 

Sent: Tuesday, March 10, 2009 12:41 PM

To: Yung Hung; mmailb...@yahoo.com; Cliff McGlamry; Scott ODonnell

Cc: ccie_voice@onlinestudylist.com

Subject: Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve 





   



I
believe the service parameter that we're looking for is 'Allow Peer to Preserve
Call'.   





  





~Chris 







   







From: Yung
Hung  





Sent: Tuesday, March 10,
2009 12:04 PM 





To: mmailb...@yahoo.com ; Cliff McGlamry
; Scott
ODonnell  





Cc: ccie_voice@onlinestudylist.com  





Subject: Re: [OSL |
CCIE_Voice] h.323 gateway config & call preserve 









   



What
do you have setup on your service parameter in CCM “Allow TCP KeepAlives For
H323*” 

   



From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Balamurugan
Singaram

Sent: Tuesday, March 10, 2009 1:20 AM

To: Cliff McGlamry; Scott ODonnell

Cc: ccie_voice@onlinestudylist.com

Subject: Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve 



   


 
  
  
   IPExpert V3 Mock lab workbook, related to h.323
  gateway I am able to see below command for call preservation: 
  
  
    
  
  
  voice service voip

  h323

  no h225 timeout keepalive 
  
  
  even after update this command, my call are not preserved
  when the both ccm SUB and PUB goes down, I am missing something, could
  please some one light me. 
  
  
    
  
  
  Thanks, 
  
  
  

  --- On Tue, 10/3/09, Scott ODonnell 
  wrote: 
  
  
  From: Scott ODonnell
  

  Subject: Re: [OSL | CCIE_Voice] h.323 gateway config & call preserve

  To: "Cliff McGlamry" 

  Cc: "Kumar, Narinder" ,
  mmailb...@yahoo.com, ccie_voice@onlinestudylist.com

  Date: Tuesday, 10 March, 2009, 12:09 PM 
  
  I think you need the "no h225 timeout
  keepalive" configured under the "voice service voip / h323". 
  This will allow the call to stay up after the GW stops
  receiving keepalives from the CM. 
     
  
http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080558061.html
 
  H.323 Gateways and SRST 
  On
  H.323 gateways, when the WAN link fails, active calls from Cisco IP phones to
  the PSTN are not maintained by default. Call preservation may work with
  the no h225 timeout keepalive command, but call preservation
  using the no h225 timeout keepalive command is not
  officially supported by Cisco Technical Support. 
  
  Under
  default configuration, the H.323 gateway maintains a keepalive signal with
  Cisco CallManager and terminates H.323-to-PSTN calls if the keepalive signal
  fails, for example if the WAN link fails. To disable this behavior and help
  preserve existing calls from local IP phones, you can use the no h225
  timeout keepalivecommand. Disabling the keepalive mechanism only affects
  calls that will be torn down as a result of the loss of the H.225 keepalive
  signal. For information regarding disconnecting a call when an inactive
  condition is detected. see the Media Inactive Call Detection document. 
  
     
  
     
  
  
     
  
  
     
  
  
     
  
  
  On Tue, Mar 10, 2009 at 2:30 AM, Cliff McGlamry 
  wrote: 
  
  
  I
  believe there is also a service parameter command to allow call
  preservation in the event CCM goes down under h.323 in the CCM Service
  parameters.  Seems like I've seen it there 
  
  
  
  
  
 

Re: [OSL | CCIE_Voice] ATA186 - not able to change anything

[Kapil Atrish]

OK. I faced this again today, but luckily cisco worked. I put in "old password" 
as cisco and new password fields as blank and was able to access the ATA.



From: ghaus...@cox.net
To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] ATA186 -  not able to change anything
Date: Mon, 9 Mar 2009 23:52:06 -0700



















I had this exact problem happen to me on Sunday.
 There was no way to get around this error.  I thought it was just my pod 8…

 









From:
ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil Atrish

Sent: Saturday, March 07, 2009
2:38 AM

To: ccie_voice@onlinestudylist.com

Subject: [OSL | CCIE_Voice] ATA186
- not able to change anything



 

Hi list,



I've an ATA on POD11 which has port 1 enabled by default. I want to enable Port
2 as well. Whenever I click on SCCP parameters it opens the Change UIP password
page. I tried other options under "Change Parameters" but it still
asks for UIP password change. 



It seems ATA web-administration has been disabled. Does anybody how to overcome
this? I tried passwords cisco, 12345 with no luck.



thanks







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Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module

[Kapil Atrish]

I tried "match protocol" but no good. See below error:


Pod28-BR1-RTR(config-if)#service-policy input EF
%Error: FastEthernet1/0 Service Policy Configuration Failed.Only Match with 
access group is supported


Date: Mon, 9 Mar 2009 16:46:00 -0800
Subject: Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module
From: vma...@ipexpert.com
To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com





Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module


Can you use “match protocol skinny” and “match protocol rtp”?

-- 

Vik Malhi – CCIE #13890, CCSI #31584 

Senior Technical Instructor - IPexpert, Inc.



Telephone: +1.810.326.1444 

Fax: +1.810.454.0130 

Mailto: vma...@ipexpert.com





Join our free online support and peer group communities: 

http://www.IPexpert.com/communities

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.

















From: Kapil Atrish 

Date: Tue, 10 Mar 2009 02:23:28 +0530

To: OSL Group 

Subject: [OSL | CCIE_Voice] DSCP marking on NM-ESW module



Any other workaround to mark all SCCP and RTP packets to respective DSCP values 
on this module? I could successfully do it on Cat 3550 with access-list + port 
range though.


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[OSL | CCIE_Voice] DSCP marking on NM-ESW module

[Kapil Atrish]

Hi List,

I tried to mark SCCP/RTP traffic on ESW module, but the policy got rejected 
with a message the it doesn't support range keyword in access-list.

Any other workaround to mark all SCCP and RTP packets to respective DSCP values 
on this module? I could successfully do it on Cat 3550 with access-list + port 
range though.

Thanks

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The new Windows Live Messenger. You don’t want to miss this.
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[OSL | CCIE_Voice] ATA186 - not able to change anything

[Kapil Atrish]

Hi list,

I've an ATA on POD11 which has port 1 enabled by default. I want to enable Port 
2 as well. Whenever I click on SCCP parameters it opens the Change UIP password 
page. I tried other options under "Change Parameters" but it still asks for UIP 
password change. 

It seems ATA web-administration has been disabled. Does anybody how to overcome 
this? I tried passwords cisco, 12345 with no luck.

thanks

_
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Re: [OSL | CCIE_Voice] Incoming TEHO for HQ from BR2 (Spain) not as advertised.

[kapil atrish]

Instead of rolling it over to HQ Unrestricted, I created a CSS having access to 
only 1x  which internally points to HQ GW. The TP can only be accessed 
from GK and am doing PreDot 1# at TP level. It always works for me.

Another TP which looks for 1#.[1-2]xxx. I do a PreDot and roll it over to CSS 
internal.

--- On Tue, 3/3/09, Cliff McGlamry  wrote:

From: Cliff McGlamry 
Subject: [OSL | CCIE_Voice] Incoming TEHO for HQ from BR2 (Spain) not as 
advertised.
To: ccie_voice@onlinestudylist.com
Date: Tuesday, March 3, 2009, 12:46 AM



 
 

I've got TEHO set up and working coming in from my 
BR2 (Spain) site.  The number arrives at CallManager as a string 
1#1212224
 
Based on the proctor guide, I set up a translation 
pattern for 1#1XX and made sure I created it last.  However, it 
wouldn't work.  
 
I had the translation pattern set up as 
1#.1XX and set it to strip predot and prefix a 9, and then roll it into 
my Unrestricted HQ CSS.  At that point, it should have been able to see the 
route patterns set up on my various route lists.  Unfortunately, it 
wouldn't work.  I attempted multiple permutations of this, none of them 
successful.
 
I ended up tightening up the translation pattern 
for internal numbers to be 1#[12]0XX and then created a route pattern for 
1#.1[2-9]X stripping pre dot so that the two did not overlap.  
While this setup did not work in a translation pattern, it DID work in a route 
pattern.  
 
I'm trying to figure out why the setup as a 
translation pattern would not work.  Has anyone else run into this?  
I'm less concerned with getting an exact match of the provided solution as I am 
understanding what would cause my setup of it (attempting to duplicate it) 
would 
fail.  Mark says that the functionality is what's required, and I was able 
to do that.  But why wouldn't my translation pattern work?  I even 
deleted it and added it back again...and got exactly the same 
results.
 
Cliff
 


  

Re: [OSL | CCIE_Voice] 6608 gw

[kapil atrish]

In the route pattern or route-group level, don't use "Use external phone number 
mask" and put area-code (3 digits) as Prefix to calling number.

--- On Mon, 3/2/09, hasan khalife  wrote:

From: hasan khalife 
Subject: [OSL | CCIE_Voice] 6608 gw
To: ccie_voice@onlinestudylist.com
Date: Monday, March 2, 2009, 10:55 PM




#yiv1601154501 .hmmessage P
{
margin:0px;padding:0px;}
#yiv1601154501 {
font-size:10pt;font-family:Verdana;}

1-when calling name should not be displaY juSt unckeck the IE BOX ?

 

 

2-   7  DIGIT CALLING NUMBER SHOULD BE DISPLAY ON THE PSTN ?where is must 
specify ?

 

 

thx

 

 

 

 

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



  

Re: [OSL | CCIE_Voice] ATA registration in SRST Mode

[Kapil Atrish]

After switching over to max-dn dual-line I realised ATA actually registers with 
two channels now. Earlier it showed two buttons both associated to Channel 1. 
If someone can comment on the usage of two buttons with single channe or if 
there is something wrongl, that'll be great.

See below output when max-dn with dual-line configured


/call-mana
filtering...
call-manager-fallback
 max-conferences 4 gain -6
 ip source-address 142.103.65.254 port 2000 strict-match
 max-ephones 6
 max-dn 15 dual-line
 voicemail 14082032600
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 142.103.65.254 142.33.65.1
 cor incoming int 2 3002
!

BR1(config-cm-fallback)#do sh ephone   

ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.54 51294 7970   keepalive 38 max_line 8
button 1: dn 1  number 3002  CM Fallback CH1   IDLE CH2   IDLE 


ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.53 52527 7970   keepalive 39 max_line 8
button 1: dn 2  number 3001  CM Fallback CH1   IDLE CH2   IDLE 


ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 
3 and Server in ver 3
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 7003 ATA Phone  keepalive 23 max_line 1
button 1: dn 3  number 3006  CM Fallback CH1   IDLE CH2   IDLE 


ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 
3 and Server in ver 3
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 6379 ATA Phone  keepalive 13 max_line 1
button 1: dn 4  number 3007  CM Fallback CH1   IDLE CH2   IDLE 



Thanks...


From: kapilatr...@hotmail.com
To: ccie_voice@onlinestudylist.com
Subject: ATA registration in SRST Mode
Date: Mon, 2 Mar 2009 14:49:24 +0530








Hi,

I observed both ports on ATA register with dual-line even though max-dn 
dual-line is not configured. Is this expected behavior?


I couldn't find much information on cisco reg this. See following capture


interface FastEthernet0/0
/call-mana
filtering...
call-manager-fallback
 max-conferences 4 gain -6
 ip source-address 142.103.65.254 port 2000 strict-match
 max-ephones 6
 max-dn 12
 voicemail 14082032600
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 142.103.65.254 142.33.65.1
 cor incoming int 2 3002
!

   
BR1(config)#do sh ephone

ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.54 53094 7970   keepalive 130 max_line 8
button 1: dn 1  number 3002  CM Fallback CH1   IDLE 


ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.53 52572 7970   keepalive 129 max_line 8
button 1: dn 2  number 3001  CM Fallback CH1   ALERTING 
Active Call on DN 2 chan 1 :3001 0.0.0.0 0 to 0.0.0.0 2000 via 142.103.65.53
G711Ulaw64k  160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn 3 


ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 ringing:1 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 7642 ATA Phone  keepalive 121 max_line 2 dual-line
button 1: dn 3  number 3006  CM Fallback CH1   RINGING  
button 2: dn 4  number 3006  CM Fallback CH1   IDLE 
call ringing on line 1


ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 6924 ATA Phone  keepalive 111 max_line 2 dual-line
button 1: dn 5  number 3007  CM Fallback CH1   IDLE 
button 2: dn 6  number 3007  CM Fallback CH1   IDLE 







BR1(config)#do sh ephone

ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.54 53094 7970   keepalive 130 max_line 8
button 1: dn 1  number 3002  CM Fallback CH1   IDLE 


ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.53 52572 7970   keepalive 130 max_line 8
button 1: dn 2  number 3001  CM Fallback CH1   CONNECTED
Active Call on DN 2 chan 1 :3001 142.103.65.53 20468 to 142.103.65.55 16385 via 
142.103.65.53
G711Ulaw64k  160 bytes no vad
Tx Pkts 494 bytes 84968 Rx Pkts 492 bytes 8462

[OSL | CCIE_Voice] ATA registration in SRST Mode

[Kapil Atrish]

Hi,

I observed both ports on ATA register with dual-line even though max-dn 
dual-line is not configured. Is this expected behavior?


I couldn't find much information on cisco reg this. See following capture


interface FastEthernet0/0
/call-mana
filtering...
call-manager-fallback
 max-conferences 4 gain -6
 ip source-address 142.103.65.254 port 2000 strict-match
 max-ephones 6
 max-dn 12
 voicemail 14082032600
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 142.103.65.254 142.33.65.1
 cor incoming int 2 3002
!

   
BR1(config)#do sh ephone

ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.54 53094 7970   keepalive 130 max_line 8
button 1: dn 1  number 3002  CM Fallback CH1   IDLE 


ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.53 52572 7970   keepalive 129 max_line 8
button 1: dn 2  number 3001  CM Fallback CH1   ALERTING 
Active Call on DN 2 chan 1 :3001 0.0.0.0 0 to 0.0.0.0 2000 via 142.103.65.53
G711Ulaw64k  160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn 3 


ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 ringing:1 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 7642 ATA Phone  keepalive 121 max_line 2 dual-line
button 1: dn 3  number 3006  CM Fallback CH1   RINGING  
button 2: dn 4  number 3006  CM Fallback CH1   IDLE 
call ringing on line 1


ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 6924 ATA Phone  keepalive 111 max_line 2 dual-line
button 1: dn 5  number 3007  CM Fallback CH1   IDLE 
button 2: dn 6  number 3007  CM Fallback CH1   IDLE 







BR1(config)#do sh ephone

ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.54 53094 7970   keepalive 130 max_line 8
button 1: dn 1  number 3002  CM Fallback CH1   IDLE 


ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.53 52572 7970   keepalive 130 max_line 8
button 1: dn 2  number 3001  CM Fallback CH1   CONNECTED
Active Call on DN 2 chan 1 :3001 142.103.65.53 20468 to 142.103.65.55 16385 via 
142.103.65.53
G711Ulaw64k  160 bytes no vad
Tx Pkts 494 bytes 84968 Rx Pkts 492 bytes 84624 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn 3 


ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:1 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 7642 ATA Phone  keepalive 121 max_line 2 dual-line
button 1: dn 3  number 3006  CM Fallback CH1   CONNECTED
button 2: dn 4  number 3006  CM Fallback CH1   IDLE 
Active Call on DN 3 chan 1 :3006 142.103.65.55 16385 to 142.103.65.53 20468 via 
142.103.65.55
G711Ulaw64k  160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn 2 calledDn -1 


ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 6924 ATA Phone  keepalive 111 max_line 2 dual-line
button 1: dn 5  number 3007  CM Fallback CH1   IDLE 
button 2: dn 6  number 3007  CM Fallback CH1   IDLE 









BR1(config)#do sh ephone

ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.54 53094 7970   keepalive 131 max_line 8
button 1: dn 1  number 3002  CM Fallback CH1   IDLE 


ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 
15 and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.53 52572 7970   keepalive 131 max_line 8
button 1: dn 2  number 3001  CM Fallback CH1   IDLE 


ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:142.103.65.55 7642 ATA Phone  keepalive 122 max_line 2 dual-line
button 1: dn 3  number 3006  CM Fallback CH1   IDLE 
button 2: dn 4  number 3006  CM Fallback CH1   IDLE 


ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 
0 and Server in ver 0
mediaActive:0 offhook:0 r

Re: [OSL | CCIE_Voice] BR1 - SRST???

[kapil atrish]

Put following on BR1 to trigger SRST:

ip route ccm pub ip /mask null 0
ip route ccm sub ip/mask null 0

--- On Sun, 3/1/09, Mike Brooks <2xcci...@gmail.com> wrote:

From: Mike Brooks <2xcci...@gmail.com>
Subject: Re: [OSL | CCIE_Voice] BR1 - SRST???
To: "Cliff McGlamry" 
Cc: ccie_voice@onlinestudylist.com
Date: Sunday, March 1, 2009, 6:30 AM

Hey Cliff,
 
There is a couple ways of doing it, but what I always do is just configure a 
call-manager group with just the SUB in it and put it int the BR1 device pool.  
Then assign all devices in the BR1 site to the BR1_DP (IP Blue as well).  Also, 
only configure the BR1 router to communicate with the SUB.  Then when you are 
ready to test SRST mode then just stop the CM Service on the SUB.

 
Of course, in the real lab you would just shut down the WAN link and test.
 
hth,
 
Mike Brooks
CCIE#16027 (R&S)


 
On Sun, Mar 1, 2009 at 8:52 AM, Cliff McGlamry  wrote:



How do you throw BR1 into SRST so that IP Blue will register to the BR1 router 
but still be able to go via PSTN to Unity?




  

[OSL | CCIE_Voice] IPIP Gatway or CME transcoding not working

[kapil atrish]

Hi list,

Actually, it all started when I tried to send a G.729 stream to CME with G.711 
at CCM side. I initially had Xcoder on CME but never got invoked. I tried IPIP 
GW but same result. Below is my scneario and results of the testing so far:


I've a trunk from CCM to GK with codec G.711 set. CME also registers to same GK 
with dial-peers (inbound/outbound) set for default G.729. I've configured IPIP 
Gateway on GK with transcoders and trying to transcode the calls locally. 

I've xcoder registered but don't see them getting invoked.

If I make CCM Trunk G.711  CME G.711 all works fine,

If I make CCM Trunk G.729  CME G.729 all works fine,

If I make CCM Trunk G.711 --- CME G.729, both ways call mature on G.729

If I make CCM Trunk G.729 --- CME G.711, I get fast busy both ways.
The call disconnect cause-value=47 on CME. It seems CME is trying to Xcode call 
locally which should've been done by GK. 

When I tried Xcoding at CME same result/same cause-code.

Attached snapshot of some relevant commands. Any input is highly appreciated.





  IPIP Gateway

!

voice-card 1
 dsp services dspfarm
!
ip cef
!
!!
voice service voip 
 allow-connections h323 to h323
!
!
interface FastEthernet0/0.103
 encapsulation dot1Q 103
 ip address 142.103.64.254 255.255.255.0
 ip helper-address 142.3.64.11
 ip helper-address 142.3.64.12
 h323-gateway voip interface
 h323-gateway voip id IPIP ipaddr 142.33.64.1 1719
 h323-gateway voip h323-id IPIP
!
! 
sccp local FastEthernet0/0.103
sccp  
sccp ccm 142.33.64.1 priority 1
sccp codec g729ar8 mask
sccp codec g729abr8 mask
! 
dspfarm transcoder maximum sessions 4
dspfarm connection interval 60
dspfarm   
! 
! 
gateway  
!
 
! 
! 
gatekeeper
 zone local IPIP cisco.com 142.33.64.1
 zone local ucm cisco.com
 zone local CME cisco.com invia IPIP outvia IPIP enable-intrazone
 zone prefix ucm 2* gw-priority 10 gk-trunk_2
 zone prefix ucm 3* gw-priority 10 gk-trunk_2
 zone prefix CME 4* gw-priority 10 gw
 gw-type-prefix 1#* default-technology
 no shutdown
! 
! 
telephony-service
 max-ephones 1
 max-dn 1 
 ip source-address 142.33.64.1 port 2000
 sdspfarm units 1
 sdspfarm transcode sessions 4
 sdspfarm tag 1 mtp001200d86500
 create cnf-files version-stamp Jan 01 2002 00:00:00
 max-conferences 4 gain -6
! 

HQ#sh sdspfarm sessions summary 

 max-mtps:1, max-streams:8, alloc-streams:8, act-streams:0
  ID   MTP  State  CallID confID Usage 
Codec/Duration
 = == === == = 
==
11 IDLE   -1  0   G711Ulaw64k 
/20ms
21 IDLE   -1  0   G711Ulaw64k 
/20ms
31 IDLE   -1  0   G711Ulaw64k 
/20ms
41 IDLE   -1  0   G711Ulaw64k 
/20ms
51 IDLE   -1  0   G711Ulaw64k 
/20ms
61 IDLE   -1  0   G711Ulaw64k 
/20ms
71 IDLE   -1  0   G711Ulaw64k 
/20ms
81 IDLE   -1  0   G711Ulaw64k 
/20ms
HQ#
HQ#
HQ#show sccp ?
  all  Display all SCCP global info
  connections  Display SCCP connections
  statistics   Display SCCP statistics
  |Output modifiers
  

HQ#show sccp conn
HQ#show sccp connections ?
  |  Output modifiers
  

HQ#show sccp connections 

Total number of active session(s) 0, and connection(s) 0

HQ#show sccp al
HQ#show sccp all 
SCCP Admin State: UP
Gateway IP Address: 142.103.64.254
Switchover Method: IMMEDIATE, Switchback Method: GUARD_TIMER
Switchback Guard Timer: 1200 sec, IP Precedence: 5
Max Supported MTP sessions: 0
User Masked Codec list: g729ar8 g729abr8 
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 142.33.64.1, Port Number: 2000
TCP Link Status: CONNECTED
Conferencing Oper State: DOWN - Cause Code: DSPFARM_DOWN
Active Call Manager: NONE
TCP Link Status: NOT_CONNECTED
Call Manager: 142.33.64.1, Port Number: 2000
Priority: 1, Version: 3.1 or Higher

SCCP Transcoding Application Statistics:
TCP packets rx 179, tx 185
Unsupported pkts rx 2, Unrecognized pkts rx 0
Register tx 2, successful 2, rejected 0, failed 0
KeepAlive tx 175, successful 175, failed 0
OpenReceiveChannel rx 0, successful 0, failed 0
CloseReceiveChannel rx 0, successful 0, failed 0
StartMediaTransmission rx 0, successful 0, failed 0
StopMediaTransmission rx 0, successful 0, failed 0
MediaStreamingFailure rx 0
Switchover 0, Switchback 0

SCCP Conferencing Application Statistics:
TCP packets rx 117, tx 121
Unsupported pkts rx 0, Unrecognized pkts rx 0
Register tx 1, successful 1, rejected 0, failed 0
KeepAlive tx 116, successful 116, failed 0
OpenReceiveChannel rx 0, successful 0, failed 0
CloseReceiveChannel rx 0, s

Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down

[Kapil Atrish]

Yeah but once Sub is offline, the Multicast Address configured at BR1 is not 
active. How do I point it to Pub dynamically.

I tried giving same Address to Pub/Sub and changing Base port 16384 and 16386 
but haven't worked since BR1 is still configured for Sub Multicast IP and port 
no.

Is there a wayout I can add two multicast entries in BR1?

From: narinder.ku...@uxcg.com.au
To: narinder.ku...@uxcg.com.au; kapilatr...@hotmail.com; basant.ya...@gmail.com
CC: ccie_voice@onlinestudylist.com
Date: Thu, 26 Feb 2009 00:06:45 +1100
Subject: RE: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down













Kapil correction. No need of route filers, You can have PUB Hop count to 1 in 
that case 239.1.1.1 will never reach Branch.
 


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com]
On Behalf Of Kumar, Narinder

Sent: Thursday, 26 February 2009 12:04 AM

To: Kapil Atrish; basant.ya...@gmail.com

Cc: ccie_voice@onlinestudylist.com

Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down


 
Kapil,
I haven’t tested this but . 

If Sub unicast  to BR1 than you can easily achieve this by spoofing the PUB 
multicast add and have PUB ur second choice for MOH for Branch  and have 
multicast
 hop of 1, and config  multicast MOH under call-manager-fall 
 
Assuming you are trying to achieve Multicast MOH end to end.

2 MOH Pub and Sub, in 2 different MGR both multicast enabled.
Say Sub is ur primary MOH with Base IP add 239.1.2.1
Pub with base IP add 239.1.1.1
 
Now you need multicast end to end

 
In the MRGL Sub MOH first and PUB MOH second.
 
Now in normal situation Branch PH’s are getting MOH from SUB.
 
In case Sub stop working. You can add some route filers to stop reaching the 
Pub base address 239.1.1.1 to the branch . The CCM will still tell the phones
 to listen to 239.1.1.1 and you can spoof it from flash.
 
I don’t know if this will work or not. I haven’t tested myself.
 
But if you have only ONE MOH and you want Branch router as backup, I am not 
sure if that can be achieved.
Cheers
Narinder
 
 
 


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com]
On Behalf Of Kapil Atrish

Sent: Wednesday, 25 February 2009 11:43 PM

To: basant.ya...@gmail.com

Cc: ccie_voice@onlinestudylist.com

Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down


 
Thanks man for your quick reply.



Actually BR1 is not in SRST mode. I am playing multicast MOH from Router's 
flash.




Let's say I am using Multicast IP/base port of Sub and sub goes down, is it 
possible for BR1 to continue playing multicast MOH from its flash.



I couldn't find a wayout to make it work.



Date: Wed, 25 Feb 2009 13:20:35 +0100

Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down

From: basant.ya...@gmail.com

To: kapilatr...@hotmail.com

CC: ccie_voice@onlinestudylist.com



Hi Kapil



Here is a link that will help you:-



http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1118060



Regards

- Basant

On Wed, Feb 25, 2009 at 12:44 PM, Kapil Atrish  wrote:

Hi List,



I want to confirm if there is a way to play MOH from BR1 flash in case Primary 
MOH server is down.



thanks,

 



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Entertainment!
Check it out!



 



Discover your phone style & WIN a Windows Mobile phone. Your style!
Try it now!
 



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Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down

[Kapil Atrish]

Thanks man for your quick reply.

Actually BR1 is not in SRST mode. I am playing multicast MOH from Router's 
flash. 

Let's say I am using Multicast IP/base port of Sub and sub goes down, is it 
possible for BR1 to continue playing multicast MOH from its flash.

I couldn't find a wayout to make it work.

Date: Wed, 25 Feb 2009 13:20:35 +0100
Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down
From: basant.ya...@gmail.com
To: kapilatr...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Hi Kapil

Here is a link that will help you:-

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1118060


Regards
- Basant

On Wed, Feb 25, 2009 at 12:44 PM, Kapil Atrish  wrote:






Hi List,

I want to confirm if there is a way to play MOH from BR1 flash in case Primary 
MOH server is down.

thanks,


Akshay Kumar takes on the two reigning Bollywood Khans. Catch the action on MSN 
Entertainment! Check it out!



_
For the freshest Indian Jobs Visit MSN Jobs
http://www.in.msn.com/jobs

[OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down

[Kapil Atrish]

Hi List,

I want to confirm if there is a way to play MOH from BR1 flash in case Primary 
MOH server is down.

thanks,


_
Chose your Life Partner! Join MSN Matrimony FREE
http://www.in.msn.com/matrimony

Re: [OSL | CCIE_Voice] CCM to GK trunk>>can load balancing be achieved

[Kapil Atrish]

OK if question pre-specifies H.323 IDs to be ccm_1 and ccm_2? Since both the 
trunks can't have same name is this scenario valid and has any workaround. 



Date: Fri, 20 Feb 2009 11:04:02 -0800
Subject: Re: [OSL | CCIE_Voice] CCM to GK trunk>>can load balancing be achieved
From: vma...@ipexpert.com
To: anil...@yahoo.com; ccie_voice@onlinestudylist.com; kapilatr...@hotmail.com





Re: [OSL | CCIE_Voice] CCM to GK trunk>>can load balancing be achieved


Two trunks: 



pub_trunk in DP containing CCM Group = PUB-ONLY

Sub_trunk in DP containing CCM Group = SUB-ONLY



RP > RL > RG > 2 x trunks with circular hunting.



-- 

Vik Malhi – CCIE #13890, CCSI #31584 

Senior Technical Instructor - IPexpert, Inc.



Telephone: +1.810.326.1444 

Fax: +1.810.454.0130 

Mailto: vma...@ipexpert.com





Join our free online support and peer group communities: 

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and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.















From: anil batra 

Reply-To: 

Date: Fri, 20 Feb 2009 10:08:12 -0800 (PST)

To: OSL Group , Kapil Atrish 


Subject: Re: [OSL | CCIE_Voice] CCM to GK trunk>>can load balancing be achieved



Not sure if it possible, In CCM when we create RG-GK we can't define HQ_Trunk_1 
or HQ_Trunk_2 it is simply HQ-Trunk so the call will go thrugh the same trunk 
and since the trunk has DP which has CCM group which has SUB is primary so the 
call will always go thru SUB and only will go thru PUB when SUB fails.



--- On Fri, 2/20/09, Kapil Atrish  wrote:

From: Kapil Atrish 

Subject: [OSL | CCIE_Voice] CCM to GK trunk>>can load balancing be achieved

To: ccie_voice@onlinestudylist.com

Date: Friday, February 20, 2009, 11:28 PM



Hi,



CCM cluster (pub, sub) is registered to a GK. GK correctly shows two trunks 
with _1 and _2.



Requirement is to enable load-balancing of outgoing calls from CCM to GK over 
those two trunks , how can it be achieved? I can achieve GK to CCM 
load-balancing via gw-priority, but requirement is to achieve load blanacing of 
calls the other way. Sub is my primary call processing agent.



Any inputs are highly appreciated...



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Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

[Kapil Atrish]

OK. I tested this and its indeed 4 digits of ANI. If I configure 4 digits the 
loop is broken, and even if I configure e.164 loop is still broken.

It seems if Unity knows the Port number before hand, whether 4 digits or 10 
digits it is preventing the loop rightly.


Date: Thu, 19 Feb 2009 08:34:36 -0800
Subject: Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI
From: vma...@ipexpert.com
To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com





Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI


I think 4 digit # in the first column within UTIM. The best way to make sure is 
use Call Viewer on Unity and have Unity originate a call to a phone which is 
fwded to send the call back to Unity. What is the ANI  for the call being 
returned? 

-- 

Vik Malhi – CCIE #13890, CCSI #31584 

Senior Technical Instructor - IPexpert, Inc.



Telephone: +1.810.326.1444 

Fax: +1.810.454.0130 

Mailto: vma...@ipexpert.com





Join our free online support and peer group communities: 

http://www.IPexpert.com/communities

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.















From: Kapil Atrish 

Date: Thu, 19 Feb 2009 09:59:28 +0530

To: , 

Subject: RE: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI



OK. That means if I un-check "Message Notification" or let it be checked on 
voice-mail ports, there is no harm as long as I implement the fix to avoid 
Unity loop, correct?



I've one more quick question, do I need to put full e.164 number or only 4 
digit number on voice-mail ports to avoid looping of any external call. I 
checked CCM traces and it showed me 4 digit number as well as FQDN  field which 
is e.164 10 digit number of the VM ports. I couldn't find a way in Unity to see 
what VM port number it is receiving? 





Thanks for your inputs Vik..



Date: Wed, 18 Feb 2009 18:13:26 -0800

Subject: Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

From: vma...@ipexpert.com

To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com



Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI Message 
Notification is different to MWI as you correctly state. When we talk about 
Unity Looping it is normally related to loops associated with Msg Notification 
but it can be any call Unity originates that is returned to Unity. See this 
link for the fix:



http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080094b01.shtml#p3a



-- 

Vik Malhi – CCIE #13890, CCSI #31584 

Senior Technical Instructor - IPexpert, Inc.



Telephone: +1.810.326.1444 

Fax: +1.810.454.0130 

Mailto: vma...@ipexpert.com <http://ipexpert.com> 





Join our free online support and peer group communities: 

http://www.IPexpert.com/communities

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.















From: Kapil Atrish http://hotmail.com> >

Date: Wed, 18 Feb 2009 23:45:52 +0530

To: http://onlinestudylist.com> >

Subject: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI



Hi List,

 

I'v a question about Message notification, I read about it, pl let me know if 
my understanding is correct.



I understand Message Notification ports are used when susbcriber has 
notifications enabled on additional devices under Subscriber>>Message 
notification settings, like Home Phone/Pager etc. 

 

So if I don't have notification enabled on additional devices do I still need 
to uncheck "Message Notification" on Ports to avoid Unity looping or is there 
something I am missing? 

 

MWI has nothing to do with "Message Notification ports" but only with "Dialout 
MWI"ports, correct?

 

Thanks in advance...



 



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acquisitions and deals on BPO Watch. Try it now! 
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[OSL | CCIE_Voice] CCM to GK trunk>>can load balancing be achieved

[Kapil Atrish]

Hi,

CCM cluster (pub, sub) is registered to a GK. GK correctly shows two trunks 
with _1 and _2.

Requirement is to enable load-balancing of outgoing calls from CCM to GK over 
those two trunks , how can it be achieved? I can achieve GK to CCM 
load-balancing via gw-priority, but requirement is to achieve load blanacing of 
calls the other way. Sub is my primary call processing agent.

Any inputs are highly appreciated...

_
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Re: [OSL | CCIE_Voice] FRTS using MQC >> is class-default required

[Kapil Atrish]

Thanks Vik for the clarification. I am not sure where I read but if I manage to 
get the source info, I'll def share.




Date: Fri, 20 Feb 2009 06:38:01 -0800
Subject: Re: [OSL | CCIE_Voice] FRTS using MQC >> is class-default required
From: vma...@ipexpert.com
To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com





Re: [OSL | CCIE_Voice] FRTS using MQC >> is class-default required


I think it is required if  you wanted to subject the default class to 
flow-based fair queuing. If you can retrieve the source that says you don’t 
then I’ll appreciate that.

-- 

Vik Malhi – CCIE #13890, CCSI #31584 

Senior Technical Instructor - IPexpert, Inc.



Telephone: +1.810.326.1444 

Fax: +1.810.454.0130 

Mailto: vma...@ipexpert.com





Join our free online support and peer group communities: 

http://www.IPexpert.com/communities

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.















From: Kapil Atrish 

Date: Fri, 20 Feb 2009 17:08:26 +0530

To: 

Subject: [OSL | CCIE_Voice] FRTS using MQC >> is class-default required



Hi List,



Quick question on LLQ with FRTS through MQC. I am not able to recall but I read 
some where the class-default is not required when FRTS is implemented using MQC 
alongwith LLQ. 



Below is the config shap-shot:

policy-map voice

class EF

priority percent 50

class AF

bandwidth percent 5

class class-default   >>>>>>>>>>>>>>>>>>>> Is this required

fair-queue

!

policy-map FRTS

class class-default

shape average xx  x

shape adaptive 

shape fr-voice-adapt deactivation 30

service-policy voice

!



Appreciate any comment on this



















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[OSL | CCIE_Voice] FRTS using MQC >> is class-default required

[Kapil Atrish]

Hi List,

Quick question on LLQ with FRTS through MQC. I am not able to recall but I read 
some where the class-default is not required when FRTS is implemented using MQC 
alongwith LLQ. 

Below is the config shap-shot:
policy-map voice
class EF
priority percent 50
class AF
bandwidth percent 5
class class-default    Is this required
fair-queue
!
policy-map FRTS
class class-default
shape average xx  x
shape adaptive 
shape fr-voice-adapt deactivation 30
service-policy voice
!

Appreciate any comment on this









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[OSL | CCIE_Voice] CUE problem on POD 26

[kapil atrish]

Hi List,
 
My cue module is in rebooting state. Reboot of Router didn't help.  It always 
comes to this stage and halt:
System Now Booting ...[BOOT-ASM]
7
Please enter '***' to change boot configuration:
__
 
I've observed following during reboot. 
 
Verifying signature now...
Signature not a valid base64 encoded entity
Invalid Kernel Signature !!!
Rebooting
 
 
 
Has anybody any idea what's wrong. My config is:
 
interface Service-Engine0/0
 ip unnumbered FastEthernet0/0.360
 service-module ip address 10.26.202.2 255.255.255.0
 service-module ip default-gateway 10.26.202.1
!


  

Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

[Kapil Atrish]

Thanks Greg,

I think there are two scenarios, one when you've to prevent looping of external 
calls which land on Unity and then looped from there onwards. Those calls are 
not originated by unity initially. To fix this you need to apply the solution 
you mentioned below. Any idea, does it have to be e.164 or 4 digit number, say 
I've external phone no mask configured for VM ports?

Second scenario, when Unity originates the calls by way of message notification 
and it gets looped back. You need to apply the fix give by Vik in URL below.


Pl correct me if you find missing link.



> From: gpu...@doc.gov
> To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com
> Date: Wed, 18 Feb 2009 13:52:28 -0500
> Subject: RE: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout 
> MWI
> 
> You are correct that MWI has no related to Message Notification.
> 
> To help prevent Unity from looping through its ports you can goto UTIM and on 
> the ports page, put the DN for the specific port(s) as they are configured in 
> CCM.
> 
> This will allow Unity to know the call came from one of its ports and not to 
> loop back to it.
> 
> greg
> 
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil Atrish
> Sent: Wednesday, February 18, 2009 1:16 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI
> 
> Hi List,
> 
> I'v a question about Message notification, I read about it, pl let me know if 
> my understanding is correct.
> 
> I understand Message Notification ports are used when susbcriber has 
> notifications enabled on additional devices under Subscriber>>Message 
> notification settings, like Home Phone/Pager etc.
> 
> So if I don't have notification enabled on additional devices do I still need 
> to uncheck "Message Notification" on Ports to avoid Unity looping or is there 
> something I am missing?
> 
> MWI has nothing to do with "Message Notification ports" but only with 
> "Dialout MWI"ports, correct?
> 
> Thanks in advance...
> 
> 
> 
> 
> 
> 
> Get the latest buzz on outsourcing. Up to date information on mergers, 
> acquisitions and deals on BPO Watch. Try it now! 
> <http://www.bpowatchindia.com/default.asp>

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Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

[Kapil Atrish]

OK. That means if I un-check "Message Notification" or let it be checked on 
voice-mail ports, there is no harm as long as I implement the fix to avoid 
Unity loop, correct?

I've one more quick question, do I need to put full e.164 number or only 4 
digit number on voice-mail ports to avoid looping of any external call. I 
checked CCM traces and it showed me 4 digit number as well as FQDN  field which 
is e.164 10 digit number of the VM ports. I couldn't find a way in Unity to see 
what VM port number it is receiving? 


Thanks for your inputs Vik..

Date: Wed, 18 Feb 2009 18:13:26 -0800
Subject: Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI
From: vma...@ipexpert.com
To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com





Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI


Message Notification is different to MWI as you correctly state. When we talk 
about Unity Looping it is normally related to loops associated with Msg 
Notification but it can be any call Unity originates that is returned to Unity. 
See this link for the fix:



http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080094b01.shtml#p3a



-- 

Vik Malhi – CCIE #13890, CCSI #31584 

Senior Technical Instructor - IPexpert, Inc.



Telephone: +1.810.326.1444 

Fax: +1.810.454.0130 

Mailto: vma...@ipexpert.com





Join our free online support and peer group communities: 

http://www.IPexpert.com/communities

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.















From: Kapil Atrish 

Date: Wed, 18 Feb 2009 23:45:52 +0530

To: 

Subject: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI



Hi List,

 

I'v a question about Message notification, I read about it, pl let me know if 
my understanding is correct.



I understand Message Notification ports are used when susbcriber has 
notifications enabled on additional devices under Subscriber>>Message 
notification settings, like Home Phone/Pager etc. 

 

So if I don't have notification enabled on additional devices do I still need 
to uncheck "Message Notification" on Ports to avoid Unity looping or is there 
something I am missing? 

 

MWI has nothing to do with "Message Notification ports" but only with "Dialout 
MWI"ports, correct?

 

Thanks in advance...



 



Get the latest buzz on outsourcing. Up to date information on mergers, 
acquisitions and deals on BPO Watch. Try it now! 
<http://www.bpowatchindia.com/default.asp> 


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http://in.msn.com/coxandkings

[OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

[Kapil Atrish]

Hi List,

 

I'v a question about Message notification, I read about it, pl let me know if 
my understanding is correct.


I understand Message Notification ports are used when susbcriber has 
notifications enabled on additional devices under Subscriber>>Message 
notification settings, like Home Phone/Pager etc. 

 

So if I don't have notification enabled on additional devices do I still need 
to uncheck "Message Notification" on Ports to avoid Unity looping or is there 
something I am missing? 

 

MWI has nothing to do with "Message Notification ports" but only with "Dialout 
MWI"ports, correct?

 

Thanks in advance...


 

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Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

I can see the precedence files inside:

C:\Program Files\Cisco\TFTPPath\English_United_States

File names start with ANNMLPP*.wav


Date: Wed, 4 Feb 2009 23:16:14 -0600
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: ryanstudyvo...@gmail.com
To: kapilatr...@hotmail.com
CC: lovingprin...@gmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com

Any idea where the precedence wav file is located?

On Tue, Feb 3, 2009 at 11:59 PM, Kapil Atrish  wrote:






Yup, you are right. Those are not under English_United States folder. You'll 
find ANN_Fastbusy.wav inside the path you;ve mentioned. So I believe it is 
correct.

Date: Tue, 3 Feb 2009 19:50:22 -0600

Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: ryanstudyvo...@gmail.com
To: lovingprin...@gmail.com

CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com


Where are the Blocked pattern reason codes greeting stored at?
Are they under program files>cisco>tftpath>united states

On Tue, Feb 3, 2009 at 12:46 AM, kamal yousaf  wrote:


Yeah..I did that but putting DN didn't work.You would need Secondary AC pilot 
#. Besides, I prefer to use Unity rather than going through this method.At 
least for lab, it won't be advisable unless strictly asked to do so.




On Tue, Feb 3, 2009 at 5:02 PM, Kapil Atrish  wrote:







Cool...I did not check for the TCD Service Parameter. I think if I set this 
parameter the second AC would not be required. I may simply put a DN as "Always 
route member" to extend fast busy to caller after initial MOH. Otherwise I'll 
also follow your solution.




Vik/Mark: Do you think it is an acceptable solution? Question is to customize 
annunciator and we are using MOH to acheive the results? 






Date: Tue, 3 Feb 2009 15:39:43 +1100

Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com
To: kapilatr...@hotmail.com



CC: anthony.ye...@gmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com




Kapil, 

 If you dial your first AC pilot # , you should hear greeting .If you dial 
second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd to 
first,i.e add 2nd AC pilot # as Always Route Member , after hold time expires, 
call will be routed to dummy pilot point and you will get 'user busy'.I did 
also change Service Parameter for TCD so that AC can route calls to directory 
numbers with unknown state.





Regds
On Mon, Feb 2, 2009 at 9:25 PM, Kapil Atrish  wrote:









Hi Kamal,

 

I created additional AC Pilot with queuing disabled and pointed first one to 
the new AC as "Alwasy Route Member". I keep on getting the MOH from first AC 
even though queuing timer is over. Can you pl comment if you achieved it 
differently?





 

I am able to route the call to a CTI_RP as "Alwasy Route Member" and point this 
RP to a route-pattern which further points it to the Gateway. The RP string is 
invalid and non-routable by the GW. Using this method, the caller simply gets 
dropped after queuing timer is over. No Fast-busy to caller but MOH gets 
played. 





 

When I try pointing AC "Always Route Member" filed to any Route-pattern 
directly, I get the following message:


The Directory Number you entered in the selected Partition is associated with a 
device that can not be a member of a Hunt Group.


 

Pl let me know how you achieved Anthony's method?


 





Date: Thu, 29 Jan 2009 18:05:31 +1100
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com




To: anthony.ye...@gmail.com
CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com





I tested it and it works great.Thanks Anthony for kind help.



On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung  wrote:

What you can try is assign a dummy AC pilot point to the original AC
Pilot Point as the 'Always Route Member' creating a linked hunt group.
Then for this second dummy AC pilot point assign a dummy AC user like




you did w/ the first. But instead, disable queuing for this second
dummy Pilot Point. After the hold time expires for the first AC pilot,
the call will be forwarded to the second AC pilot. Since queuing is
disabled, the call should drop BUT w/ a disconnect cause of 'user




busy'.




On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish  wrote:
> I did not put the TP directly inside the Hunt-Group. I put a CTIRP as
> "Always Route Member" and on CTIRP I did a forward all to the TP.




>
> I am yet to try the solution given by Christian. I'll put the call to a
> gateway vi

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Yup, you are right. Those are not under English_United States folder. You'll 
find ANN_Fastbusy.wav inside the path you;ve mentioned. So I believe it is 
correct.

Date: Tue, 3 Feb 2009 19:50:22 -0600
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: ryanstudyvo...@gmail.com
To: lovingprin...@gmail.com
CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com

Where are the Blocked pattern reason codes greeting stored at?
Are they under program files>cisco>tftpath>united states

On Tue, Feb 3, 2009 at 12:46 AM, kamal yousaf  wrote:

Yeah..I did that but putting DN didn't work.You would need Secondary AC pilot 
#. Besides, I prefer to use Unity rather than going through this method.At 
least for lab, it won't be advisable unless strictly asked to do so.



On Tue, Feb 3, 2009 at 5:02 PM, Kapil Atrish  wrote:







Cool...I did not check for the TCD Service Parameter. I think if I set this 
parameter the second AC would not be required. I may simply put a DN as "Always 
route member" to extend fast busy to caller after initial MOH. Otherwise I'll 
also follow your solution.



Vik/Mark: Do you think it is an acceptable solution? Question is to customize 
annunciator and we are using MOH to acheive the results? 






Date: Tue, 3 Feb 2009 15:39:43 +1100

Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com
To: kapilatr...@hotmail.com


CC: anthony.ye...@gmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com



Kapil, 

 If you dial your first AC pilot # , you should hear greeting .If you dial 
second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd to 
first,i.e add 2nd AC pilot # as Always Route Member , after hold time expires, 
call will be routed to dummy pilot point and you will get 'user busy'.I did 
also change Service Parameter for TCD so that AC can route calls to directory 
numbers with unknown state.




Regds
On Mon, Feb 2, 2009 at 9:25 PM, Kapil Atrish  wrote:








Hi Kamal,

 

I created additional AC Pilot with queuing disabled and pointed first one to 
the new AC as "Alwasy Route Member". I keep on getting the MOH from first AC 
even though queuing timer is over. Can you pl comment if you achieved it 
differently?




 

I am able to route the call to a CTI_RP as "Alwasy Route Member" and point this 
RP to a route-pattern which further points it to the Gateway. The RP string is 
invalid and non-routable by the GW. Using this method, the caller simply gets 
dropped after queuing timer is over. No Fast-busy to caller but MOH gets 
played. 




 

When I try pointing AC "Always Route Member" filed to any Route-pattern 
directly, I get the following message:


The Directory Number you entered in the selected Partition is associated with a 
device that can not be a member of a Hunt Group.


 

Pl let me know how you achieved Anthony's method?


 





Date: Thu, 29 Jan 2009 18:05:31 +1100
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com



To: anthony.ye...@gmail.com
CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com




I tested it and it works great.Thanks Anthony for kind help.



On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung  wrote:

What you can try is assign a dummy AC pilot point to the original AC
Pilot Point as the 'Always Route Member' creating a linked hunt group.
Then for this second dummy AC pilot point assign a dummy AC user like



you did w/ the first. But instead, disable queuing for this second
dummy Pilot Point. After the hold time expires for the first AC pilot,
the call will be forwarded to the second AC pilot. Since queuing is
disabled, the call should drop BUT w/ a disconnect cause of 'user



busy'.




On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish  wrote:
> I did not put the TP directly inside the Hunt-Group. I put a CTIRP as
> "Always Route Member" and on CTIRP I did a forward all to the TP.



>
> I am yet to try the solution given by Christian. I'll put the call to a
> gateway via RP and see if I can get fast-busy to the caller after initial
> queuing prompt.
>
> 



> Date: Tue, 27 Jan 2009 21:30:58 +1100
> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> From: lovingprin...@gmail.com
> To: kapilatr...@hotmail.com
> CC: christian.hennr...@intact-is.com; gree...@googlemail.com;



> ccie_voice@onlinestudylist.com; anil...@yahoo.com
>
> I tried same way.It plays greeting only once.I also changed service
> parameter for Cisco TCD "Allow Rout

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Cool...I did not check for the TCD Service Parameter. I think if I set this 
parameter the second AC would not be required. I may simply put a DN as "Always 
route member" to extend fast busy to caller after initial MOH. Otherwise I'll 
also follow your solution.

Vik/Mark: Do you think it is an acceptable solution? Question is to customize 
annunciator and we are using MOH to acheive the results? 






Date: Tue, 3 Feb 2009 15:39:43 +1100
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com
To: kapilatr...@hotmail.com
CC: anthony.ye...@gmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com

Kapil, 

 If you dial your first AC pilot # , you should hear greeting .If you dial 
second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd to 
first,i.e add 2nd AC pilot # as Always Route Member , after hold time expires, 
call will be routed to dummy pilot point and you will get 'user busy'.I did 
also change Service Parameter for TCD so that AC can route calls to directory 
numbers with unknown state.


Regds
On Mon, Feb 2, 2009 at 9:25 PM, Kapil Atrish  wrote:






Hi Kamal,

 

I created additional AC Pilot with queuing disabled and pointed first one to 
the new AC as "Alwasy Route Member". I keep on getting the MOH from first AC 
even though queuing timer is over. Can you pl comment if you achieved it 
differently?


 

I am able to route the call to a CTI_RP as "Alwasy Route Member" and point this 
RP to a route-pattern which further points it to the Gateway. The RP string is 
invalid and non-routable by the GW. Using this method, the caller simply gets 
dropped after queuing timer is over. No Fast-busy to caller but MOH gets 
played. 


 

When I try pointing AC "Always Route Member" filed to any Route-pattern 
directly, I get the following message:


The Directory Number you entered in the selected Partition is associated with a 
device that can not be a member of a Hunt Group.


 

Pl let me know how you achieved Anthony's method?


 





Date: Thu, 29 Jan 2009 18:05:31 +1100
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com

To: anthony.ye...@gmail.com
CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; 
gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com


I tested it and it works great.Thanks Anthony for kind help.



On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung  wrote:

What you can try is assign a dummy AC pilot point to the original AC
Pilot Point as the 'Always Route Member' creating a linked hunt group.
Then for this second dummy AC pilot point assign a dummy AC user like

you did w/ the first. But instead, disable queuing for this second
dummy Pilot Point. After the hold time expires for the first AC pilot,
the call will be forwarded to the second AC pilot. Since queuing is
disabled, the call should drop BUT w/ a disconnect cause of 'user

busy'.




On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish  wrote:
> I did not put the TP directly inside the Hunt-Group. I put a CTIRP as
> "Always Route Member" and on CTIRP I did a forward all to the TP.

>
> I am yet to try the solution given by Christian. I'll put the call to a
> gateway via RP and see if I can get fast-busy to the caller after initial
> queuing prompt.
>
> 

> Date: Tue, 27 Jan 2009 21:30:58 +1100
> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> From: lovingprin...@gmail.com
> To: kapilatr...@hotmail.com
> CC: christian.hennr...@intact-is.com; gree...@googlemail.com;

> ccie_voice@onlinestudylist.com; anil...@yahoo.com
>
> I tried same way.It plays greeting only once.I also changed service
> parameter for Cisco TCD "Allow Routing with Unknown Line State" to True ,and

> retried.Call still doesn't end.
>
> Kapil,
>  how did you add TP as member in HuntGroup.In my case, it gives error saying
> that member should be a valid DN on system.I was able to add phone/CTIRP DNs

> though.
>
> On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish 
> wrote:
>
> I tried with RP/TP >> Block this pattern and in that case call stays in

> queue. AC takes the call out of the queue only when it is routed to a
> registered end-point that's what I've observed.
>
> I'll try to route it to some unallocated number pointing it to the GW and

> see if it works.
>
> Thanks for the input.
>
>
>> Date: Tue, 27 Jan 2009 10:39:31 +0100
>> From: christian.hennr...@intact-is.com
>> To: kapilatr...@hotmail.com

>> CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us;
>> ccie_voice@onlinestudylist.com
>>

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Hi Kamal,
 
I created additional AC Pilot with queuing disabled and pointed first one to 
the new AC as "Alwasy Route Member". I keep on getting the MOH from first AC 
even though queuing timer is over. Can you pl comment if you achieved it 
differently?
 
I am able to route the call to a CTI_RP as "Alwasy Route Member" and point this 
RP to a route-pattern which further points it to the Gateway. The RP string is 
invalid and non-routable by the GW. Using this method, the caller simply gets 
dropped after queuing timer is over. No Fast-busy to caller but MOH gets 
played. 
 
When I try pointing AC "Always Route Member" filed to any Route-pattern 
directly, I get the following message:
The Directory Number you entered in the selected Partition is associated with a 
device that can not be a member of a Hunt Group.
 
Pl let me know how you achieved Anthony's method?
 



Date: Thu, 29 Jan 2009 18:05:31 +1100Subject: Re: [OSL | CCIE_Voice] 
Annunciator to PSTN - Will it be acceptable?From: lovingprin...@gmail.comto: 
anthony.ye...@gmail.comcc: kapilatr...@hotmail.com; 
ccie_voice@onlinestudylist.com; gree...@googlemail.com; 
christian.hennr...@intact-is.com; anil...@yahoo.comi tested it and it works 
great.Thanks Anthony for kind help.
On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung  wrote:
What you can try is assign a dummy AC pilot point to the original ACPilot Point 
as the 'Always Route Member' creating a linked hunt group.Then for this second 
dummy AC pilot point assign a dummy AC user likeyou did w/ the first. But 
instead, disable queuing for this seconddummy Pilot Point. After the hold time 
expires for the first AC pilot,the call will be forwarded to the second AC 
pilot. Since queuing isdisabled, the call should drop BUT w/ a disconnect cause 
of 'userbusy'.


On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish  
wrote:> I did not put the TP directly inside the Hunt-Group. I put a CTIRP as> 
"Always Route Member" and on CTIRP I did a forward all to the TP.>> I am yet to 
try the solution given by Christian. I'll put the call to a> gateway via RP and 
see if I can get fast-busy to the caller after initial> queuing prompt.>> 
> Date: Tue, 27 Jan 2009 21:30:58 +1100> 
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?> 
From: lovingprin...@gmail.com> To: kapilatr...@hotmail.com> CC: 
christian.hennr...@intact-is.com; gree...@googlemail.com;> 
ccie_voice@onlinestudylist.com; anil...@yahoo.com>> I tried same way.It plays 
greeting only once.I also changed service> parameter for Cisco TCD "Allow 
Routing with Unknown Line State" to True ,and> retried.Call still doesn't 
end.>> Kapil,>  how did you add TP as member in HuntGroup.In my case, it gives 
error saying> that member should be a valid DN on system.I was able to add 
phone/CTIRP DNs> though.>> On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish 
> wrote:>> I tried with RP/TP >> Block this pattern 
and in that case call stays in> queue. AC takes the call out of the queue only 
when it is routed to a> registered end-point that's what I've observed.>> I'll 
try to route it to some unallocated number pointing it to the GW and> see if it 
works.>> Thanks for the input.>>>> Date: Tue, 27 Jan 2009 10:39:31 +0100>> 
From: christian.hennr...@intact-is.com>> To: kapilatr...@hotmail.com>> CC: 
gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us;>> 
ccie_voice@onlinestudylist.com>> Subject: Re: [OSL | CCIE_Voice] Annunciator to 
PSTN - Will it be>> acceptable?>>>> what about routing to a number CUCM, which 
does not exist, or even to a>> PSTN number, which is unallocated?>>>> 
Christian>>>> Kapil Atrish schrieb:>> > The requirement is to drop the call 
within CCM itself. I don't want to>> > use Unity/IPCCX/TCL for this purpose.>> 
>>> > 
>> > 
Date: Tue, 27 Jan 2009 09:16:49 +>> > Subject: Re: [OSL | CCIE_Voice] 
Annunciator to PSTN - Will it be>> > acceptable?>> > From: 
gree...@googlemail.com>> > To: anil...@yahoo.com>> > CC: 
christian.hennr...@intact-is.com; cpar...@cparker.us;>> > 
kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com>> >>> > Folks,>> >>> > 
To get the call to disconnect you can use do the following:>> >>> > Create a 
CTI RP cfwd all to voicemail.>> >>> > In VM create a CH with the extension 
number of the CTI RP and configure>> > the greeting to be blank and then after 
greeting send the caller to hang>> > up.>&g

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Yes you are right Juan but we are not going to login the AC user to console 
application. It is used just to populate the line group members. I could've 
added dummy phone DNs instead of AC user and achieved the same result. "Always 
Route Member" field is what we are using here and not the AC user/lines.

Let me know if it doesn't clarify...


Date: Thu, 29 Jan 2009 11:32:27 +0100
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: juan.c...@gmail.com
To: lovingprin...@gmail.com
CC: anthony.ye...@gmail.com; kapilatr...@hotmail.com; 
ccie_voice@onlinestudylist.com; christian.hennr...@intact-is.com; 
anil...@yahoo.com; gree...@googlemail.com

I do not understand the part "Then for this second dummy AC pilot point assign 
a dummy AC user like
you did w/ the first" . These AC users (not the 'ac' user created in DC 
directory and linked to both AC pilot points) aren't they only used when using 
the Attendand Client/user-line AC members

regards,Juan




On Thu, Jan 29, 2009 at 8:05 AM, kamal yousaf  wrote:

I tested it and it works great.Thanks Anthony for kind help.


On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung  wrote:

What you can try is assign a dummy AC pilot point to the original AC

Pilot Point as the 'Always Route Member' creating a linked hunt group.

Then for this second dummy AC pilot point assign a dummy AC user like

you did w/ the first. But instead, disable queuing for this second

dummy Pilot Point. After the hold time expires for the first AC pilot,

the call will be forwarded to the second AC pilot. Since queuing is

disabled, the call should drop BUT w/ a disconnect cause of 'user

busy'.



On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish  wrote:

> I did not put the TP directly inside the Hunt-Group. I put a CTIRP as

> "Always Route Member" and on CTIRP I did a forward all to the TP.

>

> I am yet to try the solution given by Christian. I'll put the call to a

> gateway via RP and see if I can get fast-busy to the caller after initial

> queuing prompt.

>

> 

> Date: Tue, 27 Jan 2009 21:30:58 +1100

> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

> From: lovingprin...@gmail.com

> To: kapilatr...@hotmail.com

> CC: christian.hennr...@intact-is.com; gree...@googlemail.com;

> ccie_voice@onlinestudylist.com; anil...@yahoo.com

>

> I tried same way.It plays greeting only once.I also changed service

> parameter for Cisco TCD "Allow Routing with Unknown Line State" to True ,and

> retried.Call still doesn't end.

>

> Kapil,

>  how did you add TP as member in HuntGroup.In my case, it gives error saying

> that member should be a valid DN on system.I was able to add phone/CTIRP DNs

> though.

>

> On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish 

> wrote:

>

> I tried with RP/TP >> Block this pattern and in that case call stays in

> queue. AC takes the call out of the queue only when it is routed to a

> registered end-point that's what I've observed.

>

> I'll try to route it to some unallocated number pointing it to the GW and

> see if it works.

>

> Thanks for the input.

>

>

>> Date: Tue, 27 Jan 2009 10:39:31 +0100

>> From: christian.hennr...@intact-is.com

>> To: kapilatr...@hotmail.com

>> CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us;




>> ccie_voice@onlinestudylist.com

>> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be

>> acceptable?

>>

>> what about routing to a number CUCM, which does not exist, or even to a

>> PSTN number, which is unallocated?

>>

>> Christian

>>

>> Kapil Atrish schrieb:

>> > The requirement is to drop the call within CCM itself. I don't want to

>> > use Unity/IPCCX/TCL for this purpose.

>> >

>> > 

>> > Date: Tue, 27 Jan 2009 09:16:49 +

>> > Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be

>> > acceptable?

>> > From: gree...@googlemail.com

>> > To: anil...@yahoo.com

>> > CC: christian.hennr...@intact-is.com; cpar...@cparker.us;

>> > kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com

>> >

>> > Folks,

>> >

>> > To get the call to disconnect you can use do the following:

>> >

>> > Create a CTI RP cfwd all to voicemail.

>> >

>> > In VM create a CH with the extension number of the CTI RP and configure

>> > the greeting to be blank and then after greeting send the caller to hang

>> > up.

>> >

Re: [OSL | CCIE_Voice] Channels block after upgrade of DSPs.

[kapil atrish]

Hi,
   
  Do you've "dspfarm" and "dsp services dspfarm" under voice-cards??
   
  

Daniel Sobrinho  wrote:
Hello,  
  
Could please help me with a doubt? I've been made an upgrade of DSPs in my 
router 2851 to increase the capacity for transcoder and conference bridge.   
  
After applied it a number maximum of sessions for both dspfarm profiles, my 
gateway stopped receiving external calls from pstn and could not get more to do 
external calls by pstn link.  My service provider said that my voice channels 
were blocked. Does anybody knows like that or tell me if I did something wrong?
  IOS: c2800nm-adventerprisek9-mz.124-3.bin
  Before applied the new DSPs:
  sccp local GigabitEthernet0/0
sccp ccm 10.55.14.6 identifier 2 version 4.1 
sccp ccm 10.55.14.1 identifier 1 version 4.1 
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 2 register CFB_FTZ
 associate profile 1 register XCODE_FTZ
!
dspfarm profile 1 transcode
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec gsmfr
 codec g729r8
 codec g729br8
 maximum sessions 14
 associate application SCCP
!
dspfarm profile 2 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 5
 associate application SCCP
  After applied the new DSPs:
  sccp local GigabitEthernet0/0
sccp ccm 10.55.14.6 identifier 2 version 4.1 
sccp ccm 10.55.14.1 identifier 1 version 4.1 
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 2 register CFB_FTZ
 associate profile 1 register XCODE_FTZ
!
dspfarm profile 1 transcode
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec gsmfr
 codec g729r8
 codec g729br8
 maximum sessions 18
 associate application SCCP
!
dspfarm profile 2 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 16
 associate application SCCP
  ==
  #show dspfarm profile 1
Dspfarm Profile Configuration
   Profile ID = 1, Service = TRANSCODING, Resource ID = 1  
 Profile Description :  
 Profile Admin State : UP 
 Profile Operation State : ACTIVE 
 Application : SCCP   Status : ASSOCIATED 
 Resource Provider : FLEX_DSPRM   Status : UP 
 Number of Resource Configured : 18 
 Number of Resource Available : 18
 Codec Configuration 
 Codec : g711ulaw, Maximum Packetization Period : 30 
 Codec : g711alaw, Maximum Packetization Period : 30 
 Codec : g729ar8, Maximum Packetization Period : 60 
 Codec : g729abr8, Maximum Packetization Period : 60 
 Codec : gsmfr, Maximum Packetization Period : 20 
 Codec : g729r8, Maximum Packetization Period : 60 
 Codec : g729br8, Maximum Packetization Period : 60
  
#show dspfarm profile 1 2
Dspfarm Profile Configuration
   Profile ID = 2, Service = CONFERENCING, Resource ID = 2  
 Profile Description :  
 Profile Admin State : UP 
 Profile Operation State : ACTIVE 
 Application : SCCP   Status : ASSOCIATED 
 Resource Provider : FLEX_DSPRM   Status : UP 
 Number of Resource Configured : 16 
 Number of Resource Available : 16
 Codec Configuration 
 Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required 
 Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required 
 Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required 
 Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required 
 Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required 
 Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required

  
  #show diag
  
Slot 0:
C2851 Motherboard with 2GE and integrated VPN Port adapter, 2 ports
Port adapter is analyzed 
Port adapter insertion time unknown
Onboard VPN: FW ver01100200
EEPROM contents at hardware discovery:
PCB Serial Number: FOC10292RYB
Hardware Revision: 1.0
Top Assy. Part Number: 800-26922-02
Board Revision   : A0
Deviation Number : 0
Fab Version  : 03
RMA Test History : 00
RMA Number   : 0-0-0-0
RMA History  : 00
Processor type   : 87 
Hardware date code   : 20060720
Chassis Serial Number: FTX1031A3XD
Chassis MAC Address  : 0018.b9ce.5558
MAC Address block size   : 32
CLEI Code: COM3E00BRA
Product (FRU) Number : CISCO2851  
Part Number  : 73-8480-04
Version Identifier   : V03 
EEPROM format version 4
EEPROM contents (hex):
  0x00: 04 FF C1 8B 46 4F 43 31 30 32 39 32 52 59 42 40
  0x10: 03 E9 41 01 00 C0 46 03 20 00 69 2A 02 42 41 30
  0x20: 88 00 00 00 00 02 03 03 00 81 00 00 00 00 04 00
  0x30: 09 87 83 01 32 1A 30 C2 8B 46 54 58 31 30 33 31
  0x40: 41 33 58 44 C3 06 00 18 B9 CE 55 58 43 00 20 C6
  0x50: 8A 43 4F 4D 33 45 30 30 42 52 41 CB 8F 43 49 53
  0x60: 43 4F 32 38 35 31 20 20 20 20 20 20 82 49 21 20
  0x70: 04 

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Cool... I'll try this as well.
 
I noticed AC takes the call out of queue only & only when it has a registered 
device which can answer the call. Else call remains in queue. But thanks for 
opening the additional door. I'll def check this one.
 
 
> Date: Wed, 28 Jan 2009 22:52:40 -0600> Subject: Re: [OSL | CCIE_Voice] 
> Annunciator to PSTN - Will it be acceptable?> From: anthony.ye...@gmail.com> 
> To: kapilatr...@hotmail.com> CC: lovingprin...@gmail.com; 
> ccie_voice@onlinestudylist.com; gree...@googlemail.com; 
> christian.hennr...@intact-is.com; anil...@yahoo.com> > What you can try is 
> assign a dummy AC pilot point to the original AC> Pilot Point as the 'Always 
> Route Member' creating a linked hunt group.> Then for this second dummy AC 
> pilot point assign a dummy AC user like> you did w/ the first. But instead, 
> disable queuing for this second> dummy Pilot Point. After the hold time 
> expires for the first AC pilot,> the call will be forwarded to the second AC 
> pilot. Since queuing is> disabled, the call should drop BUT w/ a disconnect 
> cause of 'user> busy'.> > On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish 
>  wrote:> > I did not put the TP directly inside the 
> Hunt-Group. I put a CTIRP as> > "Always Route Member" and on CTIRP I did a 
> forward all to the TP.> >> > I am yet to try the solution given by Christian. 
> I'll put the call to a> > gateway via RP and see if I can get fast-busy to 
> the caller after initial> > queuing prompt.> >> > 
> > > Date: Tue, 27 Jan 2009 21:30:58 +1100> > 
> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?> 
> > From: lovingprin...@gmail.com> > To: kapilatr...@hotmail.com> > CC: 
> christian.hennr...@intact-is.com; gree...@googlemail.com;> > 
> ccie_voice@onlinestudylist.com; anil...@yahoo.com> >> > I tried same way.It 
> plays greeting only once.I also changed service> > parameter for Cisco TCD 
> "Allow Routing with Unknown Line State" to True ,and> > retried.Call still 
> doesn't end.> >> > Kapil,> > how did you add TP as member in HuntGroup.In my 
> case, it gives error saying> > that member should be a valid DN on system.I 
> was able to add phone/CTIRP DNs> > though.> >> > On Tue, Jan 27, 2009 at 8:48 
> PM, Kapil Atrish > > wrote:> >> > I tried with RP/TP 
> >> Block this pattern and in that case call stays in> > queue. AC takes the 
> call out of the queue only when it is routed to a> > registered end-point 
> that's what I've observed.> >> > I'll try to route it to some unallocated 
> number pointing it to the GW and> > see if it works.> >> > Thanks for the 
> input.> >> >> >> Date: Tue, 27 Jan 2009 10:39:31 +0100> >> From: 
> christian.hennr...@intact-is.com> >> To: kapilatr...@hotmail.com> >> CC: 
> gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us;> >> 
> ccie_voice@onlinestudylist.com> >> Subject: Re: [OSL | CCIE_Voice] 
> Annunciator to PSTN - Will it be> >> acceptable?> >>> >> what about routing 
> to a number CUCM, which does not exist, or even to a> >> PSTN number, which 
> is unallocated?> >>> >> Christian> >>> >> Kapil Atrish schrieb:> >> > The 
> requirement is to drop the call within CCM itself. I don't want to> >> > use 
> Unity/IPCCX/TCL for this purpose.> >> >> >> > 
> > >> 
> > Date: Tue, 27 Jan 2009 09:16:49 +> >> > Subject: Re: [OSL | CCIE_Voice] 
> Annunciator to PSTN - Will it be> >> > acceptable?> >> > From: 
> gree...@googlemail.com> >> > To: anil...@yahoo.com> >> > CC: 
> christian.hennr...@intact-is.com; cpar...@cparker.us;> >> > 
> kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com> >> >> >> > Folks,> 
> >> >> >> > To get the call to disconnect you can use do the following:> >> >> 
> >> > Create a CTI RP cfwd all to voicemail.> >> >> >> > In VM create a CH 
> with the extension number of the CTI RP and configure> >> > the greeting to 
> be blank and then after greeting send the caller to hang> >> > up.> >> >> >> 
> > In the ac hunt group config add the CTI RP as the

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

I did not put the TP directly inside the Hunt-Group. I put a CTIRP as
"Always Route Member" and on CTIRP I did a forward all to the TP.



I am yet to try the solution given by Christian. I'll put the call to a
gateway via RP and see if I can get fast-busy to the caller after
initial queuing prompt.

Date: Tue, 27 Jan 2009 21:30:58 +1100
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: lovingprin...@gmail.com
To: kapilatr...@hotmail.com
CC: christian.hennr...@intact-is.com; gree...@googlemail.com; 
ccie_voice@onlinestudylist.com; anil...@yahoo.com

I tried same way.It plays greeting only once.I also changed service parameter 
for Cisco TCD "Allow Routing with Unknown Line State" to True ,and retried.Call 
still doesn't end.


Kapil, 
 how did you add TP as member in HuntGroup.In my case, it gives error saying 
that member should be a valid DN on system.I was able to add phone/CTIRP DNs 
though.

On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish  wrote:






I tried with RP/TP >> Block this pattern and in that case call
stays in queue. AC takes the call out of the queue only when it is
routed to a registered end-point that's what I've observed.



I'll try to route it to some unallocated number pointing it to the GW and see 
if it works.



Thanks for the input.



> Date: Tue, 27 Jan 2009 10:39:31 +0100
> From: christian.hennr...@intact-is.com
> To: kapilatr...@hotmail.com

> CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; 
> ccie_voice@onlinestudylist.com

> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> 
> what about routing to a number CUCM, which does not exist, or even to a 
> PSTN number, which is unallocated?

> 
> Christian
> 
> Kapil Atrish schrieb:
> > The requirement is to drop the call within CCM itself. I don't want to 
> > use Unity/IPCCX/TCL for this purpose.
> > 
> > 

> > Date: Tue, 27 Jan 2009 09:16:49 +
> > Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> > From: gree...@googlemail.com

> > To: anil...@yahoo.com
> > CC: christian.hennr...@intact-is.com; cpar...@cparker.us; 

> > kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com
> > 
> > Folks,

> >  
> > To get the call to disconnect you can use do the following:
> >  
> > Create a CTI RP cfwd all to voicemail.
> >  
> > In VM create a CH with the extension number of the CTI RP and configure 

> > the greeting to be blank and then after greeting send the caller to hang up.
> >  
> > In the ac hunt group config add the CTI RP as the always route member.
> >  
> > In acconfig.bat for the annunicator ac pilot set the hold time to be 

> > something other than 0 seconds
> >  
> > After this time has passed the call will be forwarded to unity and 
> > disconnected - you get a little bit of ringing as the call gets to unity 

> > which I cant get rid of.
> > 
> > 2009/1/27 anil batra 
> > 
> > I too tried the way Kapil mentioned and faced same issue as he did.

> >     The call from PSTN does it the announcement but the call never gets
> > disonncted, it seems the queue is holdin git for forever. Anyone
> > here has tested this and have some workaround please.

> > 
> > --- On *Tue, 1/27/09, Kapil Atrish //* wrote:
> > 
> > From: Kapil Atrish 

> > 
> > Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be
> > acceptable?
> > To: christian.hennr...@intact-is.com, cpar...@cparker.us

> > Cc: ccie_voice@onlinestudylist.com
> > Date: Tuesday, January 27, 2009, 11:38 AM
> > 
> > 
> > Chris,

> > 
> > Your suspicion is what I've in mind that's why I am trying to
> > avoid using Unity/IPCCX/TCL.
> > 
> > I've tested AC workaround and its working for me but couple of

> > catches. First of all, the file is in form of MOH and not
> > annunciator which was the original requirement of the question.
> > Secondly, I am not able to disconnect the call. The message

> > keeps on playing until caller drops the call.
> > 
> > 
> > thanks,
> > Kapil Atrish
> > 
> >  > Date: Mon, 26 Jan 2009 18:57:28 +0100

> >  > From: christian.hennr...@intact-is.com
> >  > To: cpar...@cparker.us

> >  > CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com;
> > ccie_vo

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

I tried with RP/TP >> Block this pattern and in that case call
stays in queue. AC takes the call out of the queue only when it is
routed to a registered end-point that's what I've observed.



I'll try to route it to some unallocated number pointing it to the GW and see 
if it works.



Thanks for the input.



> Date: Tue, 27 Jan 2009 10:39:31 +0100
> From: christian.hennr...@intact-is.com
> To: kapilatr...@hotmail.com
> CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; 
> ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> 
> what about routing to a number CUCM, which does not exist, or even to a 
> PSTN number, which is unallocated?
> 
> Christian
> 
> Kapil Atrish schrieb:
> > The requirement is to drop the call within CCM itself. I don't want to 
> > use Unity/IPCCX/TCL for this purpose.
> > 
> > 
> > Date: Tue, 27 Jan 2009 09:16:49 +
> > Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> > From: gree...@googlemail.com
> > To: anil...@yahoo.com
> > CC: christian.hennr...@intact-is.com; cpar...@cparker.us; 
> > kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com
> > 
> > Folks,
> >  
> > To get the call to disconnect you can use do the following:
> >  
> > Create a CTI RP cfwd all to voicemail.
> >  
> > In VM create a CH with the extension number of the CTI RP and configure 
> > the greeting to be blank and then after greeting send the caller to hang up.
> >  
> > In the ac hunt group config add the CTI RP as the always route member.
> >  
> > In acconfig.bat for the annunicator ac pilot set the hold time to be 
> > something other than 0 seconds
> >  
> > After this time has passed the call will be forwarded to unity and 
> > disconnected - you get a little bit of ringing as the call gets to unity 
> > which I cant get rid of.
> > 
> > 2009/1/27 anil batra 
> > 
> > I too tried the way Kapil mentioned and faced same issue as he did.
> > The call from PSTN does it the announcement but the call never gets
> > disonncted, it seems the queue is holdin git for forever. Anyone
> > here has tested this and have some workaround please.
> > 
> > --- On *Tue, 1/27/09, Kapil Atrish //* wrote:
> > 
> > From: Kapil Atrish 
> > 
> > Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be
> > acceptable?
> > To: christian.hennr...@intact-is.com, cpar...@cparker.us
> > Cc: ccie_voice@onlinestudylist.com
> > Date: Tuesday, January 27, 2009, 11:38 AM
> > 
> > 
> > Chris,
> > 
> > Your suspicion is what I've in mind that's why I am trying to
> > avoid using Unity/IPCCX/TCL.
> > 
> >     I've tested AC workaround and its working for me but couple of
> > catches. First of all, the file is in form of MOH and not
> > annunciator which was the original requirement of the question.
> > Secondly, I am not able to disconnect the call. The message
> > keeps on playing until caller drops the call.
> > 
> > 
> > thanks,
> > Kapil Atrish
> > 
> >  > Date: Mon, 26 Jan 2009 18:57:28 +0100
> >  > From: christian.hennr...@intact-is.com
> >  > To: cpar...@cparker.us
> >  > CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com;
> > ccie_voice@onlinestudylist.com
> >  > Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it
> > be acceptable?
> >  >
> >  > Hi,
> >  >
> >  > what about having a MoH File, that is playing the message to
> > the caller.
> >  > MoH file is played in the AC Hunt group with queueing
> > activated and no
> >  > AC operators logged in. So you would use only CUCM to play
> > the message.
> >  >
> >  > I have not tested that idea, but it might be workable.
> >  >
> >  > As far as there is nothing stated, which prevents you from
> > using Unity,
> >  > I would use Unity.
> >  >
> >  > Regards
> >  >
> >  > Chris Parker schrieb:
> >  > > The only thing that makes me suspiciou

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

The requirement is to drop the call within CCM itself. I don't want to use 
Unity/IPCCX/TCL for this purpose.

Date: Tue, 27 Jan 2009 09:16:49 +
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
From: gree...@googlemail.com
To: anil...@yahoo.com
CC: christian.hennr...@intact-is.com; cpar...@cparker.us; 
kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com

Folks,
 
To get the call to disconnect you can use do the following:
 
Create a CTI RP cfwd all to voicemail.
 
In VM create a CH with the extension number of the CTI RP and configure the 
greeting to be blank and then after greeting send the caller to hang up.
 
In the ac hunt group config add the CTI RP as the always route member.
 
In acconfig.bat for the annunicator ac pilot set the hold time to be something 
other than 0 seconds
 
After this time has passed the call will be forwarded to unity and disconnected 
- you get a little bit of ringing as the call gets to unity which I cant get 
rid of.


2009/1/27 anil batra 





I too tried the way Kapil mentioned and faced same issue as he did. The call 
from PSTN does it the announcement but the call never gets disonncted, it seems 
the queue is holdin git for forever. Anyone here has tested this and have some 
workaround please.


--- On Tue, 1/27/09, Kapil Atrish  wrote:

From: Kapil Atrish  

Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
To: christian.hennr...@intact-is.com, cpar...@cparker.us

Cc: ccie_voice@onlinestudylist.com
Date: Tuesday, January 27, 2009, 11:38 AM 





Chris,

Your suspicion is what I've in mind that's why I am trying to avoid using 
Unity/IPCCX/TCL.

I've tested AC workaround and its working for me but couple of catches. First 
of all, the file is in form of MOH and not annunciator which was the original 
requirement of the question. Secondly, I am not able to disconnect the call. 
The message keeps on playing until caller drops the call.



thanks,
Kapil Atrish 

> Date: Mon, 26 Jan 2009 18:57:28 +0100
> From: christian.hennr...@intact-is.com
> To: cpar...@cparker.us

> CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; 
> ccie_voice@onlinestudylist.com

> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> 
> Hi,
> 
> what about having a MoH File, that is playing the message to the caller. 
> MoH file is played in the AC Hunt group with queueing activated and no 

> AC operators logged in. So you would use only CUCM to play the message.
> 
> I have not tested that idea, but it might be workable.
> 
> As far as there is nothing stated, which prevents you from using Unity, 

> I would use Unity.
> 
> Regards
> 
> Chris Parker schrieb:
> > The only thing that makes me suspicious about using Unity to play the
> > announcement is that this requirement was listed under the Media

> > section. This leads me to believe they want you to use the annunciator.
> > Otherwise wouldn't it be under the Voicemail/Unity section?
> > 
> > Regardless I don't think you can do it any other way unless you hairpin

> > the call through Unity to send the call to the annunciator since the VM
> > ports are skinny registrations.
> > 
> > Chris
> > 
> > Ryan Trauernicht wrote:
> > > That is what I thought but I opened a TAC case and they claim you

> > > can, but cant figure out how.
> > >
> > > Thanks,
> > > Ryan Trauernicht
> > >
> > > On Mon, Jan 26, 2009 at 3:21 AM, Juan  > > <mailto:juan.c...@gmail.com>> wrote:
> > >
> > > I remember reading in the SRND that you can only engage the
> > > annunciator for SCCP devices if I remember correctly - so not to

> > > the PSTN.
> > >
> > >
> > > cheers,
> > > Juan
> > >
> > >
> > > On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht
> > > mailto:ryanstudyvo...@gmail.com>> wrote:

> > >
> > > Not sure why you are going through all that trouble and not
> > > just sending it to unity as a call handler and hang up after
> > > message played.
> > >

> > > I don't know how to play an ANN from a PSTN call, I have
> > > engaged TAC and they are still working on it and they can't
> > > even figure it out right now.
> > >

> > > Any ideas?
> > >
> > > Thanks,
> > > Ryan Trauernicht
> > >
> > > On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish
> > > mailto:kapilatr...@hotmail.com>> wrote:

> > >
> > > Hi list,
> > >
> > > Following I did:
> > >
> > > Create a new MOH Audio Source using
> > > 

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Chris,

Your suspicion is what I've in mind that's why I am trying to avoid using 
Unity/IPCCX/TCL.

I've tested AC workaround and its working for me but couple of catches. First 
of all, the file is in form of MOH and not annunciator which was the original 
requirement of the question. Secondly, I am not able to disconnect the call. 
The message keeps on playing until caller drops the call.


thanks,
Kapil Atrish 

> Date: Mon, 26 Jan 2009 18:57:28 +0100
> From: christian.hennr...@intact-is.com
> To: cpar...@cparker.us
> CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; 
> ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> 
> Hi,
> 
> what about having a MoH File, that is playing the message to the caller. 
> MoH file is played in the AC Hunt group with queueing activated and no 
> AC operators logged in. So you would use only CUCM to play the message.
> 
> I have not tested that idea, but it might be workable.
> 
> As far as there is nothing stated, which prevents you from using Unity, 
> I would use Unity.
> 
> Regards
> 
> Chris Parker schrieb:
> > The only thing that makes me suspicious about using Unity to play the
> > announcement is that this requirement was listed under the Media
> > section. This leads me to believe they want you to use the annunciator.
> > Otherwise wouldn't it be under the Voicemail/Unity section?
> > 
> > Regardless I don't think you can do it any other way unless you hairpin
> > the call through Unity to send the call to the annunciator since the VM
> > ports are skinny registrations.
> > 
> > Chris
> > 
> > Ryan Trauernicht wrote:
> >  > That is what I thought but I opened a TAC case and they claim you
> >  > can, but cant figure out how.
> >  >
> >  > Thanks,
> >  > Ryan Trauernicht
> >  >
> >  > On Mon, Jan 26, 2009 at 3:21 AM, Juan  >  > <mailto:juan.c...@gmail.com>> wrote:
> >  >
> >  > I remember reading in the SRND that you can only engage the
> >  > annunciator for SCCP devices if I remember correctly - so not to
> >  > the PSTN.
> >  >
> >  >
> >  > cheers,
> >  > Juan
> >  >
> >  >
> >  > On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht
> >  > mailto:ryanstudyvo...@gmail.com>> wrote:
> >  >
> >  > Not sure why you are going through all that trouble and not
> >  > just sending it to unity as a call handler and hang up after
> >  > message played.
> >  >
> >  > I don't know how to play an ANN from a PSTN call, I have
> >  > engaged TAC and they are still working on it and they can't
> >  > even figure it out right now.
> >  >
> >  > Any ideas?
> >  >
> >  > Thanks,
> >  > Ryan Trauernicht
> >  >
> >  > On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish
> >  > mailto:kapilatr...@hotmail.com>> wrote:
> >  >
> >  > Hi list,
> >  >
> >  > Following I did:
> >  >
> >  > Create a new MOH Audio Source using
> >  > "AAExtnOutOfService.wav". Prompt available inside Wfavvid
> >  > folder
> >  >
> >  > Create a TP covering all unassigned DNs for example: 11xx,
> >  > do Called party Xform to 1155
> >  >
> >  > Create a AC Pilot 1155, give any DP say: ANN_PSTN
> >  > AC Hunt-Group>>Give any AC user. No need to login to
> >  > Attendant Console.
> >  > Run acconfig.bat>>Enable Queuing
> >  > Inside DP: ANN_PSTN give User Hold MOH Source as
> >  > "AAExtnOutOfService.wav".
> >  >
> >  > Now, whenever you dial any unassigned number withing range
> >  > 11xx, you'll hear "AAExtnOutOfService.wav" but the problem
> >  > is that I am not able to make the PSTN call drop.
> >  >
> >  > I tried routing calls to TP inside AC Hunt-Group>>Always
> >  > Route member is TP>>TP has Block Pattern --Not working.
> >  >
> >  > "AAExtnOutOfService.wav"keeps on playing.
> >  >
> >  > I tried routing calls to Route-Point (Always Route Member)

Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Chris,

Your suspicion is what I've in mind that's why I am trying to avoid using 
Unity/IPCCX/TCL.

I've tested AC workaround and its working for me but couple of catches. First 
of all, the file is in form of MOH and not annunciator which was the original 
requirement of the question. Secondly, I am not able to disconnect the call. 
The message keeps on playing until caller drops the call.


thanks,
Kapil Atrish 

> Date: Mon, 26 Jan 2009 18:57:28 +0100
> From: christian.hennr...@intact-is.com
> To: cpar...@cparker.us
> CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; 
> ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
> 
> Hi,
> 
> what about having a MoH File, that is playing the message to the caller. 
> MoH file is played in the AC Hunt group with queueing activated and no 
> AC operators logged in. So you would use only CUCM to play the message.
> 
> I have not tested that idea, but it might be workable.
> 
> As far as there is nothing stated, which prevents you from using Unity, 
> I would use Unity.
> 
> Regards
> 
> Chris Parker schrieb:
> > The only thing that makes me suspicious about using Unity to play the
> > announcement is that this requirement was listed under the Media
> > section. This leads me to believe they want you to use the annunciator.
> > Otherwise wouldn't it be under the Voicemail/Unity section?
> > 
> > Regardless I don't think you can do it any other way unless you hairpin
> > the call through Unity to send the call to the annunciator since the VM
> > ports are skinny registrations.
> > 
> > Chris
> > 
> > Ryan Trauernicht wrote:
> >  > That is what I thought but I opened a TAC case and they claim you
> >  > can, but cant figure out how.
> >  >
> >  > Thanks,
> >  > Ryan Trauernicht
> >  >
> >  > On Mon, Jan 26, 2009 at 3:21 AM, Juan  >  > <mailto:juan.c...@gmail.com>> wrote:
> >  >
> >  > I remember reading in the SRND that you can only engage the
> >  > annunciator for SCCP devices if I remember correctly - so not to
> >  > the PSTN.
> >  >
> >  >
> >  > cheers,
> >  > Juan
> >  >
> >  >
> >  > On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht
> >  > mailto:ryanstudyvo...@gmail.com>> wrote:
> >  >
> >  > Not sure why you are going through all that trouble and not
> >  > just sending it to unity as a call handler and hang up after
> >  > message played.
> >  >
> >  > I don't know how to play an ANN from a PSTN call, I have
> >  > engaged TAC and they are still working on it and they can't
> >  > even figure it out right now.
> >  >
> >  > Any ideas?
> >  >
> >  > Thanks,
> >  > Ryan Trauernicht
> >  >
> >  > On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish
> >  > mailto:kapilatr...@hotmail.com>> wrote:
> >  >
> >  > Hi list,
> >  >
> >  > Following I did:
> >  >
> >  > Create a new MOH Audio Source using
> >  > "AAExtnOutOfService.wav". Prompt available inside Wfavvid
> >  > folder
> >  >
> >  > Create a TP covering all unassigned DNs for example: 11xx,
> >  > do Called party Xform to 1155
> >  >
> >  > Create a AC Pilot 1155, give any DP say: ANN_PSTN
> >  > AC Hunt-Group>>Give any AC user. No need to login to
> >  > Attendant Console.
> >  > Run acconfig.bat>>Enable Queuing
> >  > Inside DP: ANN_PSTN give User Hold MOH Source as
> >  > "AAExtnOutOfService.wav".
> >  >
> >  > Now, whenever you dial any unassigned number withing range
> >  > 11xx, you'll hear "AAExtnOutOfService.wav" but the problem
> >  > is that I am not able to make the PSTN call drop.
> >  >
> >  > I tried routing calls to TP inside AC Hunt-Group>>Always
> >  > Route member is TP>>TP has Block Pattern --Not working.
> >  >
> >  > "AAExtnOutOfService.wav"keeps on playing.
> >  >
> >  > I tried routing calls to Route-Point (Always Route Member)

[OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?

[Kapil Atrish]

Hi list,

Following I did:

Create a new MOH Audio Source using "AAExtnOutOfService.wav". Prompt available 
inside Wfavvid folder

Create a TP covering all unassigned DNs for example: 11xx, do Called party 
Xform to 1155

Create a AC Pilot 1155, give any DP say: ANN_PSTN
AC Hunt-Group>>Give any AC user. No need to login to Attendant Console.
Run acconfig.bat>>Enable Queuing
Inside DP: ANN_PSTN give User Hold MOH Source as "AAExtnOutOfService.wav".

Now, whenever you dial any unassigned number withing range 11xx, you'll hear 
"AAExtnOutOfService.wav" but the problem is that I am not able to make the PSTN 
call drop.

I tried routing calls to TP inside AC Hunt-Group>>Always Route member is TP>>TP 
has Block Pattern --Not working. 

"AAExtnOutOfService.wav"keeps on playing.

I tried routing calls to Route-Point (Always Route Member) inside AC 
Hunt-Group>>CTI_RP has Forward all to TP>>TP has Block Pattern --Not working. 
"AAExtnOutOfService.wav"keeps on playing.

I tried routing calls to a registered Phone DN as Always Route Member>>Forward 
all to TP>>>>TP has Block Pattern --Not working. "AAExtnOutOfService.wav"keeps 
on playing.

Can someone help me achieve call drop here without using IPCCX/Unity/TCL? 


Thanks,
Kapil Atrish

_
Plug in to the MSN Tech channel for a full update on the latest gizmos that 
made an impact.
http://computing.in.msn.com/

[OSL | CCIE_Voice] Annunciator to PSTN - will it be acceptable?

[Kapil Atrish]

Hi list,

Following I did:

Create a new MOH Audio Source using "AAExtnOutOfService.wav". Prompt available 
inside Wfavvid folder

Create a TP covering all unassigned DNs for example: 11xx, do Called party 
Xform to 1155

Create a AC Pilot 1155, give any DP say: ANN_PSTN
AC Hunt-Group>>Give any AC user. No need to login to Attendant Console.
Run acconfig.bat>>Enable Queuing
Inside DP: ANN_PSTN give User Hold MOH Source as "AAExtnOutOfService.wav".

Now, whenever you dial any unassigned number withing range 11xx, you'll hear 
"AAExtnOutOfService.wav" but the problem is that I am not able to make the PSTN 
call drop.

I tried routing calls to TP inside AC Hunt-Group>>Always Route member is TP>>TP 
has Block Pattern --Not working. 

"AAExtnOutOfService.wav"keeps on playing.

I tried routing calls to Route-Point (Always Route Member) inside AC 
Hunt-Group>>CTI_RP has Forward all to TP>>TP has Block Pattern --Not working. 
"AAExtnOutOfService.wav"keeps on playing.

I tried routing calls to a registered Phone DN as Always Route Member>>Forward 
all to TP>>>>TP has Block Pattern --Not working. "AAExtnOutOfService.wav"keeps 
on playing.

Can someone help me achieve call drop here without using IPCCX/Unity/TCL? 


Thanks,
Kapil Atrish

_
Chose your Life Partner! Join MSN Matrimony FREE
http://www.in.msn.com/matrimony

Re: [OSL | CCIE_Voice] B-ACD

[kapil atrish]

Yes ReadMe is available in router flash. I've used it. But be aware it doesn't 
cover each and every parameter (for ex. drop-through). 
   
  DocCD is no more provided in the Lab. It has been replaced with following URL:
  http://www.cisco.com/web/psa/products/index.html
   
   
   
  

kamal yousaf  wrote:
  No need to memorize anything.If readme file is available,then its 
good.Otherwise,you DocCD and copy-paste script for your use.

Here is the navigation Structure:

DocCD:
Voice and Unified Communications --> IP Telephony Call Control -->CUCME --> 
Configuration Guidelines --> Cisco Unified CME B-ACD and Tcl Call-Handling 
Applications

Rgds

  On Thu, Jan 22, 2009 at 3:47 PM, Greg Hauser  wrote:
Hi –
  I have been spending a lot of time with  B-ACD and was wondering if anyone 
knew if CME flash with aap-b-acd-2.1.0.0.tcl readme will be made available to 
us or do we need to memorize the B-ACD config?  
   
  Thanks 
  Greg Hauser






   

Re: [OSL | CCIE_Voice] CME BACD - drop-through not working

[Kapil Atrish]

Hi Narinder,
 
Your configuration worked for me. After having another look at the same cisco 
doc I realised in all the examples of drop-through there is only single "param 
aa-hunt-grps 1" under "AA" TCL script. I had "aa-hunt-grps 2" which seems 
invalid in drop-through scenario.
 
 
thanks for all the help you provided.
 
thanks,
Kapil Atrish



From: narinder.ku...@uxcg.com.auto: kapilatr...@hotmail.com; 
ccie_vo...@onlinestudylist.comdate: Mon, 19 Jan 2009 23:39:09 +1100Subject: RE: 
[OSL | CCIE_Voice] CME BACD - drop-through not working





Kapil,
This is the configuration which is working on my router no issues, try it.
 
application
 service callq flash:app-b-acd-2.1.0.0.tcl
  param queue-len 10
  param aa-hunt1 3020
  param number-of-hunt-grps 1
  param queue-manager-debugs 1
 !
 service aa flash:app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 0
  param number-of-hunt-grps 1
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 3500
  paramspace english location flash:
  param second-greeting-time 60
  param drop-through-prompt _dt_prompt.au
  param call-retry-timer 15
  param voice-mail 3600
  param max-time-call-retry 700
  param service-name callq
 


From: Kapil Atrish [mailto:kapilatr...@hotmail.com] Sent: Monday, 19 January 
2009 11:35 PMTo: Kumar, Narinder; ccie_vo...@onlinestudylist.comsubject: RE: 
[OSL | CCIE_Voice] CME BACD - drop-through not working
 
Thanks Narinder for quick reply, I tried the flash: option but same results, 
not working.paramspace english location tftp://172.30.1.4/ --- I tried with 
flash: option with no luck.I am checking attached cisco doc for configuring 
drop-through.  Thanks,Kapil Atrish



From: narinder.ku...@uxcg.com.auto: kapilatr...@hotmail.com; 
ccie_vo...@onlinestudylist.comdate: Mon, 19 Jan 2009 22:50:08 +1100Subject: RE: 
[OSL | CCIE_Voice] CME BACD - drop-through not working

Kapil,
 
I am hoping the bacd script is in router flash. I have always done drop through 
with a single HG, not sure you can achieve multiple HG”s with drop through ( I 
could be wrong need to double check.
 
Change service callq tftp://172.30.1.10/app-b-acd-2.1.0.0.tcl toservice 
callq flash:app-b-acd-2.1.0.0.tcl
 
Change service dropthruaa tftp://172.30.1.10/app-b-acd-aa-2.1.0.0.tclto   
service dropthruaa flash:app-b-acd-aa-2.1.0.0.tcl
Also ur tftp path in paramspace english location tftp://172.30.1.4/ is 
different to other is it just a typo or some other reason behind this.
 
Thanks
Narinder
 
 
 
 


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil AtrishSent: 
Monday, 19 January 2009 9:34 PMTo: ccie_vo...@onlinestudylist.comsubject: [OSL 
| CCIE_Voice] CME BACD - drop-through not working
 
Hi, I've CME 3.3 (supports drop-through). I've configured BACD with 
drop-through functionality but its not working the desired way. When I dial the 
pilot no: I get silence. If I press 2 or 3 the call get routed to the 
respective hunt-group. Can someone pl suggest what can I check/change? I've 
attached:CME Config,dir flash:,Output of "SHOW CALL APPLICATION SESSION" when 
script is active,output of debug voice application script, I've reloaded the 
router but no good. MY TCL scripts are being read successfully from TFTP Server 
(another question, Is it mandatory in drop-through that TCL script must be 
accessed via TFTP? I've checked cisco docs which always shows TCL scripts 
accessed from TFTP whenver drop-through scenario is discussed).  Thanks in 
advance...



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[OSL | CCIE_Voice] CME BACD - drop-through not working

[Kapil Atrish]

Hi,
 
I've CME 3.3 (supports drop-through). I've configured BACD with drop-through 
functionality but its not working the desired way. When I dial the pilot no: I 
get silence. If I press 2 or 3 the call get routed to the respective 
hunt-group. Can someone pl suggest what can I check/change?
 
I've attached:
CME Config,
dir flash:,
Output of "SHOW CALL APPLICATION SESSION" when script is active,
output of debug voice application script,
 
I've reloaded the router but no good. MY TCL scripts are being read 
successfully from TFTP Server (another question, Is it mandatory in 
drop-through that TCL script must be accessed via TFTP? I've checked cisco docs 
which always shows TCL scripts accessed from TFTP whenver drop-through scenario 
is discussed).
 
 
Thanks in advance...
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Find a better job. We have plenty. Visit MSN Jobs
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Router#sh run
Building configuration...

Current configuration : 3930 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
resource policy
!
memory-size iomem 5
ip subnet-zero
ip cef
!
!
application
 service callq tftp://172.30.1.10/app-b-acd-2.1.0.0.tcl
  param aa-hunt3 4001
  param queue-len 10
  param queue-manager-debugs 1
  param aa-hunt2 4000
  param number-of-hunt-grps 2
 !
 service dropthruaa tftp://172.30.1.10/app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 0
  param number-of-hunt-grps 2
  param drop-through-option 3
  param handoff-string dropthruaa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 4110
  paramspace english location tftp://172.30.1.4/
  param drop-through-prompt _bacd_allagentsbusy.au
  param second-greeting-time 60
  param queue-manager-debugs 1
  param call-retry-timer 15
  param max-time-call-retry 600
  param voice-mail 5000
  param service-name callq
 !
!
!
!
!
!
!
!
interface FastEthernet0/0
 ip address 192.168.10.2 255.255.254.0
 speed 100
 full-duplex
!
interface FastEthernet0/1
 ip address 172.30.1.4 255.255.255.0
 speed 100
 full-duplex
!
ip classless
!
!
ip http server
no ip http secure-server
!
!
!
!
tftp-server flash:app-b-acd-aa-2.1.0.0.tcl
tftp-server flash:app-b-acd-2.1.0.0.tcl
tftp-server flash:en_bacd_allagentsbusy.au
tftp-server flash:en_bacd_invalidoption.au
tftp-server flash:music-on-hold.au
!
control-plane
!
!
dial-peer voice 2 voip
 modem passthrough nse codec g711ulaw
 session target ipv4:172.30.1.4
 fax-relay ecm disable
 fax rate 7200
 fax protocol cisco
!
dial-peer voice 1 voip
 destination-pattern 4100
 session target ipv4:172.30.1.4
 incoming called-number 4100
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
!
dial-peer voice 3 voip
 service dropthruaa
 destination-pattern 4110
 session target ipv4:172.30.1.4
 incoming called-number 4110
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
!
!
!
gatekeeper
 shutdown
!
!
telephony-service
 max-ephones 10
 max-dn 10
 ip source-address 172.30.1.4 port 2000
 auto assign 1 to 4
 create cnf-files version-stamp Jan 01 2002 00:00:00
 max-conferences 8 gain -6
 moh music-on-hold.au
!
ephone-dn  1
 number 4130
 intercom 4131
!
!
ephone-dn  2
 number 4131
 intercom 4130
!
!
ephone-dn  3  dual-line
 number 4183
!
!
ephone-dn  4  dual-line
 number 4184
 huntstop channel
!
!
ephone-dn  5
 number 4185
 pickup-group 4185
!
!
ephone-dn  6  dual-line
 number 4186
!
!
ephone-dn  7  dual-line
 number 4187
!
!
ephone  1
 username "ph1" password 1
 mac-address 0200.4C4F.4F50
 paging-dn 3
 type CIPC
 button  1:3 2:4
!
!
!
ephone  2
 mac-address 0050.56C0.0001
 type 7960
 button  1:6 2:7
!
!
ephone-hunt 1 sequential
 pilot 4180
 list 4183, 4184
!
!
ephone-hunt 3 sequential
 pilot 4000
 list 4183, 4186
!
!
ephone-hunt 4 sequential
 pilot 4001
 list 4184, 4187
!
!
alias exec srb show run | begin
!
line con 0
line aux 0
line vty 0 4
 login
!
!
end

Router#
Router#dir flash:
Directory of flash:/

2  -rw-   24679  app-b-acd-2.1.0.0.tcl
3  -rw-   33870  app-b-acd-aa-2.1.0.0.tcl
4  -rw-   75650  en_bacd_allagentsbusy.au
5  -rw-   83291  en_bacd_disconnect.au
6  -rw-   63055  en_bacd_enter_dest.au
7  -rw-   37952  en_bacd_invalidoption.au
8  -rw-  496521  en_bacd_music_on_hold.au
9  -rw-  123446  en_bacd_options_menu.au
   10  -rw-   42978  en_bacd_welcome.au
   11  -rw-   18346  app-b-acd-2.1.0.0.ReadMe
   12  -rw-  496521  music-on-hold.au

536870908 bytes total (534442496 bytes free)





OUTPUT OF "SHOW CALL APPLICATION SESSION" WHEN SCRIPT IN USE:

Router#
Router#show call application sessions
Session ID 16

App: 

[OSL | CCIE_Voice] Unity SR

[Kapil Atrish]

Hi List,

Want to confirm there is no SR available for Unity 4.0(5)? The only available 
one is as below which we can expect in the lab:


Filename
CiscoUnity4.0.5-ServicePacks-ENU-CD1.exe

Release Date 
10/Jun/2005

Thanks,

_
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Re: [OSL | CCIE_Voice] IPCC script change

[kapil atrish]

Nope. I've done it manier times, you don't need to restart anything. 

Mike O  wrote:   When you change to a different script in 
IPCC do  you need to restart any services?
  
  
 Thanks,
 Mike


   

Re: [OSL | CCIE_Voice] using Change B-Channel Maintenance for IOS T1 or not?

[kapil atrish]

Jeremy,
   
  I know whatever comments have come are not answering your doubt. I also've 
the same doubt and since both of the method works I think proctor is the best 
guy to tell which one to use
   
  In my last attempt I did without using Channel B maintenance status only god 
knows if that was right/wrong as I didn't get full marks in relevant section. 
  I think someone who has failed but got 100% in Voice Gateway section is the 
best guy to answer this
   
  

"Pulos, Greg"  wrote:
  The b channel maintenance should be used to allow you to change the state of 
the channels in real time while the circuit is up. Commonly used for 
troubleshooting.

CallMan can change the state of the channels in real time to either:

0 - in service
1 - graceful out of service
2 - forceful out of service

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
saralilin2...@yahoo.co.jp
Sent: Wednesday, December 24, 2008 1:36 AM
To: co jeremy; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] using Change B-Channel Maintenance for IOS T1 
or not?


we only need to do b-channel maintenance if question ask for bottom up right? 
if we choose top down this is not needed, am i right?

Sara

jeremy co wrote:

Hi,

I've seen some workbooks use Change B-Channel Maintenance option to busyout 
unused channels on T1 of IOS GWs as well as 6500 T1 while some of them only use 
this option on 6500 T1.

In cisco Docs, I can it specified to use thi option for "MGCP gateways"


So which method should be used?

btw, I use both and both works for IOS .


Jeremy




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[OSL | CCIE_Voice] AC-Broadcast hunting not working

[Kapil Atrish]

Hi,
 
Can someone pl suggest how to debug this issue? I am still not able to make it 
work?
 
Thanks...



From: kapilatr...@hotmail.comto: ccie_vo...@onlinestudylist.comsubject: 
AC-Broadcast hunting not workingDate: Mon, 22 Dec 2008 22:43:43 +0530

Hi list,I've configured Attendant Console and is working fine for Longest 
Available/Circular Hunting but not in Broadcast Hunting mode. I've line members 
and users in Line Group. When system is configured for LAA/Circular Hunting, AC 
user can see call coming in inside the AC console, but when broadcast hunting 
is configured the calling party gets MOH from the DP of AC Pilot (User hold 
Audio Source). AC User cannot see call inside the AC console. I've tried line 
group with only users (no phones), phones + users but same result. Restart of 
CCM didn't resolve.Any help is highly appreciated.
_
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[OSL | CCIE_Voice] AC-Broadcast hunting not working

[Kapil Atrish]

Hi list,

I've configured Attendant Console and is working fine for Longest 
Available/Circular Hunting but not in Broadcast Hunting mode. I've line members 
and users in Line Group. When system is configured for LAA/Circular Hunting, AC 
user can see call coming in inside the AC console, but when broadcast hunting 
is configured the calling party gets MOH from the DP of AC Pilot (User hold 
Audio Source). AC User cannot see call inside the AC console. I've tried line 
group with only users (no phones), phones + users but same result. Restart of 
CCM didn't resolve.

Any help is highly appreciated.

_
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Re: [OSL | CCIE_Voice] Announciator messages to PSTN

[kapil atrish]

It won't be a normal call even if annunciator answers the PSTN call (firstly it 
doesn't) because difference in Call Clearing Cause Code issued by GW/CCM for 
unallocated number, number busy or Call cleared normally etc..
   
  

Chris Parker <[EMAIL PROTECTED]> wrote:
  Sounds correct to me.

I guess the interesting part is that if you did have something in call 
manager to match the number (like a CTI route point with a line number 
) and then forward it to a translation pattern that has call block 
checked, why wont the annunciator play that message - "Your call cannot 
be completed as dialed"? All I have heard is that it just doesn't work. 
I realize of course from the ISDN perspective, it would just look like a 
completed call answered by the annunciator.

Christian Hennrich wrote:
> Hi,
>
> as far as I know all messages with the appropriate ISDN disconnect 
> causes are played by the network provider itself and not from any PBX.
>
> So if you would like to play message to the PSTN, you need Unity, 
> because CCM will send the ISDN disconnect cause and not play any 
> message. But if you have a translation or catch all route pattern in 
> CCM for not available numbers. Then you are able to send the call to 
> unity, where you play a message to the caller. But you need to be 
> aware, that the ISDN code is like a normal call. I would therefore 
> also think that any file manipulation will not help, because CCM sends 
> the ISDN disconnect cause.
>
> Please correct me, if I wrong
>
> Regards
>
> Chris Parker schrieb:
>> So does that mean that the annunciator will play messages to a call 
>> coming from Unity but not from a gateway? Or does it mean that Unity 
>> is used to play a recorded message to the caller in place of the 
>> annunciator? I've seen this question about annunciator pop up a few 
>> times and the answer always seems to be to send it to Unity. I'm just 
>> trying to understand what exactly Unity does in this scenario.
>>
>> I have also heard the annunciator will only play to SCCP and MGCP 
>> devices but not H323. If that is the case then if you have an MGCP 
>> gateway like the 6608 and IOS MGCP shouldn't the annunciator work 
>> with them?
>>
>> Thanks
>>
>> Chris
>>
>> Hardesty, Scott wrote:
>>> You can not use annunciator for pstn. You need to route the call 
>>> to unity and use call handler...
>>>
>>>
>>> 
>>> Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked 
>>> Solutions
>>> 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | 
>>> mailto:[EMAIL PROTECTED]
>>> D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
>>>
>>> 
>>> -Original Message-
>>>
>>> From: Michael Shavrov 
>>> Sent: Tuesday, November 18, 2008 11:56 AM
>>> To: ccie_voice@onlinestudylist.com 
>>> Subject: [OSL | CCIE_Voice] Announciator messages to PSTN
>>>
>>> Hi,
>>> 
>>> How to play messages with announciator to PSTN? For example, if PSTN 
>>> phone calls number, which belongs to location but has no configured 
>>> DN, user should hear message "Number is not in service". I tried to 
>>> configure both, Route Pattern and Translation Pattern with "Block 
>>> pattern" - it works internally, but does not work from PSTN. Also, 
>>> there is no configurable option for "Number not in service" - call 
>>> manager just rejects the call.
>>> 
>>> Mike
>>>
>>>
>>> 
>>
>>
>> __
>> This email has been scanned by the MessageLabs Email Security System.
>> For more information please visit http://www.messagelabs.com/email 
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>> __
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>
>



   

Re: [OSL | CCIE_Voice] Announciator messages to PSTN

[kapil atrish]

Hi,
   
  I also did lot of research on this requirement some time back and the only 
workable solution was to use send a call to Unity (CH, record a Greeting). I 
even tried creating dummy CTI RP to simulate unregistered device but only 
fast-busy to PSTN caller. 

Christian Hennrich <[EMAIL PROTECTED]> wrote:
  Hi,

as far as I know all messages with the appropriate ISDN disconnect 
causes are played by the network provider itself and not from any PBX.

So if you would like to play message to the PSTN, you need Unity, 
because CCM will send the ISDN disconnect cause and not play any 
message. But if you have a translation or catch all route pattern in CCM 
for not available numbers. Then you are able to send the call to unity, 
where you play a message to the caller. But you need to be aware, that 
the ISDN code is like a normal call. I would therefore also think that 
any file manipulation will not help, because CCM sends the ISDN 
disconnect cause.

Please correct me, if I wrong

Regards

Chris Parker schrieb:
> So does that mean that the annunciator will play messages to a call 
> coming from Unity but not from a gateway? Or does it mean that Unity is 
> used to play a recorded message to the caller in place of the 
> annunciator? I've seen this question about annunciator pop up a few 
> times and the answer always seems to be to send it to Unity. I'm just 
> trying to understand what exactly Unity does in this scenario.
> 
> I have also heard the annunciator will only play to SCCP and MGCP 
> devices but not H323. If that is the case then if you have an MGCP 
> gateway like the 6608 and IOS MGCP shouldn't the annunciator work with 
> them?
> 
> Thanks
> 
> Chris
> 
> Hardesty, Scott wrote:
>> You can not use annunciator for pstn. You need to route the call to 
>> unity and use call handler...
>>
>>
>> 
>> Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked 
>> Solutions
>> 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | 
>> mailto:[EMAIL PROTECTED]
>> D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
>>
>> 
>> -Original Message-
>>
>> From: Michael Shavrov 
>> Sent: Tuesday, November 18, 2008 11:56 AM
>> To: ccie_voice@onlinestudylist.com 
>> Subject: [OSL | CCIE_Voice] Announciator messages to PSTN
>>
>> Hi,
>> 
>> How to play messages with announciator to PSTN? For example, if PSTN 
>> phone calls number, which belongs to location but has no configured 
>> DN, user should hear message "Number is not in service". 
>> I tried to configure both, Route Pattern and Translation Pattern with 
>> "Block pattern" - it works internally, but does not work from PSTN. 
>> Also, there is no configurable option for "Number not in service" - 
>> call manager just rejects the call.
>> 
>> Mike
>>
>>
>> 
> 
> 
> __
> This email has been scanned by the MessageLabs Email Security System.
> For more information please visit http://www.messagelabs.com/email 
> __
> 
> __
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> For more information please visit http://www.messagelabs.com/email 
> __


   

Re: [OSL | CCIE_Voice] AAR simple question. How to use AAR in scenarios other that TEHO?

[kapil atrish]

Put Site A phones in AAR Group say SiteA, location SiteA. Put Site B phones in 
AAR Group say SiteB, location SiteB. Set AAR prefix, AAR CSS and route-patterns.

Lower down the b/w and AAR triggers.

jeremy co <[EMAIL PROTECTED]> wrote: Hi,

assuming following scenario:


   _ WAN ___
 |  |
3XXX -SITE A--   SITE B--- 4XXX
  | |
  - PSTN -

Centralized model
SITE A : CCM

if TEHO is to be implemented ,AAR locations can be set on GWs of SITE A and 
SITE B. e.g  calls to site B through WAN exceed defined BW then use siteA's GW 
to route calls to pstn by AAR .
 
if CME on siteB or GK implemented ,it's easy to run this, and location can be 
put on GWs.

 
But what happened if we want to put BW restriction for site B 4XXX phones? e.g 
site B has 10 phones and we have only 96Kbps , so we want just 4 calls go 
through wan and the fifth call should route by pstn from site A to site B.
 
 in above scenario there is no gateway to put location on it since 4xxx are 
site numbers, not pstn numbers so they route WITHIN callmanager.


Any suggestion where to apply location concept to use AAR in above scenario?


Jeremy
 

   

Re: [OSL | CCIE_Voice] cme phones to two different unity systems

[kapil atrish]

translation-profile incoming on ephone-dn and translate the voicemail number to 
CUE or Unity Pilot. Leave other phone without translation.


Balamurugan Singaram <[EMAIL PROTECTED]> wrote: Hi,
  
 We have two cme phones in BR2 two different unity systems:
  
 1st phone press messages button and go to unity 4.0.5 greetings
 2nd phone press messages button and go to CUE greetings
  
 How to make it work?
  
 Thanks,
  

   
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Re: [OSL | CCIE_Voice] Block calling name

[kapil atrish]

On route-pattern you may set CLID Name/Number to restricted/allowed.

James Key <[EMAIL PROTECTED]> wrote:  Block calling nameWhat is the 
best way to block calling name on certain route patterns, while still allowing 
it on others?  Example: hq local send calling name + number, hq international 
just calling number.
 
 thanks,
 James
  
  
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Re: [OSL | CCIE_Voice] Transcoders required for IPCC Exp

[kapil atrish]

Possible your transcoder is not getting invoked.

You've HQ region set to use G.711 within itself and G.729 with others. 
I believe CTI RP and ports would be in HQ region and IPCC Express is configured 
for G.711 codec. So you get fast-busy. When you change the region settings to 
use G.711 you can make calls successfully.

You may want to recheck xcoder registration, mapping to MRG> MRGL and applied 
to DP or device level.



"Pardeep Singh (pardsing)" <[EMAIL PROTECTED]> wrote: Hello,
  
 I have a question  regarding 6608 xcoder for IPCC.
  
 I have my 6608  Transcoder in a HQ region which does g711 within and g729 to  
others.
  
 One of my phone in  SiteB1 calls the CTI RP for IPCC and gets a fast busy..
  
 If I change my 6608  xcoder to be in all g711 region then the call works fine 
but its a g711 call  from siteb1 phone to ipcc which means I am running g711 
over WAN = no  good.
  
 Can someone shine  some light on the proper configuration needed for this.
  
 Thank you in  advance.
  
  


   

Re: [OSL | CCIE_Voice] Question regarding leaving "Overhead".

[kapil atrish]

Is your question about overhead calculation for voice calls or something else?

My explanation for calculating overhead for voice calls.

Example:
CIR 512,
Allow priority b/w for 5 g.729 calls with FRF.12 (can be MLPPP, FR:
Allow 10% overhead.

I would calculate 5x 27.2kbps = 136kbps 
Add 10% ovehead = 13.6
Total = 149.6 Kbps.
Round-off 150 Kbps

You may already be knowing all this. I want to emphasize on the point that 
overhead is calculated on b/w required for number of calls. I haven't seen any 
scenario which says leave 10% interface/CIR b/w for overheads.




Scott ODonnell <[EMAIL PROTECTED]> wrote: I've been working through several WAN 
Qos scenarios and I keep getting hung up in how to interpret requirements.

On one hand you have the max-reserved-bandwidth command, which is applied at 
the physical interface.
 Then you have the well-established rule of calculating 95% of CIR's for frame 
sub-interfaces.


Given a vague requirement of leaving 10% for overhead, how would you approach 
this?
 Do you raise the max-reserved-bandwidth of the physical interface?
Or do you adjust the calculated CIR/MINCIR, etc.


I know "Ask the proctor" is the obvious answer.

 
Just looking for input.


- Scott


 

   

Re: [OSL | CCIE_Voice] Gatekeeper E164 registration

[kapil atrish]

I ran into this problem number of times. I initially put number  without 
no-reg option under ephone-dn and when integrating it with GK later I simply 
put the command "number  no-reg: and I faced this issue.

I need to do "no number" and number  no-reg to resolve this. Same way for 
ephone-hunt Pilot number, dialplan-pattern number.



"Kumar, Narinder" <[EMAIL PROTECTED]> wrote: RE: [OSL | CCIE_Voice] 
Gatekeeper E164 registrationWhen you do no telephony setup, and 
telephony service again by default the ephones will register back as no reg/no 
reg both won’t be in the config . Do reset  telephony service all or best bet 
is to reload the CME router. No gateway sometime doesn’t fix the problem
  
   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Key
 Sent: Thursday, 13 November 2008 1:27 PM
 To: Greg Miglucci (gmiglucc); ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Gatekeeper E164 registration
 
 
  
  
 Thanks Greg.  Did that (several times actually) also shut down the gatekeeper, 
did a no gateway on cme and did a no telephony setup and started over.  Still 
same issue.  I was very frustrated.   Cost me valuable time and points as I was 
never able to resolve.
 
 -Original Message-
 From: Greg Miglucci (gmiglucc) [mailto:[EMAIL PROTECTED]
 Sent: Wed 11/12/2008 7:10 PM
 To: James Key; ccie_voice@onlinestudylist.com
 Subject: RE: [OSL | CCIE_Voice] Gatekeeper E164 registration
 
 Verify no-reg and then do no gateway gateway on the CME router.
 
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of James Key
 Sent: Wednesday, November 12, 2008 4:45 PM
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Gatekeeper E164 registration
 
 
 
 Had an issue where all my DNs on cme would register to Gatekeeper.  I
 had no-reg defined for each number, and  it still would always register
 those numbers.  Doing a no gateway and then gateway never resolved.  Did
 I miss something somewhere?  I never have run into this issue during my
 studies.  The one thing I didn't do was reboot the gatekeeper router.
 
 
 
 
 
 James
 
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Re: [OSL | CCIE_Voice] IPCC lab Gotcha

[kapil atrish]

You may need to upload the welcome-prompt under  respective language (en_US, 
en, Default). 
   
  Did you validate the script before uploading under CRS Script editor?
   
  You may've run the script in reactive mode and verify whether the call was 
hitting the script or not and which stage it was failing.
   
   
  
Steve ccietester <[EMAIL PROTECTED]> wrote:
Just made another lab attempt recently and result was not good.  One of the 
things that really made me frustrated is IPCC express.  I couldn't get it 
working in either of my two attempts.  When i dialed CTI RP number(from either 
HQ or branch site), I got no sound and then busy signal after a few seconds.  
In both attempts I checked that CTI RP was registered, partitions and CSS were 
properly assigned, transcoder was registered and placed in the right MRGL.  
Everything seemed to be correctly configured but call just couldn't get 
through.  
   
  I felt pretty confident on ipcc express before going into lab as I have been 
practising it in my own lab for at least 30 times and I knew to what to check.  
Failure twice on the same thing is really scary.  So I think maybe there is a 
gotcha in the real lab that I'm not aware of (e.g service parameter)?   Has 
anyone experienced same problem when taking the lab?  What else do I need to 
look?
   
  Thanks a lot!!


   

Re: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from Chile

[kapil atrish]

100% correct. I had the same issue and I had to remove the dialplan pattern and 
use TP under voice-port to meet the requirement of 4 digit CLID to HQ & 10 
digit DID to PSTN . I've made it a practice not to use dialplan pattern. 

However one small confusion when having BACD. For ex Question says: aa-pilot no 
should be 3223000. Now, if you use TP under voice-port and translate all 
incoming calls to 4 digit extension (DID), the your aa-pilot would be 3000 and 
not 3223000. I don't know how would proctor grade that since question asks 
aa-pilot to be 3000. There is a wayaround to use num-exp and expand 3000 to 
3223000 but I never tested this.

May be Christian can share his experience on this as well...


Christian Narvaez <[EMAIL PROTECTED]> wrote:  RE: [OSL | CCIE_Voice] 
Finally !, CCIE Voice #22488 ...from ChileHi Chunmei,
 
 In the real exam it is asked a lot of requirements related with 
transformations of ANI or DNIS for the CME and SRST. My experience in prior 
attempts say that is not recommendable using dial-plan pattern, instead of get 
used to use translation rules/profile and apply then to the dial-peer to 
accomplish each specific question requirement.
 For example imagine it is asked that phones in CME need to be seen as a 4 
digits ANI when calling to HQ or SiteB, in that case you will have problem 
using dial-plan pattern since is force you to send the 10 digits ANI, same for 
access to CUE, or in case is required all international calls from CME present 
ANI with a preempted 9011. In those cases anyway you will need to use some kind 
of translations
 
 In resumen I thinks is more flexible using Translations than dial-plan pattern 
in relation to the kind of questions asked in the exam.
 
 -Original Message-
 From: chunmei chen [mailto:[EMAIL PROTECTED]
 Sent: Sat 11/1/2008 7:18 AM
 To: Christian Narvaez
 Subject: Re: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from Chile
 
 Congra Christian!!  way to go!
  
 Your notes below is very helpful.. however could you explain a little bit more 
on avoiding dialplan pattern?  In what scenario it causes conflict?
  
 I have been using dialplan pattern command since day one never had a problem.
  
 Thanks!
 
 
 --- On Thu, 10/30/08, Christian Narvaez <[EMAIL PROTECTED]> wrote:
 
 From: Christian Narvaez <[EMAIL PROTECTED]>
 Subject: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from Chile
 To: ccie_voice@onlinestudylist.com
 Date: Thursday, October 30, 2008, 8:46 AM
 
 
 
 I am glad to announce this October 27th I finally obtained my CCIE Voice 
#22488.
 I would like to thank all the people of this forum, especially those who shed 
my path when I was lost in some topics. Special thanks to Mark and Vik and the 
team of IPexpert which support this forum
 Below is the detail of my experience and the thoughts I would like to share 
with other candidates.
 
 Some Facts of my preparation
 
 Number of attempts: 4
 Attempt Dates : May 5th, June 26th , August 20th and October 27th  
 Location Center: All in San Jose
 Months of preparations: 12 since passed the written test.
 Hours working on Virtual Rack Sessions: 416 Hours (52 IPExpert Proctorlabs 
session, 8 hours/each)
 Hours working on own lab : aprox 800
 Hours checking written material and forums : aprox 400
 Books Read: 0 ,is not needed if you are not a beginner
 Bootcamps attended : 0 , although it depends of each one, but personally I 
think is costly 
 in relation with the real benefit.
 Forums Consulted: Internetwork Expert (web-based) and IPExpert (email 
distribution), both 
 are good.
 Cost per attempt:  aprox USD 3200 (Exam=USD 1400 , AirTicket from CHILE=USD 
1500, Stay+Transportation+Food=USD 300~600)
 
 
 Strategies Used during my attempts
 --
 Strategy 1) Section-Based Approach, Configuring and Testing the whole section 
before begin 
 the next. I had a predefined amount of max time for each section that I could 
afford to complete the configuration and testing before go on with the next 
section.
 Strategy 2) At the beginning of the test spend max 20 minutes doing the 
strategy3 and a 
 brief read to just some key questions specially the one of the Location&CAC 
Section
 Strategy 3) Cut the large paper sheet given in San Jose into four smaller 
pieces
 a) One of the pieces for the topologic diagram, IP Addresses and  
Numbering Plan
 b) On the second piece, write down each section name and the task numbers.
 b.1)Besides each section name, note the max estimated time when you 
expect finish 
 the section, that is useful to self-control the time you spend specially if a 
problem is faced.
 b.2) Once configured the task mark it with a "check" besides
 b.3) Once tested the task mark it with an "OK" besides.   
 c) On the third piece write down the numbers of the PSTN IP Phone, believe 
me this simple tip 
 saves time when you are testing dial plan and you will not have

Re: [OSL | CCIE_Voice] Gateway Channel selection control ???

[kapil atrish]

When adding CAS circuit in CCM (MGCP) it gives option to enable channels for 
Outbound/Inbound/Bothways. I never tried but won't that work for us.

Secondly, if CAS circuit in H.323 mode you may create multiple DS0 and point 
DPs to respective voice-ports for outbound calls leaving others for inbound.



Paul and Bobs <[EMAIL PROTECTED]> wrote: Thanks for the reply. How can this 
this be achieved. I know from the PABX world , you can control the number of 
outbound and inbound channels and would love to find a way of doing this i the 
Cisco world on either MGCP or H.323. Doesnt matter which one (both would be 
good) but if not then just one of them.
 
Cheers

On Wed, Oct 29, 2008 at 11:02 PM, Mark Snow <[EMAIL PROTECTED]> wrote:
 No because you can't create two pri-group timeslot service mgcp commands 
because then you would have to create two mgcp gateways in CUCM and 4.1.3 won't 
allow you to do this to the same hostname. 
 
Mark SnowSr Technical Instructor
IPexpert, Inc.

Sent from my iPhone




On Oct 28, 2008, at 11:36 PM, "Paul and Bobs" <[EMAIL PROTECTED]> wrote:
 


Thanks Mark

If I wanted to just use mgcp, is there a way to control which channels are 
used. So I can reserve 10 channeles for outgoing and 10 for incoming with a 
total of 20
 
On Wed, Oct 29, 2008 at 2:19 PM, Mark Snow <[EMAIL PROTECTED]> wrote:
  BTW - that's not to say that I recommend it - but for lab purposes should be 
all good.  There could be bugs associated with it - and I would definitely 
check BugNavigator before putting it into production :)


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


Senior Technical Instructor - IPexpert, Inc.


Telephone: +1.810.326.1444
Fax: +1.309.413.4097
  Mailto: [EMAIL PROTECTED]
--
Join our free online support and peer group communities: 
http://www.IPexpert.com/communities
  --
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand 
and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE 
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab 
Certifications.
  --

 


On Oct 28, 2008, at 10:20 PM, Paul and Bobs wrote:



Hi All

I was wandering if anyone know of a way using both MGCP and H.323 to control 
the channells on an E1/T1 circuit. For example - If I have a single E1 service 
with only 20 channels and I want to say reserve 5 for outgoing and reserve 15 
for incoming, is there a way on both protocols to do this.
   
Thanks

Paul








 





 

   

[OSL | CCIE_Voice] Annunciator on unassigned number

[Kapil Atrish]

HI,

Not sure if it has been asked and answered earlier.

Requirement, play CCM annunciator on incoming call from PSTN to any unassigned 
DID.

For ex DID range 200-300. Ext 250 to 300 are not assigned to any device. If 
PSTN calls any of these DIDs, the caller gets fast-busy. Instead of fast-busy I 
want to play CCM annunciator saying something like number not available. It 
works for internal callers without any additional config. 

I tried a TP with Block this parttern, unassigned number etc options but no 
luck.
Tried creating a CTI RP covering the unassigned number. CRI TP remains 
unregistered. Same result, fast-busy to caller. 
I can achieve this by routing calls to Unity and playing required prompt but 
that's not the requirement. Questions says CCM Annunciator needs to be played 
and not Unity greeting.

Thanks for your help..


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Re: [OSL | CCIE_Voice] Where to run IPMA

[kapil atrish]

Isn't the call processing depends upon which CCM (Sub/Pub) the IP Phone/IPMA 
CTI Port is registering to? If Phone/CTI RP are registered to Sub, all calls 
will be processed by Sub even though IPMA points to Pub.

Correct me if I am missing something..

Yung Hung <[EMAIL PROTECTED]> wrote:v\:* {behavior:url(#default#VML);} 
o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape 
{behavior:url(#default#VML);} Juan,
   
  I believe you would use DNS and a DNS host to point to both IPs, that way it 
will use whichever http server that is up and running.
   
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan
 Sent: Tuesday, October 28, 2008 6:46 AM
 To: [EMAIL PROTECTED]
 Cc: Mark Snow; 
 Subject: Re: [OSL | CCIE_Voice] Where to run IPMA
  
   
  Hi Mark, Michael, Bala,
 
  
w.r.t the service URL itself, to have the manager be able to change any 
setting to the service when the primary http server is down - is it an idea to 
create 2 service URLs - a primary IPMA pointing to the primary and a 'backup' 
IPMA pointing to the backup http server? Or is this not done for some reason?
  
 
  
regards,
  
Juan
On Tue, Oct 28, 2008 at 5:41 AM, Balamurugan Singaram <[EMAIL PROTECTED]> 
wrote:
 Hi Mark,
   
   
   
  If Subscriber will be the "Primary Call Processing   Server" then the 
service URL IP Address of IPMA should be only SUB IP   address, not PUB IP 
Address could please let me know ?
   
   
   
  Thanks,
   
  Bala.
   
   --- On Mon, 27/10/08, Mark Snow <[EMAIL PROTECTED]> wrote:
   
  From: Mark Snow <[EMAIL PROTECTED]>
  
   Subject: Re: [OSL | CCIE_Voice] Where to run IPMA
   
   To: "Michael Shavrov" <[EMAIL PROTECTED]>
   Cc: ""   
   Date: Monday, 27 October, 2008, 9:57 PM
  
 All functionality will continue to work with the exception   that if 
the manager went to change any preferences, s/he would have to wait   until the 
http server came back online or phone up the asst and have them do   it.  
   
   Mark Snow 
  Sr Technical Instructor
   
  IPexpert, Inc.
   
   
  Sent from my iPhone
   
   
   
  
   On Oct 27, 2008, at 12:19 PM, "Michael Shavrov" <[EMAIL PROTECTED]>   wrote:
   
Mark,
   
   
   
  And what about IP service?   It's understood, that it's possible to run 
IPMA service and configure both in   Service Parameters. But how manager's 
phone will react on inability to access   URL for the IP service?
   
   
   
  Sincerely,
   
   
   
  Mike
   
   
   
   
   
  - Original Message - 
   
 From: Mark Snow 
   
  To: Kevin Porter 
   
  Cc:
   
  Sent: Monday, October   27, 2008 11:50 AM
   
  Subject: Re: [OSL |   CCIE_Voice] Where to run IPMA
   
   
   
  Point it to both as primary Sub and Backup Pub. IPMA   supports both a 
pri and sec.  
   
   Mark Snow 
  Sr Technical Instructor
   
  IPexpert, Inc.
   
   
  Sent from my iPhone
   
   
   
  
   On Oct 27, 2008, at 11:38 AM, "Kevin Porter" <[EMAIL PROTECTED]>   wrote:
   
In a   scenario where you are told that the Subscriber will be the 
"Primary   Call Processing Server" and the Publisher the "Backup", should   the 
IPMA parameters (Phone Service URL, IPMA Service parameters, etc…) point   to 
the Subscribers IP Address?
   
   Thanks,
   
   Kevin
 
  
   Kevin   Porter
   Systems Engineer L4   
   Netelligent   Corporation
   400 South Woods Mill Drive, Suite 105
   St. Louis , MO 63017 
   Office:   (314) 392-6921
   Cell: (314) 852-1252
   Fax: (314) 392-9760 
   [EMAIL PROTECTED]
   www.netelligent.com
   Bridging The Gap Between Good and GREAT IP Communications! 
   
   


-
  
  Get your preferred Email name! 
 Now you can @ymail.com and @rocketmail.com.
  
   
  
  
  

   

[OSL | CCIE_Voice] DC directory admin password

[Kapil Atrish]

Hi,

I am on POD 15. I am unable to access the DC directory admin page with 
credentianls: Administrator/cisco
Has anybody else faced this issue recently/any new working combination?

thanks for your time

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[OSL | CCIE_Voice] codec sampling rate

[Kapil Atrish]

Does the codec sampling rate need to match at CCM and H.323 GW or whatever 
configured at CCM H.323 GW auto-negotiates?
 
What if different codec sampling rate at CME, does CCM also need to have the 
same sampling rate?
 
Appreciate any comments on this
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[OSL | CCIE_Voice] Aux Vlan on ATA

[Kapil Atrish]

Hi,
 
I checked and found some online docs mentioning that we can use Aux Vlan with 
ATA. Can someone pl confirm if we should configure both Data and Voice Vlans on 
ATA in lab?
 
Also do we need to manually modify the Vlan settings from default: 0x002b?
 
 
Thanks for your time...
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Re: [OSL | CCIE_Voice] debug multicast MOH

[Kapil Atrish]

One quick question: 

Although I've it configured on sub-if and virtual-template, when doing MLPP 
fragmentation, I need to put ip pim-dense mode only on int virtual-template and 
not on the sub-if?



From: [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] debug multicast MOH
Date: Sat, 11 Oct 2008 19:05:13 +0530








No I didn't have ccm-manager  music-on-hold command. I've put it now.

When I do debug ip igmp, I can see message exchange the moment I put the phone 
on hold. Counters also start increasing in "show ip pim interface count" at 
both the routers. Show ip mroute also shows expected output.

I've registered couple of IP Blue sofphones at BR2 deivice pool and one IPC at 
HQ device pool for testing. I am not sure if this scenario is good enough to 
confirm Multicast MOH b'coz none of the phone is across the wan.





> CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> Subject: Re: [OSL | CCIE_Voice] debug multicast MOH
> Date: Fri, 10 Oct 2008 12:28:20 -0700
> 
> Yes it is- ccm-manager music is required for MGCP AND (I repeat AND)  
> H323 gateways.
> 
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
> 
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED]
> 
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities
> IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- 
> Demand and Audio Certification Training Tools for the Cisco CCIE R&S  
> Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
> CCIE Storage Lab Certifications.
> 
> 
> 
> On Oct 10, 2008, at 12:25 PM, Jacob Owen wrote:
> 
> > did you run the "ccm-manager music-on-hold" command on the BR1  
> > router?  I have been told that is required for Multicast MOH even  
> > when you aren't doing MOH from the routers flash.  Hopefully someone  
> > can chime in and confirm.
> 

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[OSL | CCIE_Voice] CUE - no prompt when dialing Pilot no.

[Kapil Atrish]

Hi,

I've configured CUE/CME and I need few phones to test. My softphones fail to 
register with CUE. I've tried wit IP Blue and IPC which keeps on 
registering/unregistering. I've verified loads are present in flash and CME is 
configured as TFTP for all the load files. I am doing manual registration but 
no sucess. 

Any inputs how to fix this. 


Secondly any debugs to confirm whether my transcoder is getting invoked at CME?

When I dial from HQ phone, and the call goes CFNA, I see HQ phone getting 
redirected to cue VM pilot but don't get any message/prompt. Transcoder is 
configured on CME and allow-connections h323 to sip and vice-versa. 
Below are the dial-peers I've configured on CME:

VM Pilot 3111, I am using 10 digit dial-plan pattern under telephony-system. 
8000/8001 are MWI on and Off nos. Phone 1- 3001, phone-2 3002 having mailboxes 
configured.
!
dial-peer voice 11 voip
 destination-pattern 3111
 session protocol sipv2
 session target ipv4:10.3.202.20
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 12 voip
 destination-pattern 3...$
 session protocol sipv2
 session target ipv4:10.3.202.1
 incoming called-number 800.
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 13 voip
 destination-pattern 3313233
 session protocol sipv2
 session target ipv4:10.3.202.20
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 14 voip
 destination-pattern 3313233001   Mailbox 1, ext 3001.
 session protocol sipv2
 session target ipv4:10.3.202.20
 dtmf-relay sip-notify
 codec g711ulaw
 no vad   
!
dial-peer voice 15 voip
 destination-pattern 3313233002>mailbox2, ext 3002
 session protocol sipv2
 session target ipv4:10.3.202.20
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!











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Re: [OSL | CCIE_Voice] debug multicast MOH

[Kapil Atrish]

No I didn't have ccm-manager  music-on-hold command. I've put it now.

When I do debug ip igmp, I can see message exchange the moment I put the phone 
on hold. Counters also start increasing in "show ip pim interface count" at 
both the routers. Show ip mroute also shows expected output.

I've registered couple of IP Blue sofphones at BR2 deivice pool and one IPC at 
HQ device pool for testing. I am not sure if this scenario is good enough to 
confirm Multicast MOH b'coz none of the phone is across the wan.





> CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> Subject: Re: [OSL | CCIE_Voice] debug multicast MOH
> Date: Fri, 10 Oct 2008 12:28:20 -0700
> 
> Yes it is- ccm-manager music is required for MGCP AND (I repeat AND)  
> H323 gateways.
> 
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
> 
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED]
> 
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities
> IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- 
> Demand and Audio Certification Training Tools for the Cisco CCIE R&S  
> Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
> CCIE Storage Lab Certifications.
> 
> 
> 
> On Oct 10, 2008, at 12:25 PM, Jacob Owen wrote:
> 
> > did you run the "ccm-manager music-on-hold" command on the BR1  
> > router?  I have been told that is required for Multicast MOH even  
> > when you aren't doing MOH from the routers flash.  Hopefully someone  
> > can chime in and confirm.
> 

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[OSL | CCIE_Voice] debug multicast MOH

[Kapil Atrish]

Hi,

Can pl let me know any debug commands to confirm multicast MOH reaching BR1 
Router (or Phones if possible).  I checked through perfomance monitor on CCM 
and could see one MOH Multicast resource active when call was put on hold but 
there was no MOH on phone.

I checked "show ip pim interface count" and could see counters increasing on HQ 
router interfaces but not on BR1.

Below is the summary how I configured multicast MOH:
Enabled  MOH Audio resource for multicast and MOH server for multicast with Hop 
count of 6. Codec G.711ulaw an d G.729 selected. Put MOH server in two 
different MRGs, one for BR1 and anther for HQ. BR1 MRG has MOH enabled. Put the 
MRGs in respective MRGLs and applied to correct device pools.
Enabled IGMP snooping on Cat 6k, enabled multicast-routing on HQ and BR1 
routers. Configured IP pim-dense mode on HQ - to BR1 Sub-interface, HQ 
Fast-ethernet sub-interface, BR1 Voice vlan interface.

Thanks for your time..

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Re: [OSL | CCIE_Voice] SIP call fails on G729

[Kapil Atrish]

I don't have hw resources on the router. All I configured is  he enchaned IOS 
MTP which is an IOS feature. AFAN it can only support G711. Which even CCM SW 
MTP does.
 
Since MTP is a must on SIP trunk and I am running g729 which is not supported 
by Software MTPs, I a pretty sure now I need the hw resources.
 
Protocol translation can be done by IPIPGW or even CCM. I'll check Cisco design 
guides also today to confirm.
 
I've excluded CCM MTP from the MRGL bt same result. 
 
Thanks for your time..



Subject: RE: [OSL | CCIE_Voice] SIP call fails on G729Date: Mon, 6 Oct 2008 
09:49:50 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]; 
[EMAIL PROTECTED]; ccie_voice@onlinestudylist.com







  

You would need to ensure that the Hardware transcoder is used first. Do not 
have the software and hardware in the same MRG and the MRG with the software 
must be listed below the hardware MRG
 

Cheers!
 


 
Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions
7601 Ora Glen Drive, Suite 100, Greenbelt, MD  20770 | [EMAIL PROTECTED]
D: 571.225.0132 | www.presidio.com

 



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
ShavrovSent: Monday, October 06, 2008 8:54 AMTo: Edi Hamlet; Kapil Atrish; 
[EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] SIP call fails on G729
 

So. may be we should exclude the software MTP from the MRGL, and keep the only 
hardware MTP/XCoder?


- Original Message - 

From: Edi Hamlet 

To: Kapil Atrish ; [EMAIL PROTECTED] 

Sent: Monday, October 06, 2008 8:04 AM

Subject: Re: [OSL | CCIE_Voice] SIP call fails on G729

 


in order to join 2 different call leg (sip & h323) with g729 codec, i think 
transcoder (MTP hardware)is a must. MTP software can do this but only for g711 
codec. 

 

- Original Message ----From: Kapil Atrish <[EMAIL PROTECTED]>To: Edi Hamlet 
<[EMAIL PROTECTED]>; [EMAIL PROTECTED]: Monday, October 6, 2008 6:57:13 
PMSubject: RE: [OSL | CCIE_Voice] SIP call fails on G729That;s what I am trying 
to find out if I need the Xcoder. I think if I've xcoder the call would work 
even w/o ipip gw.  Is Xcoder must in this scenario?Thanks for your time...



Date: Mon, 6 Oct 2008 04:51:56 -0700From: [EMAIL PROTECTED]: Re: [OSL | 
CCIE_Voice] SIP call fails on G729To: [EMAIL PROTECTED]; 
ccie_voice@onlinestudylist.com


try to terminate your call from CME to IPIPGW (w/transocder) on HQ router, 
don't direct to CCM. so the topology will looks like thisIP Phone -- CME -- 
SIP/H323 Trunk -- IPIPGW -- SIP/H323 trunk -- CCM -- IP Phone

 

- Original Message From: Kapil Atrish <[EMAIL PROTECTED]>To: [EMAIL 
PROTECTED]: Monday, October 6, 2008 6:47:23 PMSubject: [OSL | CCIE_Voice] SIP 
call fails on G729Hi,Following scenario:IP Phone ---CME---sip trunk---CCM--IP 
PhoneUsing g729 call fails and works fine on 711. MTP is selected on trunk, 
infact I've created an IOS enchanced software MTP on a router and given it to 
SIP trunk. Bt that's software only and I understand it would support only G711. 
Tried with CCM SW Mtp but same result.Can pl comment if that's the case why 
calls are failing on 729  or what might be missing.SIP dial-peer 
config:!dial-peer voice 9 voipdestination-pattern [1-2]...$ session protocol 
sipv2 session target ipv4:10.5.0.1dtmf-relay sip-notify codec g711ulaw no 
vadThanks for your time..



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Re: [OSL | CCIE_Voice] SIP call fails on G729

[Kapil Atrish]

That;s what I am trying to find out if I need the Xcoder. I think if
I've xcoder the call would work even w/o ipip gw.  Is Xcoder must in
this scenario?

Thanks for your time...

Date: Mon, 6 Oct 2008 04:51:56 -0700
From: [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] SIP call fails on G729
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com



try to terminate your call from CME to IPIPGW (w/transocder) on HQ router, 
don't direct to CCM. so the topology will looks like this

IP Phone -- CME -- SIP/H323 Trunk -- IPIPGW -- SIP/H323 trunk -- CCM -- IP Phone

- Original Message ----
From: Kapil Atrish <[EMAIL PROTECTED]>
To: ccie_voice@onlinestudylist.com
Sent: Monday, October 6, 2008 6:47:23 PM
Subject: [OSL | CCIE_Voice] SIP call fails on G729






Hi,
Following scenario:


IP Phone ---CME---sip trunk---CCM--IP Phone

Using g729 call fails and works fine on 711. MTP is selected on trunk, infact 
I've created an IOS enchanced software MTP on a router and given it to SIP 
trunk. Bt that's software only and I understand it would support only G711. 
Tried with CCM SW Mtp but same result.

Can pl comment if that's the case why calls are failing on 729  or what might 
be missing.

SIP dial-peer config:
!
dial-peer voice 9 voip
 destination-pattern [1-2]...$
 session protocol sipv2
 session target ipv4:10.5.0.1
 dtmf-relay sip-notify
 codec g711ulaw
 no vad


Thanks for your time..

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[OSL | CCIE_Voice] SIP call fails on G729

[Kapil Atrish]

Hi,
Following scenario:


IP Phone ---CME---sip trunk---CCM--IP Phone

Using g729 call fails and works fine on 711. MTP is selected on trunk, infact 
I've created an IOS enchanced software MTP on a router and given it to SIP 
trunk. Bt that's software only and I understand it would support only G711. 
Tried with CCM SW Mtp but same result.

Can pl comment if that's the case why calls are failing on 729  or what might 
be missing.

SIP dial-peer config:
!
dial-peer voice 9 voip
 destination-pattern [1-2]...$
 session protocol sipv2
 session target ipv4:10.5.0.1
 dtmf-relay sip-notify
 codec g711ulaw
 no vad


Thanks for your time..

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Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt

[Kapil Atrish]

Exactly that was the case. Not it works like charm. 

Thanks a ton..

CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] BACD issue  - No welcome prompt
Date: Sat, 4 Oct 2008 12:52:37 -0700

I know. Please do as I state in previous email. Any call using the loopback 
address in a voip dialpeer will still require gk authorization even though we 
know it is a local call.
 Vik Malhi – CCIE #13890 
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444 
Fax: +1.810.454.0130 
Mailto: [EMAIL PROTECTED]
Join our free online support and peer group communities: 
http://www.IPexpert.com/communitiesIPexpert - The Global Leader in Self-Study, 
Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the 
Cisco CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice 
Lab and CCIE Storage Lab Certifications.
 
On Oct 4, 2008, at 11:43 AM, Kapil Atrish wrote:Its local call from CME phone 
to bacd. No gatekeeper in between.

PH1---CME with AA/ACD---ephone=hunt

CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Date: Sat, 4 Oct 2008 11:36:07 -0700

Do a debug RAS and check if you see an ARJ from the gatekeeper.
Try unregistering the CME from the GK and try.
Any call that uses a VOIP dialpeer will require bandwidth authorization for 
128kbps and if you have a bandwidth cac restriction within the cme zone the 
call will fail.
Vik Malhi – CCIE #13890 
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444 
Fax: +1.810.454.0130 
Mailto: [EMAIL PROTECTED]
Join our free online support and peer group communities: 
http://www.IPexpert.com/communitiesIPexpert - The Global Leader in Self-Study, 
Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the 
Cisco CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice 
Lab and CCIE Storage Lab Certifications.

On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote:The attached file has full 
config and debug output if you wish to see.


!
dial-peer voice 15 voip
 destination-pattern 3700
 session target ipv4:172.22.102.1   
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 16 voip
 service aa
 incoming called-number 3700
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!

Thanks for your time

Subject: RE: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Date: Sat, 4 Oct 2008 12:39:38 -0400
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com

 Can you send your dial-peer for the BACDapplication?  Scott Hardesty | Cisco 
Engineer | MidAtlantic | Presidio Networked Solutions7601 Ora Glen Drive, Suite 
100, Greenbelt, MD  20770 | [EMAIL PROTECTED]: 301.313.2041 | C: 443.789.1219 | 
www.presidio.com From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil 
Atrish
Sent: Saturday, October 04, 20087:58 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] BACDissue - No welcome prompt HI,


Attached is my config. I get fast busy tone and Unknown number on display whenI 
dial the pilot number from any CME phone. I can dial hunt-pilot directly 
andcall get routed correctly or give the aa-pilot to hunt-pilot and ring 
thephones fine. Call in between phones are setup using G711ulaw. I've tried 
singlevoip dial-peer with incoming called-address and destination-pattern, 
reload ofrouter, re-configure script.

Below is the snapshot of bacd config and debug voice application seesion..




application
 service queue flash:app-b-acd-2.1.0.0.tcl
  param queue-len 15
  param aa-hunt5 3701
  param queue-manager-debugs 1
  param number-of-hunt-grps 2
 !
 service aa flash:app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 1
  param number-of-hunt-grps 2
  param menu-timeout 6
  param handoff-string aa
  param dial-by-extension-option 4
  paramspace english language en
  param max-time-vm-retry 2
  param max-extension-length 4
  param aa-pilot 3700
  paramspace english location flash:
  param second-greeting-time 30
  param welcome-prompt _bacd_welcome.au
  param queue-manager-debugs 1
  param call-retry-timer 15
  param max-time-call-retry 600
  param voice-mail 3005
  paramspace english prefix en
  param service-name queue
 !
!
!


BR2#dir flash:
Directory of flash:/

1  -rw-  24679 app-b-acd-2.1.0.0.tcl
2  -rw-  33870 app-b-acd-aa-2.1.0.0.tcl
3  -rw-  75650 en_bacd_allagentsbusy.au
4  -rw-  83291 en_bacd_disconnect.au
5  -rw-  63055 en_bacd_enter_dest.au
6  -rw-  37952 en_bacd_invalidoption.au
7  -rw- 496521 en_bacd_music_on_hold.au
8  -rw- 123446 en_bacd_options_menu.au
9  -r

Re: [OSL | CCIE_Voice] H323 COR ?

[kapil atrish]

When not in SRST mode, all layer-3 information (DNIS, ANI) are back-hauled to 
CCM directly and COR won't trigger.

Jacob Owen <[EMAIL PROTECTED]> wrote: Mike,
I was under the impression since the call came into the H323 gateway from UCM 
(GW isn't in SRST) it wasn't "tagged" with an incoming corlist and therefore 
could reach all remote PSTN numbers.  When the router drops back to SRST the 
phones would register with a corlist incoming and therefore be limited to where 
they could call.  Hopefully someone will let me know if I am incorrect.  You 
could also test this by adding a corlist incoming to the inbound voip dial-peer 
and see if you can call.  
 
On Sun, Oct 5, 2008 at 12:35 PM, Mike Brooks <[EMAIL PROTECTED]> wrote:
 If COR is configured on H323 dial-peers on an H323 gateway, is the
 dial-peer COR only in affect when in SRST mode ?  If not, wouldn't you
 be performing COR twice  once on the CallManager and also on the
 H323-GW ?
 
 for example:
 phones/CSS > h323-gw inbound voip dial-peer (KEY) --->  h323gw
 outbound pots dial-peer (LOCK)
 or
 h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
 dial-peer (LOCK) --> h323-gw/CSS (on CM)
 
 If COR is in affect regardless of if it the site is in SRST mode
 (which I assume it would be) should you just not configure COR
 (keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?
 
 Regards,
 
 Mike Brooks
 CCIE# 16027 (R&S)
 



-- 
Jacob Owen
CCIE #14063 (R&S, Service Provider), CCDP, CCVP
 
 

   

[OSL | CCIE_Voice] Cat6K T1 Ports fail to register

[Kapil Atrish]

HI,

I am getting following when trying to register Cat6K port to CCM on Pod 20. 
I've tried enabling/disabling the ports but all three ports (T1, Xcode and CFB) 
are in same state. Reset the DHCP service, other devices are able to take IP 
Address from DHCP and enough IPs are available in the scope. Can't clear CDP 
table due to insufficient privileges.


ort  Name Status Vlan   Duplex Speed   Type
-  -- -- -- --- 
 7/4  POD20-PSTN-T1enabled400  full - unknown

Port DHCPMAC-Address   IP-Address  Subnet-Mask
 --- - --- ---
 7/4 enable  00-d0-c0-d3-12-c3 (Failed to obtain port interface information)

Appreciate any help...


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Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt

[Kapil Atrish]

Its local call from CME phone to bacd. No gatekeeper in between.

PH1---CME with AA/ACD---ephone=hunt

CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] BACD issue  - No welcome prompt
Date: Sat, 4 Oct 2008 11:36:07 -0700

Do a debug RAS and check if you see an ARJ from the gatekeeper.
Try unregistering the CME from the GK and try.
Any call that uses a VOIP dialpeer will require bandwidth authorization for 
128kbps and if you have a bandwidth cac restriction within the cme zone the 
call will fail.
 Vik Malhi – CCIE #13890 
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444 
Fax: +1.810.454.0130 
Mailto: [EMAIL PROTECTED]
Join our free online support and peer group communities: 
http://www.IPexpert.com/communitiesIPexpert - The Global Leader in Self-Study, 
Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the 
Cisco CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice 
Lab and CCIE Storage Lab Certifications.
 
On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote:The attached file has full 
config and debug output if you wish to see.


!
dial-peer voice 15 voip
 destination-pattern 3700
 session target ipv4:172.22.102.1   
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 16 voip
 service aa
 incoming called-number 3700
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!

Thanks for your time

Subject: RE: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Date: Sat, 4 Oct 2008 12:39:38 -0400
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com

 Can you send your dial-peer for the BACDapplication?  Scott Hardesty | Cisco 
Engineer | MidAtlantic | Presidio Networked Solutions7601 Ora Glen Drive, Suite 
100, Greenbelt, MD  20770 | [EMAIL PROTECTED]: 301.313.2041 | C: 443.789.1219 | 
www.presidio.com From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil 
Atrish
Sent: Saturday, October 04, 20087:58 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] BACDissue - No welcome prompt HI,


Attached is my config. I get fast busy tone and Unknown number on display whenI 
dial the pilot number from any CME phone. I can dial hunt-pilot directly 
andcall get routed correctly or give the aa-pilot to hunt-pilot and ring 
thephones fine. Call in between phones are setup using G711ulaw. I've tried 
singlevoip dial-peer with incoming called-address and destination-pattern, 
reload ofrouter, re-configure script.

Below is the snapshot of bacd config and debug voice application seesion..




application
 service queue flash:app-b-acd-2.1.0.0.tcl
  param queue-len 15
  param aa-hunt5 3701
  param queue-manager-debugs 1
  param number-of-hunt-grps 2
 !
 service aa flash:app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 1
  param number-of-hunt-grps 2
  param menu-timeout 6
  param handoff-string aa
  param dial-by-extension-option 4
  paramspace english language en
  param max-time-vm-retry 2
  param max-extension-length 4
  param aa-pilot 3700
  paramspace english location flash:
  param second-greeting-time 30
  param welcome-prompt _bacd_welcome.au
  param queue-manager-debugs 1
  param call-retry-timer 15
  param max-time-call-retry 600
  param voice-mail 3005
  paramspace english prefix en
  param service-name queue
 !
!
!


BR2#dir flash:
Directory of flash:/

1  -rw-  24679 app-b-acd-2.1.0.0.tcl
2  -rw-  33870 app-b-acd-aa-2.1.0.0.tcl
3  -rw-  75650 en_bacd_allagentsbusy.au
4  -rw-  83291 en_bacd_disconnect.au
5  -rw-  63055 en_bacd_enter_dest.au
6  -rw-  37952 en_bacd_invalidoption.au
7  -rw- 496521 en_bacd_music_on_hold.au
8  -rw- 123446 en_bacd_options_menu.au
9  -rw-  42978 en_bacd_welcome.au
   10  -rw-  496521  Mar 012002 01:13:09 +00:01  music-on-hold_3db.au
   11  -rw-  496521  Mar 012002 02:47:07 +00:01  music-on-hold.au

536870908 bytes total (534895700 bytes free)
BR2#
BR2#


There is no output when I do debug voice application script 

OUTPUT OF 'debug voice application session' is as below. 
Calling no: 3002, called no: 3700

BR2#debug voice app
BR2#debug voice application sess
voip application session debugging is on
BR2#
Mar  1 03:17:40: //37//AFW_:/Closing_AnyEvent:  
Mar  1 03:17:40: //37//AFW_:/Session_Cleaner:  
Mar  1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler:  
Mar  1 03:17:40: 
//37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler:Received event 
CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop
Mar  1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate:  
Mar  1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: 
MOD[Session_65BFE164_0_1046004]( )
Mar  1 03:17:42: //-1//AFW_:/C_ServiceSession_E