[OSL | CCIE_Voice] CIPC strange behavior !

[jonn cozak]

Hi all. I am using CUCM 6. CIPC is installed on my pc. I am using single 
callmanager server that is also my tftp server for endpoints. Now while doing 
my studies i wanted to check how DNS might cause issues. In System->Server, i 
am using hostname instead of IP. Now what happens is that, CIPC after getting 
the .cnf.xml file, registers with tftp server successfully (which is also my 
callmanager server). Now what i read in student guide was, ip phone should not 
be able to register if its not able to resolve the hostname through DNS. (i am 
not using any DNS server and nor the entry for the hostname is present in my pc 
host file).

Can someone tell me why is this the case ? i have searched alot but i couldnt 
find any thing stating that ip phone will fall back to tftp server in case 
primary callmanager fails !!

Any input on this pls ?



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

[OSL | CCIE_Voice] CME busy-trigger-per-button

[Aman Arora]

Hey Folks



Is it possible to set different busy-trigger-per-button for each button (line) 
on a phone on CME.

For example :



If I have line 1 : 1000

And line 2 : 1001



I need to limit 4 incoming calls on line 1 and limit 2 incoming calls on line 2.



How can I achieve this. I guess busy-trigger-per-button sets limits for all the 
buttons on the particular ephone.



Thanks

Aman

___
For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] CME busy-trigger-per-button

[Aman Arora]

 1 dn 1
>  dtmf-relay rtp-nte
>  username 3002 password cisco
> !
> voice register pool  2
>  id mac 001F.6C7E.D6FE
>  type 7941
>  number 1 dn 2
>  dtmf-relay rtp-nte
>  username 3003 password cisco
>
>
>
>
>
> dial-peer voice 200 voip
>  max-conn 1
>  destination-pattern 3600
>  session protocol sipv2
>  session target ipv4:10.10.210.13
>  dtmf-relay rtp-nte
>  codec g711ulaw
> !
> !
>
>
>
> telephony-service
>   no auto-reg-ephone
>  em logout 0:0 0:0 0:0
>  max-ephones 8
>  max-dn 8
>  ip source-address 10.10.202.1 port 2000
>  voicemail 3600
>  mwi relay
>  max-conferences 8 gain -6
>  transfer-system full-consult
>  transfer-pattern .T
>  create cnf-files version-stamp 7960 Mar 10 2010 15:22:39
> !
> !
> ephone-dn  1  dual-line
>  number 3001 no-reg primary
>  label Br2 pHone 1
>  name Br2 Phone 1
>  call-forward busy 3600
>  call-forward noan 3600 timeout 12
> !
> !
>
> sip-ua
>  mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp
> unsolicited
>
> !
> !
> ephone  1
>  device-security-mode none
>  mac-address 001E.EC15.996D
>  type CIPC
>  button  1:1
> !
>
>
>
> Thanks for the anticipated support
>
>
>
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Message: 2
Date: Fri, 12 Mar 2010 14:33:41 -0800
From: Jeff Cotter 
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
To: Omotayo , Otto Sanchez 
Cc: "ccie_voice@onlinestudylist.com" 
Message-ID: <54cc1bd3093b6e41b86926c1657432f187a06...@ssfex1>
Content-Type: text/plain; charset="us-ascii"

FYI, I was only able to get this to work using transcoder on CME.  Had to match 
the codec between UCM trunk and incoming dial-peer on CME...then xcoder would 
engage on CME for the SIP phone.  I have a hardware limitation in my home lab 
so I am not able to configure a xcoder on both UCM and CME simultaneously.




From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 6:33 AM
To: Otto Sanchez
Cc: Jeff Cotter; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

Hello Otto,

i had same issue

The transcoder can be on the trunk?

When i did the transcoder on the br2 router, i get a busy tone when the sip 
phone is being called from the hq phone

REgards
On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez 
mailto:o...@ipexpert.com>> wrote:
Hi Jeff,

Would you please tell us more about the call flow and the end to end codec 
requirements for this call. If doing g.729 over the wan, and your sip phone is 
using g.711 you should transcode at br2,

Please let us know,

Thanks,
On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter 
mailto:jcot...@voxns.com>> wrote:
Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM.  
Can't seem to get a call from Call Manager to CME sip phone working.  I can 
call from CME to UCM but not the other way around. Rings but disconnects when 
answered.  Transcoder shows registered in Call manager.  Thanks


Jeff

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com<http://www.ipexpert.com/>



--
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com<http://www.ipexpert.com/>

___
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Message: 3
Date: Sat, 13 Mar 2010 02:04:50 +0100
From: Omotayo 
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
To: Jeff Cotter 
Cc: "ccie_voice@onlinestudylist.com" 
Message-ID:
<3082f9d41003121704j6e368e5egc989092a7343e...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"

Hello Jeff,

All calls worked when i configure the xcoder on the cme

The question says use the hq router resources- that is where i have issues

thanks

On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter  wrote:

>  FYI, I was only able to get this to work using transcoder on CME.  Had to
> match the codec between UCM trunk and incoming dial-peer on CME?then xcoder
> would engage on CME for the SIP phone.  I have a hardware limitation in my
> home lab so I am not able to configure a xcoder on both UCM and CME
> simultaneously.
>
>
>
>
>
>
>
>
>
> *From:* Omotayo [mailto:adef

Re: [OSL | CCIE_Voice] MVA

[anupam TYAGI]

i have the route pattern partion assigned to the CSS and this CSS  is
assigned to RDP >but still the call disconnect when i dial the external
number in MVZ

On Fri, Mar 12, 2010 at 11:06 PM, Omotayo  wrote:

> Hello,
>
> Berry is right.
>
> create a partition called pt-mva
>
> crease a CSS called css-mva
>
> put the partition in the css
>
> create a route pattern like 9.011! in partition pt-mva. the gateway can be
> the hq gateway if you wish
> discard predot
>
> assign the css to the remote destination profile
>
> this will work for you
>
>
>
>
> On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI  wrote:
>
>> if i dial that external number without MVA it goes through ,but when in
>> MVA i get a disconnect when calling this external number ( so don't seems to
>> be codec issue )
>>
>>
>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer 
>> wrote:
>>
>>> are you maybe calling to a remote location (g.729) and therefore a xcoder
>>> is required, but not set up correctly?
>>>
>>> 2010/3/12 anupam TYAGI 
>>>
  i saw the call hit the gateway .  RDP is having the same CSS as phone
 CSS



 On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
 mjbe...@krollontrack.com> wrote:

>   Check the CSS on the remote destination profile you’re calling from.
>
> If you do a “debug isdn q931” on the PSTN gateway, do you see the call
> hit the gateway?
>
>
>
> Your rerouting CSS on the RDP is used for calls out to your RD.
>
> Your CSS on the RDP is used for calls through MVA that are routed out
> through your PSTN gateway.
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
> *Sent:* Friday, March 12, 2010 9:27 AM
> *To:* ccie_voice-requ...@onlinestudylist.com;
> ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MVA
>
>
>
> Hi Folks
>
> I am doing MVA , When i dial the MVA number ,  I am able to hear the
> prompt.  I dial a  PSTN number , but the call disconnect . Can any body
> suggest me what can be the reason .
>
>
> Rgds
> Anu.
>


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

Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[Omotayo]

Hello Jeff,

All calls worked when i configure the xcoder on the cme

The question says use the hq router resources- that is where i have issues

thanks

On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter  wrote:

>  FYI, I was only able to get this to work using transcoder on CME.  Had to
> match the codec between UCM trunk and incoming dial-peer on CME…then xcoder
> would engage on CME for the SIP phone.  I have a hardware limitation in my
> home lab so I am not able to configure a xcoder on both UCM and CME
> simultaneously.
>
>
>
>
>
>
>
>
>
> *From:* Omotayo [mailto:adefilabi...@gmail.com]
> *Sent:* Friday, March 12, 2010 6:33 AM
> *To:* Otto Sanchez
> *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com
>
> *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
>
>
>
> Hello Otto,
>
>
>
> i had same issue
>
>
>
> The transcoder can be on the trunk?
>
>
>
> When i did the transcoder on the br2 router, i get a busy tone when the sip
> phone is being called from the hq phone
>
>
>
> REgards
>
> On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez  wrote:
>
> Hi Jeff,
>
> Would you please tell us more about the call flow and the end to end codec
> requirements for this call. If doing g.729 over the wan, and your sip phone
> is using g.711 you should transcode at br2,
>
> Please let us know,
>
> Thanks,
>
> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>
>   Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder
> on UCM.  Can’t seem to get a call from Call Manager to CME sip phone
> working.  I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks
>
>
>
>
>
> Jeff
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com 
>
>
> ___
> 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] SIP Hardware Transcoder

[Jeff Cotter]

FYI, I was only able to get this to work using transcoder on CME.  Had to match 
the codec between UCM trunk and incoming dial-peer on CME...then xcoder would 
engage on CME for the SIP phone.  I have a hardware limitation in my home lab 
so I am not able to configure a xcoder on both UCM and CME simultaneously.




From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 6:33 AM
To: Otto Sanchez
Cc: Jeff Cotter; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

Hello Otto,

i had same issue

The transcoder can be on the trunk?

When i did the transcoder on the br2 router, i get a busy tone when the sip 
phone is being called from the hq phone

REgards
On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez 
mailto:o...@ipexpert.com>> wrote:
Hi Jeff,

Would you please tell us more about the call flow and the end to end codec 
requirements for this call. If doing g.729 over the wan, and your sip phone is 
using g.711 you should transcode at br2,

Please let us know,

Thanks,
On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter 
mailto:jcot...@voxns.com>> wrote:
Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM.  
Can't seem to get a call from Call Manager to CME sip phone working.  I can 
call from CME to UCM but not the other way around. Rings but disconnects when 
answered.  Transcoder shows registered in Call manager.  Thanks


Jeff

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



--
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com

___
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] UC and cme sip integration

[Omotayo]

Hello,

it work ok now

I was using the wrong ip address on the unity connection all the while

Thanks

On Fri, Mar 12, 2010 at 8:30 AM, Flemming Ortvald  wrote:

>  Unity connection can do both g729 and g711, you can use “voice class
> codec” on “voice register dn” to expand codec support for sip.
>
>
>
> Med venlig hilsen
>
> Flemming Ortvald
> Network System Eng.
> NetDesign A/S
> +45 4435 8346
>
> Tænk på miljøet inden udskrivning af denne e-post og tilknyttede
> vedhæftninger
>
>
> *From:* Omotayo [mailto:adefilabi...@gmail.com]
> *Sent:* 11 March, 2010 20:58
> *To:* Flemming Ortvald
>
> *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration
>
>
>
>
> Hello,
>
>
>
> I have to configure a transcoder on the br2 router?
>
>
>
> Unity connection support g729 only?
>
>
>
> Rgd
>
> On Thu, Mar 11, 2010 at 8:24 PM, Flemming Ortvald  wrote:
>
> You will need a transcoder or chnage the sip endpoints to support g.711,
> natively it only supports g.729
>
>
>
> Best regards
>
> Flemming Ortvald
> Network System Eng.
> NetDesign A/S
> +45 4435 8346
>
> Tænk på miljøet inden udskrivning af denne e-post og tilknyttede
> vedhæftninger
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo
> *Sent:* 11 March, 2010 20:07
> *To:* OSL Group
> *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration
>
>
>
> Hello all,
>
>
>
> As anyone been able to get the SIP integration between Unity Connection and
> Cme to work? I followed the Proctorlabs Guide
>
>
>
> I posted this sometime lat week and revised as advised but keep getting a
> reorder tone( Number Unknown) when the message button is pressed
>
> Below is the relevant configuration
>
>
>
> voice service voip
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>  no supplementary-service sip moved-temporarily
>  no supplementary-service sip refer
>  sip
>   bind control source-interface Loopback0
>   bind media source-interface Loopback0
>   registrar server expires max 600 min 60
>
>
>
>
>
>
>
> voice register global
>  mode cme
>  source-address 10.10.110.3 port 5060
>  max-dn 3
>  max-pool 6
>  authenticate register
>  mwi reg-e164
>  voicemail 3600
>  tftp-path flash:
>  create profile sync 0006855418337003
> !
> voice register dn  1
>  number 3002
>  call-forward b2bua busy 3600
>  call-forward b2bua mailbox 3002
>  call-forward b2bua noan 3600 timeout 12
>  name br2 phone 2
>  no-reg
>  label br2 phone 2
>  mwi
> !
> voice register dn  2
>  number 3003
>  call-forward b2bua busy 3600
>  call-forward b2bua mailbox 3003
>  call-forward b2bua noan 3600 timeout 12
>  name br2 phone 3
>  no-reg
>  label br2 phone 3
>  mwi
> !
> voice register pool  1
>  id mac ..
>  type 7941
>  number 1 dn 1
>  dtmf-relay rtp-nte
>  username 3002 password cisco
> !
> voice register pool  2
>  id mac 001F.6C7E.D6FE
>  type 7941
>  number 1 dn 2
>  dtmf-relay rtp-nte
>  username 3003 password cisco
>
>
>
>
>
> dial-peer voice 200 voip
>  max-conn 1
>  destination-pattern 3600
>  session protocol sipv2
>  session target ipv4:10.10.210.13
>  dtmf-relay rtp-nte
>  codec g711ulaw
> !
> !
>
>
>
> telephony-service
>   no auto-reg-ephone
>  em logout 0:0 0:0 0:0
>  max-ephones 8
>  max-dn 8
>  ip source-address 10.10.202.1 port 2000
>  voicemail 3600
>  mwi relay
>  max-conferences 8 gain -6
>  transfer-system full-consult
>  transfer-pattern .T
>  create cnf-files version-stamp 7960 Mar 10 2010 15:22:39
> !
> !
> ephone-dn  1  dual-line
>  number 3001 no-reg primary
>  label Br2 pHone 1
>  name Br2 Phone 1
>  call-forward busy 3600
>  call-forward noan 3600 timeout 12
> !
> !
>
> sip-ua
>  mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp
> unsolicited
>
> !
> !
> ephone  1
>  device-security-mode none
>  mac-address 001E.EC15.996D
>  type CIPC
>  button  1:1
> !
>
>
>
> Thanks for the anticipated support
>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] IPCCx 7.x

[Thomas Koch]

Jason,

Thanks…I have our sales team trying get me a copy…

 

Thomas J Koch

Unified Communicatons Consultant 

CCDA, CCNA, CCVP

Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

 

From: Jason Granat [mailto:j...@slash128.com] 
Sent: Friday, March 12, 2010 3:31 PM
To: Thomas Koch
Cc: wilson.sam...@usc-bt.com; vpree...@cisco.com; adefilabi...@gmail.com; 
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

 

I haven't tried that route. If you get the NFR kit it comes with the OS discs. 

Sent while mobile.


On Mar 12, 2010, at 13:22, "Thomas Koch"  wrote:

No…Just WIN2k3 enterprise…

 

Thomas J Koch

Unified Communicatons Consultant 

CCDA, CCNA, CCVP

Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

 

From: Jason Granat [mailto:j...@slash128.com] 
Sent: Friday, March 12, 2010 3:05 PM
To: wilson.sam...@usc-bt.com; koch1...@comcast.net; vpree...@cisco.com; 
adefilabi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

 

Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 
proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. 
After the OS is loaded and rebooted it will start the add new hardware wizard. 
Don’t click that. Just run the install VM Ware tools. After the tools are 
installed the drivers get updated and the wizard goes away. Then reboot again. 
Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR 
disc. That’s it… You could install VNC if you want instead of managing from the 
VMWare console. I’m sure other VM settings combos might work but this has 
always worked for me.

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
wilson.sam...@usc-bt.com
Sent: Friday, March 12, 2010 12:41 PM
To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

 

Hi All,

 

I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if 
anyone has done it, please could you do me a favor by sharing your secret?

 

Regards

Wilson Samuel

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch
Sent: Friday, March 12, 2010 2:49 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

 

Yes…of course

Thanks…

 

Thomas J Koch

Consultant 

CCDA, CCNA, CCVP

Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

 

From: Vishal Preenja [mailto:vpree...@cisco.com] 
Sent: Friday, March 12, 2010 1:39 PM
To: 'Thomas Koch'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Hi Thomas,

 

   Are ladies also allowed to reply? J

 

Pls check the hardware compatibility matrix below:

 

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf

 

HP DL380 G3 is not supported with IPCC 7.x onwards.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Thomas Koch [mailto:koch1...@comcast.net] 
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Gent’s,

Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?

I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said “non 
Cisco hardware” install will now stop”

T 

 

E-mail:thomas.k...@compucom.com

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Yes, I am referring to MRG that is there in the MRGL of the trunk.

 

Just check that you should not have a MTP allocated for that call from CCM to 
CME.

You can verify by making a call and then check in RTMT that whether Transcoder 
is being invoked or MTP.

 

Thanks and Regards,
Vishal Preenja
 

 

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello,

 

Also what do you mean by MTP above transcoder. Are you reffering to the MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or get 
me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, 

Re: [OSL | CCIE_Voice] IPad Support

[Tanner Ezell]

There is a universal rule for sales.

If the content is worth it, they will buy it (or steal it).

I suspect the iPad will be similar in success to the iPhone, but for a
different (and perhaps less general) audience.

Those who think the iPad is just a "large iPod Touch" are mistaken,
and the differences will soon become apparent in the content available
to it.

People bah'd at Steve Jobs calling the iPad a revolutionary device. A
very bold claim, but I believe in a big way he is correct.

For instance, take a look at the following link: http://bit.ly/d9W8fJ

That is only the beginning. Consider the impact this kind of
technology will have on education. The way we consume content will
forever be changed.

As another note, the only people who seem to have claimed the iPad is
'big', 'bulky', or 'awkward' are those whom have not laid a finger on
the device. Those who have would seem to disagree. But with all
things, it comes down to personal preference and (to some extent) how
much you love or hate the company.

Lastly, despite not being best suited to the purpose, the iPad will be
a content creation device. In some ways it may excel where others
fail, and will certainly lack where others are successful (having a
keyboard, for instance). But the device itself is not limited, and is
considerably more capable than people give it credit.

As a last bit regarding the pricing. Yes, that's a shame. But remember
that pricing is set by the publisher, not Apple. However, as the
device is yet to be released we cannot confirm pricing (despite what
was seen in the iPad commercial). And lastly, Apple is not a stupid
business, they know that when presented with the same product at two
different price points, the consumer will chose the cheaper, obviously
this is not in Apple's best interest as a company. ZDnet
(http://blogs.zdnet.com/BTL/?p=30943) seems to think Apple will put a
price cap on the cost of eBooks to remain competitive to their
Amazonian counterparts. It'll be interesting to see how this pans out
but I can only see this benefitting the consumer.

$0.02

Cheers,

Tanner Ezell

On Fri, Mar 12, 2010 at 4:14 PM,   wrote:
> Personally, I wouldn't prefer to buy Apple products (sorry mates, after the 
> ebook price debate, I have lost respect for Mr. Jobs and APPL in general).
>
> I would prefer them to develop it for Kindle and Sony E-Readers (I guess 
> these at the moment have the max. marketshare) also iPad comes at a very high 
> price and hasn't been even launched, though the demand seems to be looming.
>
> I'm not sure if people have around 600 USD to buy E-readers just for the sake 
> of a reader and some computing, esp. when Mac OS challenged people like me 
> have remained loyal to Linux and Windows OS.
>
> Again, it's the survival of the fittest, May the Best Product Win!!
>
> Regards
> Wilson Samuel
>
> -Original Message-
> From: Tanner Ezell [mailto:tanner.ez...@gmail.com]
> Sent: Friday, March 12, 2010 4:10 PM
> To: Samuel, Wilson
> Cc: bontacommunicati...@gmail.com; ccie_voice-boun...@onlinestudylist.com; 
> ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] IPad Support
>
> I challenge IP Expert to build interactive learning content for the
> iPad (as many publishers are now looking into doing).
>
> If done properly, would give IPX an incredible advantage over the competition.
>
> As they say, Evolve or Die. :)
>
> On Fri, Mar 12, 2010 at 3:56 PM,   wrote:
>> IMHO,
>>
>> The best way to read PDFs is to read on a "computer" (aka Laptop, Palmtop, 
>> Tablet PC or Mac versions of the equivalent) and find yourself the most 
>> portable one that suits you.
>>
>> I guess the E-Readers (Kindle, Sony et al) are not that user friendly when 
>> it comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not 
>> really getting benefitted out of it.
>>
>> However I'm keeping the Kindle because, all the newspaper and magazine 
>> subscriptions are a bit ok deal and its really portable, also most of the 
>> books are so far cheaper on Kindle edition (excluding the Cisco Press).
>>
>> HTH
>>
>> Kind Regards
>> Wilson Samuel
>>
>> -Original Message-
>> From: ccie_voice-boun...@onlinestudylist.com 
>> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
>> bontacommunications
>> Sent: Friday, March 12, 2010 9:42 AM
>> To: ccie_voice-boun...@onlinestudylist.com
>> Cc: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] IPad Support
>>
>> I want to "green up" my library. I am considering purchase of an IPad. Does 
>> anyone know if the IPExpert PDF's security will be supported VIA IPad? If 
>> not currently supported, is there a roadmap to get to that point?
>>
>>
>> thanks
>> Chris
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> ___
>> For more information regarding industry leading CCIE Lab training, ple

Re: [OSL | CCIE_Voice] Pt2. Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO)

[Brian Mulgrew]

Hi Mathew - excellent post, and looks like you have successfully answered your 
own question!

 

I'm going through the CUCM normalization study section (again) and i would 
agreed that we set the Called Party Number type at the exiting PSTN gateway 
using called party xform pattern as:

- The call may take multiple routes, if we apply at the xlate pattern, the 
number type may need to be changed again before the call exits to the PSTN.

- RP / RL is still not close enough to the actual PSTN gateway (especially when 
using Loc RG) to avoid the type to be changed again.

- Called Party xform pattern is set at the PSTN gateway so no risk of 
additional type changes before exiting to PSTN.  We could also apply this at DP 
level for phones and gateways (as the phones will ignore called party xform 
rules).

 

Just my opinion - but always open to ideas!

 

thks

b

 


 


From: mjbe...@krollontrack.com
To: ccie_voice@onlinestudylist.com
Date: Fri, 12 Mar 2010 09:22:08 -0600
Subject: [OSL | CCIE_Voice] Pt2. Globalized + Dialing - Applying Called Party 
Number Type for Secondary Route Scenario (non-TEHO)





I could also simplify my translation patterns and make them available to ALL 
sites.
By creating a separate partition called PT_US_Translate and setting up the 
following patterns:
9.!à   Predot, Prefix +1  à 
  Result: +19525163748 (local MN)
9.1!à   Predot, Prefix +à   
Result: +1615444 (remote long-distance in TN)
9011.! à   Predot, Prefix +à
   Result: +3432141861 (remote international)
 
 


From: Berry, Matthew J. 
Sent: Friday, March 12, 2010 9:12 AM
To: OSL Group
Subject: Globalized + Dialing - Applying Called Party Number Type for Secondary 
Route Scenario (non-TEHO)
 
All –
 
I am setting up + dialing on a self-made lab.  A question has come up as to 
where the Called Party Number Type should be set.  For this exercise, I want to 
find the best way to route calls through a system, utilizing alternate paths 
for failover scenarios.  Those this does not take TEHO into account, I want a 
format that can easily accommodate TEHO situations.  I believe my method below 
will do that.
 
PSTN is expecting:
Subscriber: Seven digits, Subscriber
National: Eleven digits (incl. 1), National
Intl: Undefined digits, Intl
 
I have also setup translation patterns in PT_US_MN_EP_PSTN setup as:
9.952[2-9]XXà   Predot, Prefix +1   
   à   Result: +19525163748 (local MN)
9.1[2-9]XX[2-9]XX à   Predot, Prefix +  
  à   Result: +1615444 (remote long-distance in TN)
9.011!à   Predot, 
Prefix +à   Result: +3432141861 (remote international)
 
I have route patterns in PT_US_MN_EP_PSTN setup as:
\+1952[2-9]XXà   +19525163748
\+1[2-9]XX[2-9]XXà   +1615444
\+!  à   
+3432141861
\+!#   à   
+3432141861
Note: I am not using predot in the route patterns!!
 
At this point, all dialed numbers have been globalized from their localized 
variants.
 
I am trying to figure out if it would be best to apply the Called Party Number 
Type (Sub, Nat, Intl) at the Translation Pattern, Route Pattern, Route List 
(i.e. Route Group settings), or Called Party Transformation on the gateway 
level.  If I did not modify the Called Party Number Type at the Translation 
Pattern or Route Pattern, it would allow me to configure different call 
treatment at the route list level for TEHO.  However, I think it would be best 
to apply that setting at the Called Party Transformation pattern on the MGCP 
gateways (avoiding H.323 for sake of conversation).
 
I have two locations that calls will be sent out, MN and TN.  For example:
 
RL_US_MN_PSTN
RG_US_MN
RG_US_TN
 
RL_US_TN_PSTN
RG_US_TN
RG_US_MN
 
At this point, I need to convert the globalized numbers to their localized 
variants at the specific locations’ voice gateways
 
Called Party Transformation on MN gateway:
+\1952.XXX  à   Strip predot, Subscriber
+\.1[2-9]XX[2-9]XX   à   Strip predot, National
+\.!à   Strip 
predot, Prefix 011, International

Called Party Transformation on TN gateway:
+\1615.XXX  à   Strip predot, Subscriber
+\.1[2-9]XX[2-9]XX   à   Strip predot, National
+\.!à   Strip 
predot, Prefix 011, International
 
Primary Route (MN call out MN gateway) Verification:
1.   User 

Re: [OSL | CCIE_Voice] IPCCx 7.x

[Jason Granat]

I haven't tried that route. If you get the NFR kit it comes with the OS discs.

Sent while mobile.

On Mar 12, 2010, at 13:22, "Thomas Koch" 
mailto:koch1...@comcast.net>> wrote:

No…Just WIN2k3 enterprise…

Thomas J Koch
Unified Communicatons Consultant
CCDA, CCNA, CCVP
Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

From: Jason Granat [mailto:j...@slash128.com]
Sent: Friday, March 12, 2010 3:05 PM
To: wilson.sam...@usc-bt.com; 
koch1...@comcast.net; 
vpree...@cisco.com;  
adefilabi...@gmail.com
Cc:  
ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 
proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. 
After the OS is loaded and rebooted it will start the add new hardware wizard. 
Don’t click that. Just run the install VM Ware tools. After the tools are 
installed the drivers get updated and the wizard goes away. Then reboot again. 
Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR 
disc. That’s it… You could install VNC if you want instead of managing from the 
VMWare console. I’m sure other VM settings combos might work but this has 
always worked for me.

From: 
ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
wilson.sam...@usc-bt.com
Sent: Friday, March 12, 2010 12:41 PM
To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

Hi All,

I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if 
anyone has done it, please could you do me a favor by sharing your secret?

Regards
Wilson Samuel

From: 
ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch
Sent: Friday, March 12, 2010 2:49 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc:  
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

Yes…of course
Thanks…

Thomas J Koch
Consultant
CCDA, CCNA, CCVP
Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

From: Vishal Preenja [mailto:vpree...@cisco.com]
Sent: Friday, March 12, 2010 1:39 PM
To: 'Thomas Koch'; 'Omotayo'
Cc:  
ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Hi Thomas,

   Are ladies also allowed to reply? ☺

Pls check the hardware compatibility matrix below:

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf

HP DL380 G3 is not supported with IPCC 7.x onwards.


Thanks and Regards,

Vishal Preenja




From: Thomas Koch [mailto:koch1...@comcast.net]
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc:  
ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Gent’s,
Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?
I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said “non 
Cisco hardware” install will now stop”
T

E-mail:thomas.k...@compucom.com

From: 
ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc:  
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Yes, I am referring to MRG that is there in the MRGL of the trunk.

Just check that you should not have a MTP allocated for that call from CCM to 
CME.
You can verify by making a call and then check in RTMT that whether Transcoder 
is being invoked or MTP.


Thanks and Regards,

Vishal Preenja



From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc:  
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Hello,

Also what do you mean by MTP above transcoder. Are you reffering to the MRGL?
On Fri, Mar 12, 2010 at 7:09 PM, Vish

Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[Omotayo]

Hello Otto,

Yes requirement is to transport g729 over the WAN

if i want to transcoder on the trunk.

What do i need to do because quetion says use the hq resources


The last time i applied the transcoder to the trunk,

When hq phone call the sip phone on br2, i get a reorder tone

When the sip phone on the br2 calls the hq phone, it disconnects on pick up
and continues to ring on the sip phone

Thanks

On Fri, Mar 12, 2010 at 8:37 PM, Otto Sanchez  wrote:

> Hi,
>
> You want to transcode at the br2 rtr as I suppose your requirement is to
> transport the call using g.729 over the wan right?, if that's the case, make
> sure the incoming dial-peer codec is set to g.729, in that case the
> transcoder at br2 shoud be invoked if the sip phone codec is set to g.711,
>
>
>
> On Fri, Mar 12, 2010 at 10:03 AM, Omotayo  wrote:
>
>> Hello Otto,
>>
>> i had same issue
>>
>> The transcoder can be on the trunk?
>>
>> When i did the transcoder on the br2 router, i get a busy tone when the
>> sip phone is being called from the hq phone
>>
>> REgards
>>
>>   On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez wrote:
>>
>>> Hi Jeff,
>>>
>>> Would you please tell us more about the call flow and the end to end
>>> codec requirements for this call. If doing g.729 over the wan, and your sip
>>> phone is using g.711 you should transcode at br2,
>>>
>>> Please let us know,
>>>
>>> Thanks,
>>>
>>>  On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>>>
   Can anybody tell me if a PVDM2-32 can be used as a hardware
 transcoder on UCM.  Can’t seem to get a call from Call Manager to CME sip
 phone working.  I can call from CME to UCM but not the other way around.
 Rings but disconnects when answered.  Transcoder shows registered in Call
 manager.  Thanks





 Jeff

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


>>>
>>>
>>> --
>>> Regards,
>>>
>>> Otto Sanchez
>>> CCIE #25592 (Voice)
>>> Support Engineer - IPexpert, Inc.
>>> URL: http://www.IPexpert.com 
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com 
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] IPad Support

[Ashar Siddiqui]

iPad sucks big time! Sorry!

It's not that I am not an Apple fan..have got iPhone, iPod etc but this
big bulky thing looks very awkward..

Ash>

wilson.sam...@usc-bt.com wrote:

  Personally, I wouldn't prefer to buy Apple products (sorry mates, after the ebook price debate, I have lost respect for Mr. Jobs and APPL in general).

I would prefer them to develop it for Kindle and Sony E-Readers (I guess these at the moment have the max. marketshare) also iPad comes at a very high price and hasn't been even launched, though the demand seems to be looming.

I'm not sure if people have around 600 USD to buy E-readers just for the sake of a reader and some computing, esp. when Mac OS challenged people like me have remained loyal to Linux and Windows OS.

Again, it's the survival of the fittest, May the Best Product Win!!

Regards
Wilson Samuel

-Original Message-
From: Tanner Ezell [mailto:tanner.ez...@gmail.com] 
Sent: Friday, March 12, 2010 4:10 PM
To: Samuel, Wilson
Cc: bontacommunicati...@gmail.com; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPad Support

I challenge IP Expert to build interactive learning content for the
iPad (as many publishers are now looking into doing).

If done properly, would give IPX an incredible advantage over the competition.

As they say, Evolve or Die. :)

On Fri, Mar 12, 2010 at 3:56 PM,   wrote:
  
  
IMHO,

The best way to read PDFs is to read on a "computer" (aka Laptop, Palmtop, Tablet PC or Mac versions of the equivalent) and find yourself the most portable one that suits you.

I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really getting benefitted out of it.

However I'm keeping the Kindle because, all the newspaper and magazine subscriptions are a bit ok deal and its really portable, also most of the books are so far cheaper on Kindle edition (excluding the Cisco Press).

HTH

Kind Regards
Wilson Samuel

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of bontacommunications
Sent: Friday, March 12, 2010 9:42 AM
To: ccie_voice-boun...@onlinestudylist.com
Cc: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPad Support

I want to "green up" my library. I am considering purchase of an IPad. Does anyone know if the IPExpert PDF's security will be supported VIA IPad? If not currently supported, is there a roadmap to get to that point?


thanks
Chris


___
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


  
  


  




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

Re: [OSL | CCIE_Voice] IPCCx 7.x

[Thomas Koch]

No.Just WIN2k3 enterprise.

 

Thomas J Koch

Unified Communicatons Consultant 

CCDA, CCNA, CCVP

Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

 

From: Jason Granat [mailto:j...@slash128.com] 
Sent: Friday, March 12, 2010 3:05 PM
To: wilson.sam...@usc-bt.com; koch1...@comcast.net; vpree...@cisco.com;
adefilabi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

 

Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1
proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc.
After the OS is loaded and rebooted it will start the add new hardware
wizard. Don't click that. Just run the install VM Ware tools. After the
tools are installed the drivers get updated and the wizard goes away. Then
reboot again. Assign an IP to one NIC. Leave the second NIC disabled.
Install the UCCX IP IVR disc. That's it. You could install VNC if you want
instead of managing from the VMWare console. I'm sure other VM settings
combos might work but this has always worked for me.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
wilson.sam...@usc-bt.com
Sent: Friday, March 12, 2010 12:41 PM
To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

 

Hi All,

 

I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if
anyone has done it, please could you do me a favor by sharing your secret?

 

Regards

Wilson Samuel

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch
Sent: Friday, March 12, 2010 2:49 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

 

Yes.of course

Thanks.

 

Thomas J Koch

Consultant 

CCDA, CCNA, CCVP

Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

 

From: Vishal Preenja [mailto:vpree...@cisco.com] 
Sent: Friday, March 12, 2010 1:39 PM
To: 'Thomas Koch'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Hi Thomas,

 

   Are ladies also allowed to reply? J

 

Pls check the hardware compatibility matrix below:

 

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr
s/express_compatibility/matrix/crscomtx.pdf

 

HP DL380 G3 is not supported with IPCC 7.x onwards.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Thomas Koch [mailto:koch1...@comcast.net] 
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Gent's,

Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?

I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said "non
Cisco hardware" install will now stop"

T 

 

E-mail:thomas.k...@compucom.com

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Yes, I am referring to MRG that is there in the MRGL of the trunk.

 

Just check that you should not have a MTP allocated for that call from CCM
to CME.

You can verify by making a call and then check in RTMT that whether
Transcoder is being invoked or MTP.

 

Thanks and Regards,
Vishal Preenja
 

 

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello,

 

Also what do you mean by MTP above transcoder. Are you reffering to the
MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or
get me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello Visha,

 

when i did it as you described.

 

when sccp phone call sip phone on the cme, i get a reorder tone

when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

 

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If t

Re: [OSL | CCIE_Voice] IPad Support

[Wilson.Samuel]

Personally, I wouldn't prefer to buy Apple products (sorry mates, after the 
ebook price debate, I have lost respect for Mr. Jobs and APPL in general).

I would prefer them to develop it for Kindle and Sony E-Readers (I guess these 
at the moment have the max. marketshare) also iPad comes at a very high price 
and hasn't been even launched, though the demand seems to be looming.

I'm not sure if people have around 600 USD to buy E-readers just for the sake 
of a reader and some computing, esp. when Mac OS challenged people like me have 
remained loyal to Linux and Windows OS.

Again, it's the survival of the fittest, May the Best Product Win!!

Regards
Wilson Samuel

-Original Message-
From: Tanner Ezell [mailto:tanner.ez...@gmail.com] 
Sent: Friday, March 12, 2010 4:10 PM
To: Samuel, Wilson
Cc: bontacommunicati...@gmail.com; ccie_voice-boun...@onlinestudylist.com; 
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPad Support

I challenge IP Expert to build interactive learning content for the
iPad (as many publishers are now looking into doing).

If done properly, would give IPX an incredible advantage over the competition.

As they say, Evolve or Die. :)

On Fri, Mar 12, 2010 at 3:56 PM,   wrote:
> IMHO,
>
> The best way to read PDFs is to read on a "computer" (aka Laptop, Palmtop, 
> Tablet PC or Mac versions of the equivalent) and find yourself the most 
> portable one that suits you.
>
> I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it 
> comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really 
> getting benefitted out of it.
>
> However I'm keeping the Kindle because, all the newspaper and magazine 
> subscriptions are a bit ok deal and its really portable, also most of the 
> books are so far cheaper on Kindle edition (excluding the Cisco Press).
>
> HTH
>
> Kind Regards
> Wilson Samuel
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
> bontacommunications
> Sent: Friday, March 12, 2010 9:42 AM
> To: ccie_voice-boun...@onlinestudylist.com
> Cc: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] IPad Support
>
> I want to "green up" my library. I am considering purchase of an IPad. Does 
> anyone know if the IPExpert PDF's security will be supported VIA IPad? If not 
> currently supported, is there a roadmap to get to that point?
>
>
> thanks
> Chris
>
>
> ___
> 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
>



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

Re: [OSL | CCIE_Voice] IPad Support

[Tanner Ezell]

I challenge IP Expert to build interactive learning content for the
iPad (as many publishers are now looking into doing).

If done properly, would give IPX an incredible advantage over the competition.

As they say, Evolve or Die. :)

On Fri, Mar 12, 2010 at 3:56 PM,   wrote:
> IMHO,
>
> The best way to read PDFs is to read on a "computer" (aka Laptop, Palmtop, 
> Tablet PC or Mac versions of the equivalent) and find yourself the most 
> portable one that suits you.
>
> I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it 
> comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really 
> getting benefitted out of it.
>
> However I'm keeping the Kindle because, all the newspaper and magazine 
> subscriptions are a bit ok deal and its really portable, also most of the 
> books are so far cheaper on Kindle edition (excluding the Cisco Press).
>
> HTH
>
> Kind Regards
> Wilson Samuel
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
> bontacommunications
> Sent: Friday, March 12, 2010 9:42 AM
> To: ccie_voice-boun...@onlinestudylist.com
> Cc: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] IPad Support
>
> I want to "green up" my library. I am considering purchase of an IPad. Does 
> anyone know if the IPExpert PDF's security will be supported VIA IPad? If not 
> currently supported, is there a roadmap to get to that point?
>
>
> thanks
> Chris
>
>
> ___
> 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
>



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

Re: [OSL | CCIE_Voice] IPCCx 7.x

[Jason Granat]

Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 
proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. 
After the OS is loaded and rebooted it will start the add new hardware wizard. 
Don't click that. Just run the install VM Ware tools. After the tools are 
installed the drivers get updated and the wizard goes away. Then reboot again. 
Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR 
disc. That's it... You could install VNC if you want instead of managing from 
the VMWare console. I'm sure other VM settings combos might work but this has 
always worked for me.

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
wilson.sam...@usc-bt.com
Sent: Friday, March 12, 2010 12:41 PM
To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

Hi All,

I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if 
anyone has done it, please could you do me a favor by sharing your secret?

Regards
Wilson Samuel

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch
Sent: Friday, March 12, 2010 2:49 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

Yes...of course
Thanks...

Thomas J Koch
Consultant
CCDA, CCNA, CCVP
Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

From: Vishal Preenja [mailto:vpree...@cisco.com]
Sent: Friday, March 12, 2010 1:39 PM
To: 'Thomas Koch'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Hi Thomas,

   Are ladies also allowed to reply? :)

Pls check the hardware compatibility matrix below:

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf

HP DL380 G3 is not supported with IPCC 7.x onwards.


Thanks and Regards,

Vishal Preenja




From: Thomas Koch [mailto:koch1...@comcast.net]
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Gent's,
Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?
I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said "non 
Cisco hardware" install will now stop"
T

E-mail:thomas.k...@compucom.com

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Yes, I am referring to MRG that is there in the MRGL of the trunk.

Just check that you should not have a MTP allocated for that call from CCM to 
CME.
You can verify by making a call and then check in RTMT that whether Transcoder 
is being invoked or MTP.


Thanks and Regards,

Vishal Preenja



From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Hello,

Also what do you mean by MTP above transcoder. Are you reffering to the MRGL?
On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja 
mailto:vpree...@cisco.com>> wrote:
It will work as I described.

Can you send me the detailed ccm traces from all servers in the clusters or get 
me access of your box.


Thanks and Regards,

Vishal Preenja




From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Hello Visha,

when i did it as you described.

when sccp phone call sip phone on the cme, i get a reorder tone
when sip phone on the cme calls the sccp phone on the hq, it disconnects when 
hwq phone is picked and the sip phone continues to ring

How can this be fixed
On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja 
mailto:vpree...@cisco.com>> wrote:
Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Plea

Re: [OSL | CCIE_Voice] IPad Support

[Wilson.Samuel]

IMHO, 

The best way to read PDFs is to read on a "computer" (aka Laptop, Palmtop, 
Tablet PC or Mac versions of the equivalent) and find yourself the most 
portable one that suits you.

I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it 
comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really 
getting benefitted out of it.

However I'm keeping the Kindle because, all the newspaper and magazine 
subscriptions are a bit ok deal and its really portable, also most of the books 
are so far cheaper on Kindle edition (excluding the Cisco Press).

HTH

Kind Regards
Wilson Samuel

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of bontacommunications
Sent: Friday, March 12, 2010 9:42 AM
To: ccie_voice-boun...@onlinestudylist.com
Cc: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPad Support

I want to "green up" my library. I am considering purchase of an IPad. Does 
anyone know if the IPExpert PDF's security will be supported VIA IPad? If not 
currently supported, is there a roadmap to get to that point?


thanks
Chris


___
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] IPCCx 7.x

[Wilson.Samuel]

Hi All,

I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if 
anyone has done it, please could you do me a favor by sharing your secret?

Regards
Wilson Samuel

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch
Sent: Friday, March 12, 2010 2:49 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x

Yes...of course
Thanks...

Thomas J Koch
Consultant
CCDA, CCNA, CCVP
Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

From: Vishal Preenja [mailto:vpree...@cisco.com]
Sent: Friday, March 12, 2010 1:39 PM
To: 'Thomas Koch'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Hi Thomas,

   Are ladies also allowed to reply? :)

Pls check the hardware compatibility matrix below:

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf

HP DL380 G3 is not supported with IPCC 7.x onwards.


Thanks and Regards,

Vishal Preenja




From: Thomas Koch [mailto:koch1...@comcast.net]
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x

Gent's,
Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?
I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said "non 
Cisco hardware" install will now stop"
T

E-mail:thomas.k...@compucom.com

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Yes, I am referring to MRG that is there in the MRGL of the trunk.

Just check that you should not have a MTP allocated for that call from CCM to 
CME.
You can verify by making a call and then check in RTMT that whether Transcoder 
is being invoked or MTP.


Thanks and Regards,

Vishal Preenja



From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Hello,

Also what do you mean by MTP above transcoder. Are you reffering to the MRGL?
On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja 
mailto:vpree...@cisco.com>> wrote:
It will work as I described.

Can you send me the detailed ccm traces from all servers in the clusters or get 
me access of your box.


Thanks and Regards,

Vishal Preenja




From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

Hello Visha,

when i did it as you described.

when sccp phone call sip phone on the cme, i get a reorder tone
when sip phone on the cme calls the sccp phone on the hq, it disconnects when 
hwq phone is picked and the sip phone continues to ring

How can this be fixed
On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja 
mailto:vpree...@cisco.com>> wrote:
Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter 
mailto:jcot...@voxns.com>> wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


___
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] IPCCx 7.x

[Thomas Koch]

Yes.of course

Thanks.

 

Thomas J Koch

Consultant 

CCDA, CCNA, CCVP

Cisco IPT Design Specalist
CompuCom
Cell: 630-808-4910
E-mail:thom.k...@compucom.com

 

From: Vishal Preenja [mailto:vpree...@cisco.com] 
Sent: Friday, March 12, 2010 1:39 PM
To: 'Thomas Koch'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Hi Thomas,

 

   Are ladies also allowed to reply? J

 

Pls check the hardware compatibility matrix below:

 

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr
s/express_compatibility/matrix/crscomtx.pdf

 

HP DL380 G3 is not supported with IPCC 7.x onwards.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Thomas Koch [mailto:koch1...@comcast.net] 
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Gent's,

Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?

I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said "non
Cisco hardware" install will now stop"

T 

 

E-mail:thomas.k...@compucom.com

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Yes, I am referring to MRG that is there in the MRGL of the trunk.

 

Just check that you should not have a MTP allocated for that call from CCM
to CME.

You can verify by making a call and then check in RTMT that whether
Transcoder is being invoked or MTP.

 

Thanks and Regards,
Vishal Preenja
 

 

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello,

 

Also what do you mean by MTP above transcoder. Are you reffering to the
MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or
get me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello Visha,

 

when i did it as you described.

 

when sccp phone call sip phone on the cme, i get a reorder tone

when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

 

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


___
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] IPCCx 7.x

[Tanner Ezell]

Thomas,

I can't verify this will work for you (and I certainly wouldn't do
this for a production machine) however, you can apply the following
registry hack (which on a plain windows install, will allow UCCX to
install).

If it works, let me know.

Cheers,
Tanner Ezell

On Fri, Mar 12, 2010 at 2:39 PM, Vishal Preenja  wrote:
> Hi Thomas,
>
>
>
>    Are ladies also allowed to reply? J
>
>
>
> Pls check the hardware compatibility matrix below:
>
>
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf
>
>
>
> HP DL380 G3 is not supported with IPCC 7.x onwards.
>
>
>
> Thanks and Regards,
>
> Vishal Preenja
>
>
>
>
>
> 
>
> From: Thomas Koch [mailto:koch1...@comcast.net]
> Sent: Friday, March 12, 2010 1:50 PM
> To: 'Vishal Preenja'; 'Omotayo'
>
> Cc: ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x
>
>
>
> Gent’s,
>
> Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?
>
> I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said “non
> Cisco hardware” install will now stop”
>
> T
>
>
>
> E-mail:thomas.k...@compucom.com
>
>
>
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
> Sent: Friday, March 12, 2010 12:46 PM
> To: 'Omotayo'
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
>
>
>
> Yes, I am referring to MRG that is there in the MRGL of the trunk.
>
>
>
> Just check that you should not have a MTP allocated for that call from CCM
> to CME.
>
> You can verify by making a call and then check in RTMT that whether
> Transcoder is being invoked or MTP.
>
>
>
> Thanks and Regards,
>
> Vishal Preenja
>
>
>
>
>
> From: Omotayo [mailto:adefilabi...@gmail.com]
> Sent: Friday, March 12, 2010 1:25 PM
> To: Vishal Preenja
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
>
>
>
> Hello,
>
>
>
> Also what do you mean by MTP above transcoder. Are you reffering to the
> MRGL?
>
> On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:
>
> It will work as I described.
>
>
>
> Can you send me the detailed ccm traces from all servers in the clusters or
> get me access of your box.
>
>
>
> Thanks and Regards,
>
> Vishal Preenja
>
>
>
>
>
> 
>
> From: Omotayo [mailto:adefilabi...@gmail.com]
> Sent: Friday, March 12, 2010 12:33 PM
> To: Vishal Preenja
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
>
>
>
> Hello Visha,
>
>
>
> when i did it as you described.
>
>
>
> when sccp phone call sip phone on the cme, i get a reorder tone
>
> when sip phone on the cme calls the sccp phone on the hq, it disconnects
> when hwq phone is picked and the sip phone continues to ring
>
>
>
> How can this be fixed
>
> On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:
>
> Hi,
>
>   While making a call from the UCM to CME Sip phone ( because you have
> G711ulaw configured in the voice register pool), if you are getting
> disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
> and also make sure that you don't have MTP listed above transcoder. If there
> is MTP configured above transcoder, it will be allocated when transcoder is
> requested and the call will fail.
>
> Thanks and regards,
> Vishal Preenja.
>
> Hi Jeff,
>
> Would you please tell us more about the call flow and the end to end codec
> requirements for this call. If doing g.729 over the wan, and your sip phone
> is using g.711 you should transcode at br2,
>
> Please let us know,
>
> Thanks,
>
> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>
>>  Can anybody tell me if a PVDM2-32 can be used as a hardware
>> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
> phone working.
>> I can call from CME to UCM but not the other way around. Rings but
>> disconnects when answered.  Transcoder shows registered in Call manager.
>> Thanks
>
>
> ___
> 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
>
>



-- 
Regards,
Tanner Ezell


win2k3-reg-cisco.reg
Description: Binary data
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] IPCCx 7.x

[Vishal Preenja]

Hi Thomas,

 

   Are ladies also allowed to reply? :-)

 

Pls check the hardware compatibility matrix below:

 

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr
s/express_compatibility/matrix/crscomtx.pdf

 

HP DL380 G3 is not supported with IPCC 7.x onwards.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Thomas Koch [mailto:koch1...@comcast.net] 
Sent: Friday, March 12, 2010 1:50 PM
To: 'Vishal Preenja'; 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x 

 

Gent's,

Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?

I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said "non
Cisco hardware" install will now stop"

T 

 

E-mail:thomas.k...@compucom.com

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Yes, I am referring to MRG that is there in the MRGL of the trunk.

 

Just check that you should not have a MTP allocated for that call from CCM
to CME.

You can verify by making a call and then check in RTMT that whether
Transcoder is being invoked or MTP.

 

Thanks and Regards,
Vishal Preenja
 

 

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello,

 

Also what do you mean by MTP above transcoder. Are you reffering to the
MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or
get me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello Visha,

 

when i did it as you described.

 

when sccp phone call sip phone on the cme, i get a reorder tone

when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

 

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


___
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] SIP Hardware Transcoder

[Otto Sanchez]

Hi,

You want to transcode at the br2 rtr as I suppose your requirement is to
transport the call using g.729 over the wan right?, if that's the case, make
sure the incoming dial-peer codec is set to g.729, in that case the
transcoder at br2 shoud be invoked if the sip phone codec is set to g.711,


On Fri, Mar 12, 2010 at 10:03 AM, Omotayo  wrote:

> Hello Otto,
>
> i had same issue
>
> The transcoder can be on the trunk?
>
> When i did the transcoder on the br2 router, i get a busy tone when the sip
> phone is being called from the hq phone
>
> REgards
>
> On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez  wrote:
>
>> Hi Jeff,
>>
>> Would you please tell us more about the call flow and the end to end codec
>> requirements for this call. If doing g.729 over the wan, and your sip phone
>> is using g.711 you should transcode at br2,
>>
>> Please let us know,
>>
>> Thanks,
>>
>>  On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>>
>>>   Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder
>>> on UCM.  Can’t seem to get a call from Call Manager to CME sip phone
>>> working.  I can call from CME to UCM but not the other way around. Rings but
>>> disconnects when answered.  Transcoder shows registered in Call manager.
>>> Thanks
>>>
>>>
>>>
>>>
>>>
>>> Jeff
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>>
>> --
>> Regards,
>>
>> Otto Sanchez
>> CCIE #25592 (Voice)
>> Support Engineer - IPexpert, Inc.
>> URL: http://www.IPexpert.com 
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] IPCCx 7.x

[Thomas Koch]

Gent's,

Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..?

I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said "non
Cisco hardware" install will now stop"

T 

 

E-mail:thomas.k...@compucom.com

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja
Sent: Friday, March 12, 2010 12:46 PM
To: 'Omotayo'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Yes, I am referring to MRG that is there in the MRGL of the trunk.

 

Just check that you should not have a MTP allocated for that call from CCM
to CME.

You can verify by making a call and then check in RTMT that whether
Transcoder is being invoked or MTP.

 

Thanks and Regards,
Vishal Preenja
 

 

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello,

 

Also what do you mean by MTP above transcoder. Are you reffering to the
MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or
get me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello Visha,

 

when i did it as you described.

 

when sccp phone call sip phone on the cme, i get a reorder tone

when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

 

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


___
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] CCIE_Voice Digest, Vol 49, Issue 68

[Vishal Preenja]

Yes, I am referring to MRG that is there in the MRGL of the trunk.

 

Just check that you should not have a MTP allocated for that call from CCM
to CME.

You can verify by making a call and then check in RTMT that whether
Transcoder is being invoked or MTP.

 

Thanks and Regards,
Vishal Preenja
 

 

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 1:25 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello,

 

Also what do you mean by MTP above transcoder. Are you reffering to the
MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or
get me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello Visha,

 

when i did it as you described.

 

when sccp phone call sip phone on the cme, i get a reorder tone

when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

 

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


___
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] CCIE_Voice Digest, Vol 49, Issue 68

[Omotayo]

Hello,

Also what do you mean by MTP above transcoder. Are you reffering to the
MRGL?

On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja  wrote:

>  It will work as I described.
>
>
>
> Can you send me the detailed ccm traces from all servers in the clusters or
> get me access of your box.
>
>
>
> Thanks and Regards,
>
> Vishal Preenja
>
>
>
>
>  --
>
> *From:* Omotayo [mailto:adefilabi...@gmail.com]
> *Sent:* Friday, March 12, 2010 12:33 PM
> *To:* Vishal Preenja
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
>
>
>
> Hello Visha,
>
>
>
> when i did it as you described.
>
>
>
> when sccp phone call sip phone on the cme, i get a reorder tone
>
> when sip phone on the cme calls the sccp phone on the hq, it disconnects
> when hwq phone is picked and the sip phone continues to ring
>
>
>
> How can this be fixed
>
> On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja 
> wrote:
>
> Hi,
>
>   While making a call from the UCM to CME Sip phone ( because you have
> G711ulaw configured in the voice register pool), if you are getting
> disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
> and also make sure that you don't have MTP listed above transcoder. If
> there
> is MTP configured above transcoder, it will be allocated when transcoder is
> requested and the call will fail.
>
> Thanks and regards,
> Vishal Preenja.
>
> Hi Jeff,
>
> Would you please tell us more about the call flow and the end to end codec
> requirements for this call. If doing g.729 over the wan, and your sip phone
> is using g.711 you should transcode at br2,
>
> Please let us know,
>
> Thanks,
>
> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>
> >  Can anybody tell me if a PVDM2-32 can be used as a hardware
> > transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
> phone working.
> > I can call from CME to UCM but not the other way around. Rings but
> > disconnects when answered.  Transcoder shows registered in Call manager.
> > Thanks
>
>
> ___
> 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] CCIE_Voice Digest, Vol 49, Issue 68

[Vishal Preenja]

It will work as I described.

 

Can you send me the detailed ccm traces from all servers in the clusters or
get me access of your box.

 

Thanks and Regards,
Vishal Preenja
 

 

  _  

From: Omotayo [mailto:adefilabi...@gmail.com] 
Sent: Friday, March 12, 2010 12:33 PM
To: Vishal Preenja
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

 

Hello Visha,

 

when i did it as you described.

 

when sccp phone call sip phone on the cme, i get a reorder tone

when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

 

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

Hi,

  While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


___
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

[OSL | CCIE_Voice] ip phone on layer 3 interface

[Angel Perez]

Hello:

 

I want to connect an ip phone to a 2811 fast ether 0/0 interface (pstn router) 
this way I wouldn't need a switch for the pstn phone

A xcable is needed but I'm not sure if a layer 3 interface is l2 switching 
capable

 

Any suggestion?

 

Thanks
  
_
Recibe en tu móvil un SMS con tu Hotmail recibido. ¡Date de alta ya!
http://serviciosmoviles.es.msn.com/___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] via gatekeeper "invia" key word

[Jeff Cotter]

Thanks Otto, if this is the case then I believe the explanation Mark S. gives 
on the VOD is incorrect.  As he references the invia between local zones.

From: Otto Sanchez [mailto:o...@ipexpert.com]
Sent: Friday, March 12, 2010 6:14 AM
To: Jeff Cotter
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] via gatekeeper "invia" key word

Hi Jeff,

According to your lab results, you are describing the expected behavior, more 
information at:

http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/ve-gk-config_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1225776

Thanks!,
On Thu, Mar 11, 2010 at 9:55 AM, Jeff Cotter 
mailto:jcot...@voxns.com>> wrote:
I am struggling a bit with the invia concept.  I think I understand the 
"outvia".  When I lab this up I find the following.

Invia only applies to calls coming from a remote GK.  In order for call to use 
cube I had to configure the invia key word on the actual remote zone.not on 
the destination zone. Sample config of my invia GK

gk zone local ucm cisco.com 1.1.1.1
gk zone local cube
gk zone local cme
gk zone remote gk2 lab.com 2.2.2.2 invia cube
zone prefixs omitted

So calls coming FROM gk2 destined for either ucm or cme zone used the cube.  If 
I applied the invia key word on either ucm or cme zone directly, the cube was 
not invoked.  This seems to conflict with the proctor guide mock lab 1 
statement "invia command when defining the UCME zone would invoke the cube for 
calls coming in from a remote zone".  In my lab applying invia directly to 
destination zone had no affect and cube was not invoked.

Am I missing something.

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



--
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[Jeff Cotter]

Thanks Otto I was able to figure it out.  I was looking/configuring  the wrong 
inbound dial-peer!!  Thank you for your response however.



Jeff
From: Otto Sanchez [mailto:o...@ipexpert.com]
Sent: Friday, March 12, 2010 4:00 AM
To: Jeff Cotter
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec 
requirements for this call. If doing g.729 over the wan, and your sip phone is 
using g.711 you should transcode at br2,

Please let us know,

Thanks,
On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter 
mailto:jcot...@voxns.com>> wrote:
Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM.  
Can't seem to get a call from Call Manager to CME sip phone working.  I can 
call from CME to UCM but not the other way around. Rings but disconnects when 
answered.  Transcoder shows registered in Call manager.  Thanks


Jeff

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



--
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jason Granat]

Hi Otto,

Thanks for the advice. In your second paragraph the opposite was actually the 
case. The E1 voice-ports were originally showing a-law, and had distortion. I 
hard set u-law on the E1 ports between the gateway and PSTN router and the 
distortion went away. Perhaps that is what you meant?

I took a look at the link you included. I'll have to do some testing but my 
main question is how is this handled in the real world at the provider level?

Thanks,

Jason

From: Otto Sanchez [mailto:o...@ipexpert.com]
Sent: Friday, March 12, 2010 4:59 AM
To: Jason Granat
Cc: 
Subject: Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

Hello Jason,

E1's and T1's will always use a-law and u-law companding mechanism 
respectively, this is used to give more "resolution" to low voice frequencies 
when digitizing an analog signal (the mechanism is also used in the other end 
for digital to analogue conversion), each mechanism is designed exclusively to 
work with its voice digital standard and cannot be used conversely,

In that sense, my guess is that before applying that command in your E1 port, 
the companding type was u-law, you can verify this using the sh voice port 
command (perhaps the default configuration of a-law was somehow overwritten by 
a cptone command in the same port configuration), and when you hardcoded the 
a-law companding type everything worked as expected,

I also found a note in the Cisco IOS Voice Port Configuration Guide, which says 
that the command is used when cross-connecting in a local router,

http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871


HTH,
On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat 
mailto:j...@slash128.com>> wrote:
So I've got this partially figured out. It had to do with the compand-type. E1 
was a-law and T1 was u-law. I set the E1 side for u-law and it sounds correct 
now.

The final thing I am trying to figure out is how to 'trans-compand' (if that is 
the correct term) on the PSTN gateway. As it sits I had to change the 
compand-type between the PSTN and E1 gateway. I don't have experience with 
foreign connectivity so maybe this is the way it is done in the real world but 
I am thinking that perhaps the E1 site may not want or be able to change their 
compand-type, so can it be changed at the PSTN level between a-law and u-law 
locations?

Thanks,

Jason

From: Jason Granat
Sent: Thursday, March 11, 2010 9:46 AM
To: mailto:ccie_voice@onlinestudylist.com>>
Subject: PSTN Call Distortion Between T1/E1

Perhaps this is something simple that I am overlooking but I have the generic 
setup running in my home lab with 3 gateways and one PSTN router. 2 of the 
gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 
to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. 
Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, 
like the gain is way too high. I tried playing with the gain on the voice-port 
but no luck. I'm not finding much online or in Cisco docs. Any suggestions?

Thanks,

Jason




http://slash128.com

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--
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com




http://slash128.com
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

[Omotayo]

Hello Visha,

when i did it as you described.

when sccp phone call sip phone on the cme, i get a reorder tone
when sip phone on the cme calls the sccp phone on the hq, it disconnects
when hwq phone is picked and the sip phone continues to ring

How can this be fixed

On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja  wrote:

> Hi,
>
>   While making a call from the UCM to CME Sip phone ( because you have
> G711ulaw configured in the voice register pool), if you are getting
> disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
> and also make sure that you don't have MTP listed above transcoder. If
> there
> is MTP configured above transcoder, it will be allocated when transcoder is
> requested and the call will fail.
>
> Thanks and regards,
> Vishal Preenja.
>
> Hi Jeff,
>
> Would you please tell us more about the call flow and the end to end codec
> requirements for this call. If doing g.729 over the wan, and your sip phone
> is using g.711 you should transcode at br2,
>
> Please let us know,
>
> Thanks,
>
> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>
> >  Can anybody tell me if a PVDM2-32 can be used as a hardware
> > transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
> phone working.
> > I can call from CME to UCM but not the other way around. Rings but
> > disconnects when answered.  Transcoder shows registered in Call manager.
> > Thanks
>
>
> ___
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] MVA

[anupam TYAGI]

if i dial that external number without MVA it goes through ,but when in MVA
i get a disconnect when calling this external number ( so don't seems to be
codec issue )

On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer wrote:

> are you maybe calling to a remote location (g.729) and therefore a xcoder
> is required, but not set up correctly?
>
> 2010/3/12 anupam TYAGI 
>
>> i saw the call hit the gateway .  RDP is having the same CSS as phone CSS
>>
>>
>>
>> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
>> mjbe...@krollontrack.com> wrote:
>>
>>>   Check the CSS on the remote destination profile you’re calling from.
>>>
>>> If you do a “debug isdn q931” on the PSTN gateway, do you see the call
>>> hit the gateway?
>>>
>>>
>>>
>>> Your rerouting CSS on the RDP is used for calls out to your RD.
>>>
>>> Your CSS on the RDP is used for calls through MVA that are routed out
>>> through your PSTN gateway.
>>>
>>>
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
>>> *Sent:* Friday, March 12, 2010 9:27 AM
>>> *To:* ccie_voice-requ...@onlinestudylist.com;
>>> ccie_voice@onlinestudylist.com
>>> *Subject:* [OSL | CCIE_Voice] MVA
>>>
>>>
>>>
>>> Hi Folks
>>>
>>> I am doing MVA , When i dial the MVA number ,  I am able to hear the
>>> prompt.  I dial a  PSTN number , but the call disconnect . Can any body
>>> suggest me what can be the reason .
>>>
>>>
>>> Rgds
>>> Anu.
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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Re: [OSL | CCIE_Voice] MVA

[Patrick Fischer]

are you maybe calling to a remote location (g.729) and therefore a xcoder is
required, but not set up correctly?

2010/3/12 anupam TYAGI 

> i saw the call hit the gateway .  RDP is having the same CSS as phone CSS
>
>
>
> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
> mjbe...@krollontrack.com> wrote:
>
>>   Check the CSS on the remote destination profile you’re calling from.
>>
>> If you do a “debug isdn q931” on the PSTN gateway, do you see the call hit
>> the gateway?
>>
>>
>>
>> Your rerouting CSS on the RDP is used for calls out to your RD.
>>
>> Your CSS on the RDP is used for calls through MVA that are routed out
>> through your PSTN gateway.
>>
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
>> *Sent:* Friday, March 12, 2010 9:27 AM
>> *To:* ccie_voice-requ...@onlinestudylist.com;
>> ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] MVA
>>
>>
>>
>> Hi Folks
>>
>> I am doing MVA , When i dial the MVA number ,  I am able to hear the
>> prompt.  I dial a  PSTN number , but the call disconnect . Can any body
>> suggest me what can be the reason .
>>
>>
>> Rgds
>> Anu.
>>
>
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] MVA

[anupam TYAGI]

i saw the call hit the gateway .  RDP is having the same CSS as phone CSS


On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J.  wrote:

>  Check the CSS on the remote destination profile you’re calling from.
>
> If you do a “debug isdn q931” on the PSTN gateway, do you see the call hit
> the gateway?
>
>
>
> Your rerouting CSS on the RDP is used for calls out to your RD.
>
> Your CSS on the RDP is used for calls through MVA that are routed out
> through your PSTN gateway.
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
> *Sent:* Friday, March 12, 2010 9:27 AM
> *To:* ccie_voice-requ...@onlinestudylist.com;
> ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MVA
>
>
>
> Hi Folks
>
> I am doing MVA , When i dial the MVA number ,  I am able to hear the
> prompt.  I dial a  PSTN number , but the call disconnect . Can any body
> suggest me what can be the reason .
>
>
> Rgds
> Anu.
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68

[Vishal Preenja]

Hi,

   While making a call from the UCM to CME Sip phone ( because you have
G711ulaw configured in the voice register pool), if you are getting
disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk
and also make sure that you don't have MTP listed above transcoder. If there
is MTP configured above transcoder, it will be allocated when transcoder is
requested and the call will fail.

Thanks and regards,
Vishal Preenja.

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware 
> transcoder on UCM.  Can?t seem to get a call from Call Manager to CME sip
phone working.
> I can call from CME to UCM but not the other way around. Rings but 
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks


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Re: [OSL | CCIE_Voice] MVA

[Berry, Matthew J.]

Check the CSS on the remote destination profile you're calling from.
If you do a "debug isdn q931" on the PSTN gateway, do you see the call hit the 
gateway?

Your rerouting CSS on the RDP is used for calls out to your RD.
Your CSS on the RDP is used for calls through MVA that are routed out through 
your PSTN gateway.

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anupam TYAGI
Sent: Friday, March 12, 2010 9:27 AM
To: ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MVA

Hi Folks

I am doing MVA , When i dial the MVA number ,  I am able to hear the  prompt.  
I dial a  PSTN number , but the call disconnect . Can any body suggest me what 
can be the reason .


Rgds
Anu.
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[OSL | CCIE_Voice] MVA

[anupam TYAGI]

Hi Folks

I am doing MVA , When i dial the MVA number ,  I am able to hear the
prompt.  I dial a  PSTN number , but the call disconnect . Can any body
suggest me what can be the reason .


Rgds
Anu.
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[OSL | CCIE_Voice] Pt2. Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO)

[Berry, Matthew J.]

I could also simplify my translation patterns and make them available to ALL 
sites.
By creating a separate partition called PT_US_Translate and setting up the 
following patterns:
9.!-->   Predot, Prefix +1  --> 
  Result: +19525163748 (local MN)
9.1!-->   Predot, Prefix +-->   
Result: +1615444 (remote long-distance in TN)
9011.! -->   Predot, Prefix +-->
   Result: +3432141861 (remote international)


From: Berry, Matthew J.
Sent: Friday, March 12, 2010 9:12 AM
To: OSL Group
Subject: Globalized + Dialing - Applying Called Party Number Type for Secondary 
Route Scenario (non-TEHO)

All -

I am setting up + dialing on a self-made lab.  A question has come up as to 
where the Called Party Number Type should be set.  For this exercise, I want to 
find the best way to route calls through a system, utilizing alternate paths 
for failover scenarios.  Those this does not take TEHO into account, I want a 
format that can easily accommodate TEHO situations.  I believe my method below 
will do that.

PSTN is expecting:
Subscriber: Seven digits, Subscriber
National: Eleven digits (incl. 1), National
Intl: Undefined digits, Intl

I have also setup translation patterns in PT_US_MN_EP_PSTN setup as:
9.952[2-9]XX-->   Predot, Prefix +1 
 -->   Result: +19525163748 (local MN)
9.1[2-9]XX[2-9]XX -->   Predot, Prefix +
-->   Result: +1615444 (remote long-distance in TN)
9.011!-->   Predot, 
Prefix +-->   Result: +3432141861 (remote international)

I have route patterns in PT_US_MN_EP_PSTN setup as:
\+1952[2-9]XX-->   +19525163748
\+1[2-9]XX[2-9]XX-->   +1615444
\+!  -->   
+3432141861
\+!#   -->   
+3432141861
Note: I am not using predot in the route patterns!!

At this point, all dialed numbers have been globalized from their localized 
variants.

I am trying to figure out if it would be best to apply the Called Party Number 
Type (Sub, Nat, Intl) at the Translation Pattern, Route Pattern, Route List 
(i.e. Route Group settings), or Called Party Transformation on the gateway 
level.  If I did not modify the Called Party Number Type at the Translation 
Pattern or Route Pattern, it would allow me to configure different call 
treatment at the route list level for TEHO.  However, I think it would be best 
to apply that setting at the Called Party Transformation pattern on the MGCP 
gateways (avoiding H.323 for sake of conversation).

I have two locations that calls will be sent out, MN and TN.  For example:

RL_US_MN_PSTN
RG_US_MN
RG_US_TN

RL_US_TN_PSTN
RG_US_TN
RG_US_MN

At this point, I need to convert the globalized numbers to their localized 
variants at the specific locations' voice gateways

Called Party Transformation on MN gateway:
+\1952.XXX  -->   Strip predot, 
Subscriber
+\.1[2-9]XX[2-9]XX   -->   Strip predot, National
+\.!-->   Strip 
predot, Prefix 011, International

Called Party Transformation on TN gateway:
+\1615.XXX  -->   Strip predot, 
Subscriber
+\.1[2-9]XX[2-9]XX   -->   Strip predot, National
+\.!-->   Strip 
predot, Prefix 011, International

Primary Route (MN call out MN gateway) Verification:

1.   User in MN dials local number as 9.9525163748 (my desk phone, give me 
a call :))

2.   Translation pattern changes to +19525163748

3.   Matches route pattern of +1952[2-9]XX

4.   Route pattern sent via RL_US_MN_PSTN to RG_US_MN

5.   RG_US_MN sends call to US_MN_Gateway1

6.   US_MN_Gateway1 has a called transformation pattern of +\1952.XXX 
(Subscriber)

7.   Call goes out the US_MN_Gateway1 as 5163748 (Subscriber)

Secondary Route (MN call out TN gateway) Verification:

1.   User in MN dials local number as 9.9525163748 (my desk phone, give me 
a call :))

2.   Translation pattern changes to +19525163748

3.   Matches route pattern of +1952[2-9]XX

4.   Route pattern sent via RL_US_MN_PSTN to RG_US_TN (RG_US_MN is down and 
not functioning)

5.   RG_US_TN sends call to US_TN_Gateway1

6.   US_TN_Gateway1 has a called transformation pattern of 
+\.1[2-9]XX[2-9]XX (National)

7.   Call goes out the US_TN_Gateway1 as 19525163748 (National)

Any feedback would be appreciated.  It took me about 30 minutes to think this 
through and 

[OSL | CCIE_Voice] Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO)

[Berry, Matthew J.]

All -

I am setting up + dialing on a self-made lab.  A question has come up as to 
where the Called Party Number Type should be set.  For this exercise, I want to 
find the best way to route calls through a system, utilizing alternate paths 
for failover scenarios.  Those this does not take TEHO into account, I want a 
format that can easily accommodate TEHO situations.  I believe my method below 
will do that.

PSTN is expecting:
Subscriber: Seven digits, Subscriber
National: Eleven digits (incl. 1), National
Intl: Undefined digits, Intl

I have also setup translation patterns in PT_US_MN_EP_PSTN setup as:
9.952[2-9]XX-->   Predot, Prefix +1 
 -->   Result: +19525163748 (local MN)
9.1[2-9]XX[2-9]XX -->   Predot, Prefix +
-->   Result: +1615444 (remote long-distance in TN)
9.011!-->   Predot, 
Prefix +-->   Result: +3432141861 (remote international)

I have route patterns in PT_US_MN_EP_PSTN setup as:
\+1952[2-9]XX-->   +19525163748
\+1[2-9]XX[2-9]XX-->   +1615444
\+!  -->   
+3432141861
\+!#   -->   
+3432141861
Note: I am not using predot in the route patterns!!

At this point, all dialed numbers have been globalized from their localized 
variants.

I am trying to figure out if it would be best to apply the Called Party Number 
Type (Sub, Nat, Intl) at the Translation Pattern, Route Pattern, Route List 
(i.e. Route Group settings), or Called Party Transformation on the gateway 
level.  If I did not modify the Called Party Number Type at the Translation 
Pattern or Route Pattern, it would allow me to configure different call 
treatment at the route list level for TEHO.  However, I think it would be best 
to apply that setting at the Called Party Transformation pattern on the MGCP 
gateways (avoiding H.323 for sake of conversation).

I have two locations that calls will be sent out, MN and TN.  For example:

RL_US_MN_PSTN
RG_US_MN
RG_US_TN

RL_US_TN_PSTN
RG_US_TN
RG_US_MN

At this point, I need to convert the globalized numbers to their localized 
variants at the specific locations' voice gateways

Called Party Transformation on MN gateway:
+\1952.XXX  -->   Strip predot, 
Subscriber
+\.1[2-9]XX[2-9]XX   -->   Strip predot, National
+\.!-->   Strip 
predot, Prefix 011, International

Called Party Transformation on TN gateway:
+\1615.XXX  -->   Strip predot, 
Subscriber
+\.1[2-9]XX[2-9]XX   -->   Strip predot, National
+\.!-->   Strip 
predot, Prefix 011, International

Primary Route (MN call out MN gateway) Verification:

1.   User in MN dials local number as 9.9525163748 (my desk phone, give me 
a call :))

2.   Translation pattern changes to +19525163748

3.   Matches route pattern of +1952[2-9]XX

4.   Route pattern sent via RL_US_MN_PSTN to RG_US_MN

5.   RG_US_MN sends call to US_MN_Gateway1

6.   US_MN_Gateway1 has a called transformation pattern of +\1952.XXX 
(Subscriber)

7.   Call goes out the US_MN_Gateway1 as 5163748 (Subscriber)

Secondary Route (MN call out TN gateway) Verification:

1.   User in MN dials local number as 9.9525163748 (my desk phone, give me 
a call :))

2.   Translation pattern changes to +19525163748

3.   Matches route pattern of +1952[2-9]XX

4.   Route pattern sent via RL_US_MN_PSTN to RG_US_TN (RG_US_MN is down and 
not functioning)

5.   RG_US_TN sends call to US_TN_Gateway1

6.   US_TN_Gateway1 has a called transformation pattern of 
+\.1[2-9]XX[2-9]XX (National)

7.   Call goes out the US_TN_Gateway1 as 19525163748 (National)

Any feedback would be appreciated.  It took me about 30 minutes to think this 
through and type it out.  Because it takes so long, I am trying to build a 
strawman structure that I can easily drop into the lab and modify to support my 
needs.

What say ye?

Matthew Berry
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[OSL | CCIE_Voice] IPad Support

[bontacommunications]

I want to "green up" my library. I am considering purchase of an IPad. Does 
anyone know if the IPExpert PDF's security will be supported VIA IPad? If not 
currently supported, is there a roadmap to get to that point?


thanks
Chris


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Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[Omotayo]

Hello Otto,

i had same issue

The transcoder can be on the trunk?

When i did the transcoder on the br2 router, i get a busy tone when the sip
phone is being called from the hq phone

REgards

On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez  wrote:

> Hi Jeff,
>
> Would you please tell us more about the call flow and the end to end codec
> requirements for this call. If doing g.729 over the wan, and your sip phone
> is using g.711 you should transcode at br2,
>
> Please let us know,
>
> Thanks,
>
>  On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>
>>   Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder
>> on UCM.  Can’t seem to get a call from Call Manager to CME sip phone
>> working.  I can call from CME to UCM but not the other way around. Rings but
>> disconnects when answered.  Transcoder shows registered in Call manager.
>> Thanks
>>
>>
>>
>>
>>
>> Jeff
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com 
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] via gatekeeper "invia" key word

[Otto Sanchez]

Hi Jeff,

According to your lab results, you are describing the expected behavior,
more information at:

http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/ve-gk-config_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1225776

Thanks!,

On Thu, Mar 11, 2010 at 9:55 AM, Jeff Cotter  wrote:

>  I am struggling a bit with the invia concept.  I think I understand the
> “outvia”.  When I lab this up I find the following.
>
>
>
> Invia only applies to calls coming from a remote GK.  In order for call to
> use cube I had to configure the invia key word on the actual remote
> zone…..not on the destination zone. Sample config of my invia GK
>
>
>
> gk zone local ucm cisco.com 1.1.1.1
>
> gk zone local cube
>
> gk zone local cme
>
> gk zone remote gk2 lab.com 2.2.2.2 invia cube
>
> zone prefixs omitted
>
>
>
> So calls coming FROM gk2 destined for either ucm or cme zone used the
> cube.  If I applied the invia key word on either ucm or cme zone directly,
> the cube was not invoked.  This seems to conflict with the proctor guide
> mock lab 1 statement “invia command when defining the UCME zone would invoke
> the cube for calls coming in from a remote zone”.  In my lab applying invia
> directly to destination zone had no affect and cube was not invoked.
>
>
>
> Am I missing something.
>
> ___
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> visit www.ipexpert.com
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
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Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem

[Otto Sanchez]

Hi Mike,

I'm noticing from your initial debugs that the 156.26.1.70:1719 ip
address/port the one confirming the GRQ message from BR2-RTR

***
value RasMessage ::= gatekeeperConfirm :
{
  requestSeqNum 126
  protocolIdentifier { 0 0 8 2250 0 4 }
  gatekeeperIdentifier {"PL"}
  rasAddress ipAddress :
  {
ip '*9C1A0146*'H
port *1719*
  }
}

After the Angel's suggestion this should have been corrected, so would you
please send the same debugs now from both routers? plus a show gatek zone
status, also I see that you are not pinging from the br2 l0 interface but
the closest to hq l0 interface (which might be the serial interface), please
try to use ping with options to verify that loopbacks can see each other,

Thanks!,

On Thu, Mar 11, 2010 at 2:24 PM, Mike Peterson  wrote:

> Hi Angel,
>
> Thanks for helping me out with this GK issue. Yes indeed the GW doesn't
> receive the message , that is why we are seeing GRQ and GCF.
> I do have full connectivity b/w HQ/BR2/PUB/SUB .
> Below are the ping's you sugested to post.
>
> Thanks a lot in advance for your time and help.
>
> HQ#ping 192.21.66.254->from HQ to loopback of BR2
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.21.66.254, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 1/5/12 ms
> HQ#
>
>
> BR2-RTR#ping 192.21.64.254-->from BR2 to loopback of HQ
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.21.64.254, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 8/14/32 ms
> BR2-RTR#
>
>
>
>
>
> HQ#ping 192.168.0.11  >from HQ to  CUCM PUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 1/4/8 ms
>
>
>
> HQ#ping 192.168.0.12 -> from HQ to CUCM SUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 16/25/40 ms
> HQ#
>
>
> BR2-RTR#ping 192.168.0.11  ->from BR2 CUCM PUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 16/27/44 ms
>
>
> BR2-RTR#ping 192.168.0.12---> from BR2 to CUCM SUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 20/40/52 ms
> BR2-RTR#
>
>
>
> --
> *From:* Angel Perez **
> *Sent:* Thu, March 11, 2010 1:17:17 PM
> *Subject:* RE: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
>
> Just to verify, can you ping hq loo 0 192.21.64.254 from br2? And br2 loop
> 192.21.66.254 from hq?
>
> It looks like br2 gw ask for registration GRQ, and then gk try to confirm
> GCF but the gw can't recieve the message
>
> hth
> --
> Date: Thu, 11 Mar 2010 08:54:30 -0800
>
>
> Subject: Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
>
> Hi All,
>
> I did tried your suggestion (to add loopback IP address :
>  zone local PL cisco.com 192.21.64.254 ) which does make sense  but it
> doesn't work.
> I took a look at "deb ras" and I am seeing only GRQ (a message sent by
> endpoint to GK ) and GCF (A reply from gatekeeper to endpoint
> which indicates the transport address of the gatekeeper RAS channel) and I
> am not seeing GRJ (the reject the endpoint request for registration) so
> something I am missing or  I am hitting a BUG!
> The "deb gatekeeper main 19" or "deb h225 asn1" still doesn't give me a
> clue of why GK is failing to register.
>
> Once again thanks for your time and help.
>
> Kind Regards,
>
> Mike
>
>
> Note: This is the change I made:
>
> gatekeeper
>  zone local PL cisco.com 192.21.64.254
>  zone prefix PL 1... gw-priority 10 gk-trunk_2
>  zone prefix PL 1... gw-priority 9 gk-trunk_1
>  zone prefix PL 1... gw-priority 0 BR2-RTR
>  zone prefix PL 5... gw-priority 10 gk-trunk_2
>  zone prefix PL 5... gw-priority 9 gk-trunk_1
>  zone prefix PL 5... gw-priority 0 BR2-RTR
>  no shutdown
> !
>
>
>
>
> --
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> cómo!
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Otto Sanchez]

Hello Jason,

E1's and T1's will always use a-law and u-law companding mechanism
respectively, this is used to give more "resolution" to low voice
frequencies when digitizing an analog signal (the mechanism is also used in
the other end for digital to analogue conversion), each mechanism is
designed exclusively to work with its voice digital standard and cannot be
used conversely,

In that sense, my guess is that before applying that command in your E1
port, the companding type was u-law, you can verify this using the sh voice
port command (perhaps the default configuration of a-law was somehow
overwritten by a cptone command in the same port configuration), and when
you hardcoded the a-law companding type everything worked as expected,

I also found a note in the Cisco IOS Voice Port Configuration Guide, which
says that the command is used when cross-connecting in a local router,

http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871


HTH,

On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat  wrote:

>  So I’ve got this partially figured out. It had to do with the
> compand-type. E1 was a-law and T1 was u-law. I set the E1 side for u-law and
> it sounds correct now.
>
>
>
> The final thing I am trying to figure out is how to ‘trans-compand’ (if
> that is the correct term) on the PSTN gateway. As it sits I had to change
> the compand-type between the PSTN and E1 gateway. I don’t have experience
> with foreign connectivity so maybe this is the way it is done in the real
> world but I am thinking that perhaps the E1 site may not want or be able to
> change their compand-type, so can it be changed at the PSTN level between
> a-law and u-law locations?
>
>
>
> Thanks,
>
>
>
> Jason
>
>
>
> *From:* Jason Granat
> *Sent:* Thursday, March 11, 2010 9:46 AM
> *To:* 
> *Subject:* PSTN Call Distortion Between T1/E1
>
>
>
> Perhaps this is something simple that I am overlooking but I have the
> generic setup running in my home lab with 3 gateways and one PSTN router. 2
> of the gateways are T1 and one is E1. The PSTN router is also running CME
> with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN
> phone are fine. Calls between T1 sites are fine. Calls between T1 and E1
> sites are distorted, like the gain is way too high. I tried playing with the
> gain on the voice-port but no luck. I’m not finding much online or in Cisco
> docs. Any suggestions?
>
>
>
> Thanks,
>
>
>
> Jason
>
> --
>
>
> http://slash128.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
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Re: [OSL | CCIE_Voice] RSVP WITH MLPoFR

[Angel Perez]

Hello:

 

The same happend with multicast moh traffic, after activating auto qos  you 
need to move ip pim sparse-dense mode to the virtual template interface

 

thanks
 


From: roger.kallb...@cygate.se
To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com
Date: Fri, 12 Mar 2010 06:18:37 +0100
Subject: RE: [OSL | CCIE_Voice] RSVP WITH MLPoFR







That’s the expected behavior. Auto qos won’t move the ip rsvp bandwith command. 
That’s one of the quirks with auto qos.
 
Brgds,

Roger Källberg
Unified Communication Consultant
Cygate AB


 


From: Angel Perez [mailto:gorr...@hotmail.com] 
Sent: den 10 mars 2010 19:37
To: osl osl
Subject: [OSL | CCIE_Voice] RSVP WITH MLPoFR
 
Hello:
 
I was configurin MLPoFR and LFI on a link between hq and br1, on the serial 
interface I had:
 
interface Serial0/2/0.202 point-to-point
  ip rsvp bandwidth 64 
 
Calls where progressing as configured (two g729 calls)
 
Then after apply auto qos voip trust fr-atm new virtual templates and virtual 
access interfaces are created
 
Then trying to test the policy-map just created and tuned I noticed that I 
could not make calls from hq to br1 (rsvp was rejecting the call)
 
So I added the following at hq and br1:
 
interface Virtual-Template200
 ip rsvp bandwidth 64
 
And the problem get solved
 
Is this the normal situation? I suppose it is but not 100% sure
 
Thanks
 
 



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Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[Otto Sanchez]

Hi Jeff,

Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone
is using g.711 you should transcode at br2,

Please let us know,

Thanks,

On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:

>  Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on
> UCM.  Can’t seem to get a call from Call Manager to CME sip phone working.
> I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered.  Transcoder shows registered in Call manager.
> Thanks
>
>
>
>
>
> Jeff
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
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Re: [OSL | CCIE_Voice] UCCX/IVR Script Repository

[Angel Perez]

Hello:

 

This is the most complet guide

 

Scripting and Development Series: Volume 1 to 3, 7.0(1)

 

You can find it at UCCX documentation area at Cisco

 

hth
 


Date: Fri, 12 Mar 2010 08:43:07 +0200
From: chip...@gmail.com
To: tanner.ez...@gmail.com
CC: ccie_voice@onlinestudylist.com; mjbe...@krollontrack.com
Subject: Re: [OSL | CCIE_Voice] UCCX/IVR Script Repository

a bit out of topic,what material is best for writing the UCCX scripts?


On Thu, Mar 11, 2010 at 9:06 PM, Tanner Ezell  wrote:

C:\program files\wfavvid\Scripts\Templates




On Thu, Mar 11, 2010 at 1:53 PM, Berry, Matthew J.
 wrote:
> Does anyone know of where Cisco’s UCCX/IVR sample script repository is?  I
> can’t find it.
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>



--
Regards,
Tanner Ezell
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