Re: [OSL | CCIE_Voice] CUPS Mailstore

2013-09-17 Thread Edgar Feliz
I don't configure a mail store and it works, did you exit/sign out complete
and log back in?


On Tue, Sep 17, 2013 at 4:40 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote:

  Hi,

 ** **

 My CUPC client for site b phone 1 will NOT show a voicemail was left
 (notification) unless I have the mailstore configured on CUPS Server, I
 thought we didn’t need the mailstore for this to work? 

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Re: [OSL | CCIE_Voice] SiteB CFUR

2013-09-17 Thread Edgar Feliz
If you are talking lab4 than only HQ phones need to call SB phones...SC is
a H323 site and if GK calls fail it will use PSTN..

Edgar


On Tue, Sep 17, 2013 at 9:46 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote:

  Hi,

 ** **

 Been labbing all day and really uncertain with CFUR. If I have Site B in
 SRST mode and Site A and C need to call x30XX how do I accomplish this with
 one RP? Wouldn’t I have to use callED party xform patterns? But still only
 getting Site A to show correct callING party information when call leaves
 SiteA gwy and comes into SiteB gwy, SiteC’s callING party info is the
 problem. 

 ** **

 Uncertain any input would be appreciated? 

 ** **

 ** **

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Re: [OSL | CCIE_Voice] SiteB CFUR

2013-09-17 Thread Edgar Feliz
Right this question requires two called party transforms with PT/CSS for
each GW HQ and SC and set the called party CSS on the GW configuration.

Edgar


On Tue, Sep 17, 2013 at 10:29 PM, Josh Petro josh.pe...@gmail.com wrote:

 Also, the best way to setup outbound dialing with one route pattern is to
 use \+! or \+.!, depending on your requirements. You can then modify
 Called/Calling number in either the Route list or Xforms. I'm not sure if
 that was your question or not though. :)


 On Tue, Sep 17, 2013 at 10:06 PM, Edgar Feliz ejzi...@gmail.com wrote:

 If you are talking lab4 than only HQ phones need to call SB phones...SC
 is a H323 site and if GK calls fail it will use PSTN..

 Edgar


 On Tue, Sep 17, 2013 at 9:46 PM, Barrera, Hugo 
 hugo.barr...@nexusis.comwrote:

  Hi,

 ** **

 Been labbing all day and really uncertain with CFUR. If I have Site B in
 SRST mode and Site A and C need to call x30XX how do I accomplish this with
 one RP? Wouldn’t I have to use callED party xform patterns? But still only
 getting Site A to show correct callING party information when call leaves
 SiteA gwy and comes into SiteB gwy, SiteC’s callING party info is the
 problem. 

 ** **

 Uncertain any input would be appreciated? 

 ** **

 ** **

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Re: [OSL | CCIE_Voice] OWLE lab 4 - Voicemail issue

2013-09-15 Thread Edgar Feliz
If you are using the IPX rack there is a known bug you are hitting.

Edgar


On Sun, Sep 15, 2013 at 1:55 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 hi folks.

 in OWLE lab 4, i can't get the VM  when I left the messasge from PSTN
 phones to HQ and SB phones.

 any parameter to check ?

 K

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Re: [OSL | CCIE_Voice] Softkey template parameter

2013-09-15 Thread Edgar Feliz
Can also refer to iDivert which will transfer your call directly to
voicemail without any other action.

EF


On Sun, Sep 15, 2013 at 4:21 PM, Hesham Abdelkereem 
heshamcentr...@gmail.com wrote:

 Transfer connected is when you have an active call and you find on the
 softkey template it gives you a tone first then you dial the number and
 then transfer this operation is Transfer + Number + Transfer again

 I believe for Transfer direct is when you have a speed dial button
 configured on your phone so when you got a call ringing and you want to
 transfer it directly without making Transfer + Number + Transfer again then
 you just hit the speedial 2 times or hold it for 2 seconds then it will
 transfer immediately




 On 15 September 2013 12:50, Dharambir kumar varma 
 dharambi...@gmail.comwrote:

 Hi All

 can u please guide..
 difference between Transfer/ Direct transfer/connected transfer.   in
 cisco IP Phone Softkey template

 --
  Regards,
  Dharambir Kumar
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Re: [OSL | CCIE_Voice] One Button Login Bulk Subscription

2013-09-04 Thread Edgar Feliz
yes you can,

but what version of CM, there is a bug in one of the recent version that
BAT does not work on.

Edgar


On Wed, Sep 4, 2013 at 1:05 PM, Hesham Abdelkereem heshamcentr...@gmail.com
 wrote:

 Dear Experts,

 I have one of my customers using normal IPPA they manually hit the button
 and enter the information.

 I have proposed One Button Login is way better in this scenario but my
 question is can I make a bulk subscription of the service in some phone and
 can I BAT their information such as User ID , Ext and Pwd?



 Many Thanks,
 Hesham

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Re: [OSL | CCIE_Voice] Generate a report for number of calls into PRI

2013-09-04 Thread Edgar Feliz
TELCO can provide a usage report for each PRI, who is the SP?

Edgar


On Tue, Sep 3, 2013 at 2:23 PM, Hesham Abdelkereem heshamcentr...@gmail.com
 wrote:

 Dear Experts,

 I have 12 PRI configured as MGCP gateways and would like to replace them
 by a CUBE.

 Now, I would like to make Statistics/Feasability study about the number of
 concurrent calls on each PRI for example today from 8am to 5PM.
 Is there is anyway I can do that? That will help me in the calculation to
 order the number of concurrent calls properly when I migrate into SIP.


 Thanks,
 Hesham

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Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore?

2013-08-29 Thread Edgar Feliz
Drake

I can tell you that there is a big push to put more services in the cloud
at major SPs. UCS platforms has made it easier to host multiple customers
in a multi-tenant environment and provide customers their own dedicated
voice and video application customized to each customer.

Video is definitely a growth area the high end video conferencing using
dedicated rooms with equipment worth 200-300k is a niche market, growth is
in the desktop, mobile areas.

EJ


On Thu, Aug 29, 2013 at 7:30 AM, Drake J jdrake...@gmail.com wrote:



 hi Laksh,

 Thanks for your inputs here.This was a good discussion.   It is always
 good for us to all know about things that happen outside  . Talking about
 Telco OTTs we can already see  few of the Telcos have come out  with
 Webrtc solutions for enterprise and service providers .  Check this video
 out too depicting their solution...


 http://www.youtube.com/watch?v=Nz-BQZMp3sk


 Most of these applications written on software are  supposed to  open
 source and left for the users to customize .   No real networking staff
 expertise required   just download the  SDK/API and customize and no more
 complex network topologies in future.  Also no licensing fee too .  Hence a
 real killer  of  techology  in the future  most likely we will see a wide
 spread of this starting 2014 if all predictions are to be believed.


 Hope someone from any of the TELCOs  on this alias can add a few comments
 as well.


 Thanks once again for your inputs everyone.






 On Wed, Aug 28, 2013 at 11:05 PM, Lakshmish NS lakshmish...@gmail.comwrote:

 Hi Drake,

 I totally understand your concern, I'd be worried too. Having said that,
 we should always update ourselves with the latest technology. However, in
 future I believe Asterisk might be able to give tough run to Cisco UC. Not
 sure though, I hear stories that it is unstable and featureless compared to
 CUCM. I hope if someone aware of Asterisk would help us out here.

 Regard,

 Laksh




 On Wed, Aug 28, 2013 at 9:56 PM, Drake J jdrake...@gmail.com wrote:

 Hi Guys,

 Thanks for your responses  I see u guys have empathized on call routing
 and and UC hardware for backend deployments.  However Telco OTTs are coming
 up with directly provide these services over the cloud . Here is a
 disruptive analysis :


 http://www.slideshare.net/deanb/disruptive-analysis-web-rtc-overview-april-2013


 Anyways, this might be not be so serious afterall . Just thought of
 brainstorming  .

 Thanks guys for your responses again.



 On Tue, Aug 27, 2013 at 6:20 PM, Lakshmish NS lakshmish...@gmail.comwrote:

 Didn't have time to go through the video, I believe WebRTC is nothing
 but a Protocol, similar to SIP, H.323. Moreover, this protocol would only
 appeal to the Web audience, just like Skype, or Google talk. You still need
 to use UC hardware and their design for enterprise deployments. I mean we
 don't use Google talk and Skype in companies right? SIP is open source, but
 still Cisco uses it. As FAQ's suggest WebRTC is an open framework for
 the web that enables Real Time Communications in the browser. If only UC
 was that easy that could be implemented through browser, we didn't have to
 work this hard for CCIE numbers. You might want to go through this...
 http://www.webrtc.org/faq

 You've clearly misinterpreted WebRTC here..


 On Tue, Aug 27, 2013 at 5:17 PM, Drake J jdrake...@gmail.com wrote:


 hi All,


 Had a troubling question hence thought of putting it out .Looking at
 the UC and networking trends worldwide it appears that
 the future of UC and collaboration is web based. Webrtc is
 the protocol that the world will use and individuals and organizations
 just need to code their requirement based on the WEBRTC.

 Here is the presentation that Google recently made

 http://www.youtube.com/watch?v=E8C8ouiXHHk


 Clearly many of the UC vendors are already losing out and will be
 losing out in year 2014.


 Most of the customers are already looking at reducing the cost involved
 in maintaining costly UC vendor networks and their networking staff .


 Therefore that brings me to my question is the CCIE voice worth
 anymore?


 -Drake



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Re: [OSL | CCIE_Voice] Mailbox Transfer

2013-08-25 Thread Edgar Feliz
I just put the VM ports in my HQ CSS and did not have to make any other
changes.

Edgar


On Sun, Aug 25, 2013 at 4:52 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote:

  Hi,

 ** **

 When a caller from the pstn hits the SiteA User 1 mailbox and can press 9
 to dial out I usually create a vm-css that only has two partitions my
 sa-pt and my meetme-pt. The call should work because it uses the
 sa-ptroute pattern correct? Or am I missing something here? 

 ** **

 - Hugo 

 ** **

 ** **

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Re: [OSL | CCIE_Voice] MGCP fail over

2013-07-29 Thread Edgar Feliz
It should be valid if the Question asked for a specific order for that GW.

EF


On Mon, Jul 29, 2013 at 1:02 PM, Alex Mendoza aa.mend...@icloud.com wrote:

 Hi All

 I configured BR1 as a MGCP gateway, and is working good.

 Call takes Bchan 1 on incoming and outgoing calls.

 My doubts are with mgcp-fallback. On SRST mode, outgoing calls take bchan
 3. I used the isdn bchan-number-order ascending to fix this.

 It's valid use this command or there is nothing to complain if I used this
 command or not.

 thanks.

 AA

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Re: [OSL | CCIE_Voice] call-manager fall back issue

2013-07-28 Thread Edgar Feliz
Bug!!! IOU have to restart the router. 

Edgar

Sent from my iPhone

On Jul 28, 2013, at 9:10 AM, Todd Carswell tcar0...@gmail.com wrote:

 Were the phones registered to CM with DNs assigned BEFORE they failed over to 
 SRST?  
 
 --Todd
 
 On Jul 28, 2013, at 5:56, IE Target myfrnd...@gmail.com wrote:
 
 Facing this issue plz help
 
 When  sb phone fall back to SRSt(call-manager fallback)
 
 They show registered in show ephone
 BUT
 they do not get any dn
 I tried reloading router, factory reset and every thing
 
 What is the solution . Has any one faced problem like this
 plz reply
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Re: [OSL | CCIE_Voice] call-manager fall back issue

2013-07-28 Thread Edgar Feliz
Register phones back to CM before restarting. 

Sent from my iPhone

On Jul 28, 2013, at 9:10 AM, Todd Carswell tcar0...@gmail.com wrote:

 Were the phones registered to CM with DNs assigned BEFORE they failed over to 
 SRST?  
 
 --Todd
 
 On Jul 28, 2013, at 5:56, IE Target myfrnd...@gmail.com wrote:
 
 Facing this issue plz help
 
 When  sb phone fall back to SRSt(call-manager fallback)
 
 They show registered in show ephone
 BUT
 they do not get any dn
 I tried reloading router, factory reset and every thing
 
 What is the solution . Has any one faced problem like this
 plz reply
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Re: [OSL | CCIE_Voice] Fast Start

2013-07-13 Thread Edgar Feliz
For F/S on site B you do not need the transcoder just MTP. Typical : max
session 100 would work.




On Sat, Jul 13, 2013 at 1:16 AM, IE Target myfrnd...@gmail.com wrote:

 Ya
 MRGL_FS Contains MRG_FS which contains SB_Xcode and SB_MTP (G711)

 dspfarm profile 1 mtp
  associate application sccp
  codec g711ulaw
  maximum session software 4
  no shut
 dspfarm profile 2 transcode
   associate application sccp
   codec g729r8
   maximum session 2
   no shut


 MRGL_FS is applied to only  h323 GW. It works for me

 But My question is can we use only transcoder , i.e MRG_FS contains only
 XCode not MTP ?
 Other part is how many session to assign in dspfarm profile


 On Sat, Jul 13, 2013 at 1:57 AM, Karen Johnson 
 karen.johnson...@yahoo.cawrote:

 hi friend,

 may i see your IOS config for this ?

 and also what in inside MRG-SB and MRGL-SB

 you only assign MRGL SB to GW h323, right ?

 K

   *From:* myfriend myfriend myfrnd...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com
 *Sent:* Friday, July 12, 2013 12:45:35 AM
 *Subject:* Fast Start

  These are my finding

 No need to enable fast start on Router , by default it will use Fast Start

 Assgn MRGL to SB Gw which contain Only G711 MTP (and transcoder)

 Check MTP Required
 Check Outbound FS
 Check Inbound FS

 That it
 Am i missing any thing




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Re: [OSL | CCIE_Voice] Fast Start

2013-07-12 Thread Edgar Feliz
Transcoder?

you mean MTP

Edgar


On Fri, Jul 12, 2013 at 3:45 PM, Karen Johnson karen.johnson...@yahoo.cawrote:


 tks ,

 so what is the Transcoder use for Fast start here?

 K
   *From:* myfriend myfriend myfrnd...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com
 *Sent:* Friday, July 12, 2013 12:45:35 AM
 *Subject:* Fast Start

  These are my finding

 No need to enable fast start on Router , by default it will use Fast Start

 Assgn MRGL to SB Gw which contain Only G711 MTP (and transcoder)

 Check MTP Required
 Check Outbound FS
 Check Inbound FS

 That it
 Am i missing any thing



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Re: [OSL | CCIE_Voice] CUPC client to play voicemail

2013-07-09 Thread Edgar Feliz
Also make sure that you have expanded that section in the client.




On Mon, Jul 8, 2013 at 4:48 PM, ccie2k12 ccie2...@gmail.com wrote:

  check server health in CUPC, to make voice mail is showing green

 also have you entered the voice mail user id and password in preferences**
 **

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Drake J
 *Sent:* Monday, July 08, 2013 9:46 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] CUPC client to play voicemail

 ** **

 hi Guys,

 How do we get our CUPC client to play voicemail and display it on the
 client as a red Icon?

 I tried many times but not see this icon.

 -Drake

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Re: [OSL | CCIE_Voice] H323 Fast Start

2013-07-08 Thread Edgar Feliz
I may be wrong but I do not recall g729 being a requirement. for that task,
they are not asking HQ  SB, which would be inter-region, just F/S between
CM  R2 = SB phones dial PSTN.


Edgar


On Mon, Jul 8, 2013 at 8:03 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote:

  Guy’s,

 ** **

 I used g711 for the codec, on mtp resource, but a peer of mine is stating
 I should have used g729 and by using this I would have to add a transcoder
 on the GW as well. Any thoughts? 

 ** **

 *Hugo *

 ** **

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Re: [OSL | CCIE_Voice] fast start H323 failed

2013-07-04 Thread Edgar Feliz
Karen,

Here is is the sh call active voice output from a call from pstn to my
SB-Ph1

1- R2 -I do not do anything with voice service voip - I call start fast
is default you do not need to do anything.
2- CM- enabled inbound and outbound F/S

R2#show clock
16:08:38.881 DST Thu Jul 4 2013
R2#sh call active voice
Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Media call-legs: 0
Total call-legs: 2

 GENERIC:
SetupTime=1340160 ms
Index=1
PeerAddress=
PeerSubAddress=
PeerId=1
PeerIfIndex=46
LogicalIfIndex=12
ConnectTime=1342160 ms
CallDuration=00:01:05 sec
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=3283
TransmitBytes=551544
ReceivePackets=3279
ReceiveBytes=524640
TELE:
ConnectionId=[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
IncomingConnectionId=[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
CallID=21
Port=0/0/0:23 (21)
BearerChannel=0/0/0.1
TxDuration=65580 ms
VoiceTxDuration=65580 ms
FaxTxDuration=0 ms
CoderTypeRate=g711ulaw
NoiseLevel=-48
ACOMLevel=9
OutSignalLevel=-46
InSignalLevel=-44
InfoActivity=1
ERLLevel=9
SessionTarget=
ImgPages=0
CallerName=Emergency Services
CallerIDBlocked=False
LongDurationCallDetected=no
LongDurCallTimeStamp=
LongDurCallDuration=
OriginalCallingNumber=
OriginalCallingOctet=0x0
OriginalCalledNumber=4083873001
OriginalCalledOctet=0xA1
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=
TranslatedCallingOctet=0x0
TranslatedCalledNumber=3001
TranslatedCalledOctet=0xA1
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=4083873001
GwReceivedCalledOctet3=0xA1
DSPIdentifier=0/1:1

 GENERIC:
SetupTime=1340390 ms
Index=1
PeerAddress=3001
PeerSubAddress=
PeerId=3001
PeerIfIndex=52
LogicalIfIndex=0
ConnectTime=1342140 ms
CallDuration=00:01:06 sec
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=speech
TransmitPackets=3309
TransmitBytes=529440
ReceivePackets=3313
ReceiveBytes=530080
VOIP:
ConnectionId[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
IncomingConnectionId[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
CallID=23
RemoteIPAddress=192.168.1.50
RemoteUDPPort=16578
RemoteSignallingIPAddress=192.168.1.50
RemoteSignallingPort=1720
RemoteMediaIPAddress=10.10.201.120
RemoteMediaPort=16578
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=h245-alphanumeric
FastConnect=TRUE  Fast-Start 

AnnexE=FALSE

Separate H245 Connection=TRUE

H245 Tunneling=FALSE

SessionProtocol=cisco
ProtocolCallId=
SessionTarget=ipv4:192.168.1.50
OnTimeRvPlayout=66110
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=65 ms
LoWaterPlayoutDelay=55 ms
TxPakNumber=3308
TxSignalPak=0
TxComfortNoisePak=0
TxDuration=66160
TxVoiceDuration=66160
RxPakNumber=3310
RxSignalPak=0
RxComfortNoisePak=0
RxDuration=66160
RxVoiceDuration=66110
RxOutOfSeq=0
RxLatePak=0
RxEarlyPak=1
RxBadProtocol=0
PlayDelayCurrent=55
PlayDelayMin=55
PlayDelayMax=65
PlayDelayClockOffset=-1506743064
PlayDelayJitter=0 ms
PlayErrPredictive=0
PlayErrInterpolative=0
PlayErrSilence=0
PlayErrBufferOverFlow=30
PlayErrRetroactive=2
PlayErrTalkspurt=0
OutSignalLevel=-50
InSignalLevel=-48
LevelTxPowerMean=0
LevelRxPowerMean=0
LevelBgNoise=0
ERLLevel=9
ACOMLevel=9
ErrRxDrop=0
ErrTxDrop=0
ErrTxControl=0
ErrRxControl=0
ReceiveDelay=55 ms
LostPackets=0
EarlyPackets=1
LatePackets=0
SRTP = off
TextRelay = off
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
Media Setting=flow-through
AlertTimepoint=1340730 ms
CallerName=Emergency Services
CallerIDBlocked=False
OriginalCallingNumber=
OriginalCallingOctet=0x0
OriginalCalledNumber=3001
OriginalCalledOctet=0xA1
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=
TranslatedCallingOctet=0x0
TranslatedCalledNumber=3001
TranslatedCalledOctet=0xA1
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=4083873001
GwReceivedCalledOctet3=0xA1
GwOutpulsedCalledNumber=3001
GwOutpulsedCalledOctet3=0xA1
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurCallTimestamp=
LongDurcallDuration=
Username=

Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Media call-legs: 0
Total call-legs: 2

R2#



On Thu, Jul 4, 2013 at 4:28 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 hi Edgar and all,

 Here is what I did and still not able to see Fast Connect= True. (still
 show FALSE)

 - MRM SB : SB-xcode  (also try with MTP)
 - MRGL SB assigned to SB-GW H323.
 - checked Inbound Fast start   and Outbound Fast Start

 when i call from SB phone to 911 and vice versa. I did not see  Fast Start.

 - Can you write correct detailed steps here?

 tks


   --
  *From:* Edgar Feliz ejzi...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca

Re: [OSL | CCIE_Voice] fast start H323 failed

2013-07-04 Thread Edgar Feliz
Karen,

From what I hear you have to show both inbound and outbound F/S: I would
check both to be safe.

Edgar


On Thu, Jul 4, 2013 at 8:35 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 hi Edgar,


 you right, i see it now.

 Question :

 - CM- enabled inbound and outbound F/S , do we need to check both ?  as I
 try to check one and it also works.
 - for test call, incoming to Ip Phone or outgoing doesn't matter ,right ?
 What they want in exam  In or out ?

 tks
 k

   --
  *From:* Edgar Feliz ejzi...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Thursday, July 4, 2013 3:12:00 PM
 *Subject:* Re: fast start H323 failed

 Karen,

 Here is is the sh call active voice output from a call from pstn to my
 SB-Ph1

 1- R2 -I do not do anything with voice service voip - I call start fast
 is default you do not need to do anything.
 2- CM- enabled inbound and outbound F/S

 R2#show clock
 16:08:38.881 DST Thu Jul 4 2013
 R2#sh call active voice
 Telephony call-legs: 1
 SIP call-legs: 0
 H323 call-legs: 1
 Call agent controlled call-legs: 0
 SCCP call-legs: 0
 Multicast call-legs: 0
 Media call-legs: 0
 Total call-legs: 2

  GENERIC:
 SetupTime=1340160 ms
 Index=1
 PeerAddress=
 PeerSubAddress=
 PeerId=1
 PeerIfIndex=46
 LogicalIfIndex=12
 ConnectTime=1342160 ms
 CallDuration=00:01:05 sec
 CallState=4
 CallOrigin=2
 ChargedUnits=0
 InfoType=speech
 TransmitPackets=3283
 TransmitBytes=551544
 ReceivePackets=3279
 ReceiveBytes=524640
 TELE:
 ConnectionId=[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 IncomingConnectionId=[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 CallID=21
 Port=0/0/0:23 (21)
 BearerChannel=0/0/0.1
 TxDuration=65580 ms
 VoiceTxDuration=65580 ms
 FaxTxDuration=0 ms
 CoderTypeRate=g711ulaw
 NoiseLevel=-48
 ACOMLevel=9
 OutSignalLevel=-46
 InSignalLevel=-44
 InfoActivity=1
 ERLLevel=9
 SessionTarget=
 ImgPages=0
 CallerName=Emergency Services
 CallerIDBlocked=False
 LongDurationCallDetected=no
 LongDurCallTimeStamp=
 LongDurCallDuration=
 OriginalCallingNumber=
 OriginalCallingOctet=0x0
 OriginalCalledNumber=4083873001
 OriginalCalledOctet=0xA1
 OriginalRedirectCalledNumber=
 OriginalRedirectCalledOctet=0xFF
 TranslatedCallingNumber=
 TranslatedCallingOctet=0x0
 TranslatedCalledNumber=3001
 TranslatedCalledOctet=0xA1
 TranslatedRedirectCalledNumber=
 TranslatedRedirectCalledOctet=0xFF
 GwReceivedCalledNumber=4083873001
 GwReceivedCalledOctet3=0xA1
 DSPIdentifier=0/1:1

  GENERIC:
 SetupTime=1340390 ms
 Index=1
 PeerAddress=3001
 PeerSubAddress=
 PeerId=3001
 PeerIfIndex=52
 LogicalIfIndex=0
 ConnectTime=1342140 ms
 CallDuration=00:01:06 sec
 CallState=4
 CallOrigin=1
 ChargedUnits=0
 InfoType=speech
 TransmitPackets=3309
 TransmitBytes=529440
 ReceivePackets=3313
 ReceiveBytes=530080
 VOIP:
 ConnectionId[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 IncomingConnectionId[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 CallID=23
 RemoteIPAddress=192.168.1.50
 RemoteUDPPort=16578
 RemoteSignallingIPAddress=192.168.1.50
 RemoteSignallingPort=1720
 RemoteMediaIPAddress=10.10.201.120
 RemoteMediaPort=16578
 RoundTripDelay=0 ms
 SelectedQoS=best-effort
 tx_DtmfRelay=h245-alphanumeric
 FastConnect=TRUE  Fast-Start 

 AnnexE=FALSE

 Separate H245 Connection=TRUE

 H245 Tunneling=FALSE

 SessionProtocol=cisco
 ProtocolCallId=
 SessionTarget=ipv4:192.168.1.50
 OnTimeRvPlayout=66110
 GapFillWithSilence=0 ms
 GapFillWithPrediction=0 ms
 GapFillWithInterpolation=0 ms
 GapFillWithRedundancy=0 ms
 HiWaterPlayoutDelay=65 ms
 LoWaterPlayoutDelay=55 ms
 TxPakNumber=3308
 TxSignalPak=0
 TxComfortNoisePak=0
 TxDuration=66160
 TxVoiceDuration=66160
 RxPakNumber=3310
 RxSignalPak=0
 RxComfortNoisePak=0
 RxDuration=66160
 RxVoiceDuration=66110
 RxOutOfSeq=0
 RxLatePak=0
 RxEarlyPak=1
 RxBadProtocol=0
 PlayDelayCurrent=55
 PlayDelayMin=55
 PlayDelayMax=65
 PlayDelayClockOffset=-1506743064
 PlayDelayJitter=0 ms
 PlayErrPredictive=0
 PlayErrInterpolative=0
 PlayErrSilence=0
 PlayErrBufferOverFlow=30
 PlayErrRetroactive=2
 PlayErrTalkspurt=0
 OutSignalLevel=-50
 InSignalLevel=-48
 LevelTxPowerMean=0
 LevelRxPowerMean=0
 LevelBgNoise=0
 ERLLevel=9
 ACOMLevel=9
 ErrRxDrop=0
 ErrTxDrop=0
 ErrTxControl=0
 ErrRxControl=0
 ReceiveDelay=55 ms
 LostPackets=0
 EarlyPackets=1
 LatePackets=0
 SRTP = off
 TextRelay = off
 VAD = disabled
 CoderTypeRate=g711ulaw
 CodecBytes=160
 Media Setting=flow-through
 AlertTimepoint=1340730 ms
 CallerName=Emergency Services
 CallerIDBlocked=False
 OriginalCallingNumber=
 OriginalCallingOctet=0x0
 OriginalCalledNumber=3001
 OriginalCalledOctet=0xA1
 OriginalRedirectCalledNumber=
 OriginalRedirectCalledOctet=0xFF
 TranslatedCallingNumber=
 TranslatedCallingOctet=0x0
 TranslatedCalledNumber=3001
 TranslatedCalledOctet=0xA1
 TranslatedRedirectCalledNumber=
 TranslatedRedirectCalledOctet=0xFF
 GwReceivedCalledNumber=4083873001
 GwReceivedCalledOctet3=0xA1
 GwOutpulsedCalledNumber=3001

Re: [OSL | CCIE_Voice] fast start H323 failed

2013-07-04 Thread Edgar Feliz
Karen,

I would get familiar with all the options to acquire the results.

GW - show call active voice and debugh225 asn1
CUCM enable h225/245 trace and tail the sdi recent log on CLI or use RTMT

For MTP/DSP there is was a good forum discussion a while back you can
search for it.

Edgar


On Thu, Jul 4, 2013 at 8:45 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 Thanks Edgar, what is your detailed dspfarm profile mtp look like ?
 -how many session needed ?  I also curios what factor determine max
 session for xcode, cfb, this mtp ?
 -so sh call acti voice is not enough, we still need to debug h225 asn
 1 ?

 tks
 K



   --
  *From:* Edgar Feliz ejzi...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Thursday, July 4, 2013 6:41:32 PM

 *Subject:* Re: fast start H323 failed

 Karen,

 From what I hear you have to show both inbound and outbound F/S: I would
 check both to be safe.

 Edgar


 On Thu, Jul 4, 2013 at 8:35 PM, Karen Johnson 
 karen.johnson...@yahoo.cawrote:

 hi Edgar,


 you right, i see it now.

 Question :

 - CM- enabled inbound and outbound F/S , do we need to check both ?  as I
 try to check one and it also works.
 - for test call, incoming to Ip Phone or outgoing doesn't matter ,right ?
 What they want in exam  In or out ?

 tks
 k

   --
  *From:* Edgar Feliz ejzi...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Thursday, July 4, 2013 3:12:00 PM
 *Subject:* Re: fast start H323 failed

 Karen,

 Here is is the sh call active voice output from a call from pstn to my
 SB-Ph1

 1- R2 -I do not do anything with voice service voip - I call start fast
 is default you do not need to do anything.
 2- CM- enabled inbound and outbound F/S

 R2#show clock
 16:08:38.881 DST Thu Jul 4 2013
 R2#sh call active voice
 Telephony call-legs: 1
 SIP call-legs: 0
 H323 call-legs: 1
 Call agent controlled call-legs: 0
 SCCP call-legs: 0
 Multicast call-legs: 0
 Media call-legs: 0
 Total call-legs: 2

  GENERIC:
 SetupTime=1340160 ms
 Index=1
 PeerAddress=
 PeerSubAddress=
 PeerId=1
 PeerIfIndex=46
 LogicalIfIndex=12
 ConnectTime=1342160 ms
 CallDuration=00:01:05 sec
 CallState=4
 CallOrigin=2
 ChargedUnits=0
 InfoType=speech
 TransmitPackets=3283
 TransmitBytes=551544
 ReceivePackets=3279
 ReceiveBytes=524640
 TELE:
 ConnectionId=[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 IncomingConnectionId=[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 CallID=21
 Port=0/0/0:23 (21)
 BearerChannel=0/0/0.1
 TxDuration=65580 ms
 VoiceTxDuration=65580 ms
 FaxTxDuration=0 ms
 CoderTypeRate=g711ulaw
 NoiseLevel=-48
 ACOMLevel=9
 OutSignalLevel=-46
 InSignalLevel=-44
 InfoActivity=1
 ERLLevel=9
 SessionTarget=
 ImgPages=0
 CallerName=Emergency Services
 CallerIDBlocked=False
 LongDurationCallDetected=no
 LongDurCallTimeStamp=
 LongDurCallDuration=
 OriginalCallingNumber=
 OriginalCallingOctet=0x0
 OriginalCalledNumber=4083873001
 OriginalCalledOctet=0xA1
 OriginalRedirectCalledNumber=
 OriginalRedirectCalledOctet=0xFF
 TranslatedCallingNumber=
 TranslatedCallingOctet=0x0
 TranslatedCalledNumber=3001
 TranslatedCalledOctet=0xA1
 TranslatedRedirectCalledNumber=
 TranslatedRedirectCalledOctet=0xFF
 GwReceivedCalledNumber=4083873001
 GwReceivedCalledOctet3=0xA1
 DSPIdentifier=0/1:1

  GENERIC:
 SetupTime=1340390 ms
 Index=1
 PeerAddress=3001
 PeerSubAddress=
 PeerId=3001
 PeerIfIndex=52
 LogicalIfIndex=0
 ConnectTime=1342140 ms
 CallDuration=00:01:06 sec
 CallState=4
 CallOrigin=1
 ChargedUnits=0
 InfoType=speech
 TransmitPackets=3309
 TransmitBytes=529440
 ReceivePackets=3313
 ReceiveBytes=530080
 VOIP:
 ConnectionId[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 IncomingConnectionId[0x9410023C 0xE42411E2 0x80060025 0x45D31A08]
 CallID=23
 RemoteIPAddress=192.168.1.50
 RemoteUDPPort=16578
 RemoteSignallingIPAddress=192.168.1.50
 RemoteSignallingPort=1720
 RemoteMediaIPAddress=10.10.201.120
 RemoteMediaPort=16578
 RoundTripDelay=0 ms
 SelectedQoS=best-effort
 tx_DtmfRelay=h245-alphanumeric
 FastConnect=TRUE  Fast-Start 

 AnnexE=FALSE

 Separate H245 Connection=TRUE

 H245 Tunneling=FALSE

 SessionProtocol=cisco
 ProtocolCallId=
 SessionTarget=ipv4:192.168.1.50
 OnTimeRvPlayout=66110
 GapFillWithSilence=0 ms
 GapFillWithPrediction=0 ms
 GapFillWithInterpolation=0 ms
 GapFillWithRedundancy=0 ms
 HiWaterPlayoutDelay=65 ms
 LoWaterPlayoutDelay=55 ms
 TxPakNumber=3308
 TxSignalPak=0
 TxComfortNoisePak=0
 TxDuration=66160
 TxVoiceDuration=66160
 RxPakNumber=3310
 RxSignalPak=0
 RxComfortNoisePak=0
 RxDuration=66160
 RxVoiceDuration=66110
 RxOutOfSeq=0
 RxLatePak=0
 RxEarlyPak=1
 RxBadProtocol=0
 PlayDelayCurrent=55
 PlayDelayMin=55
 PlayDelayMax=65
 PlayDelayClockOffset=-1506743064
 PlayDelayJitter=0 ms
 PlayErrPredictive=0
 PlayErrInterpolative=0
 PlayErrSilence=0
 PlayErrBufferOverFlow=30

Re: [OSL | CCIE_Voice] fast start H323

2013-07-03 Thread Edgar Feliz
Also on CM side with h245/h225 trace on this is the out F/S

05/21/2013 19:10:04.653 CCM|H225Cdpc::null0_CcSetupReq(3, 39229128):
FastStartInfo: Outbound FastStart  Triggered - PreAllocating
MTP|CLID::StandAloneClusterNID::192.168.1.51CT::2,100,37,1.241129IP::10.10.202.121DEV::SEPD4D748FE5095LVL::State
TransitionMASK::0100

There is a similar message for inbound.

Edgar


On Wed, Jul 3, 2013 at 8:20 PM, Edgar Feliz ejzi...@gmail.com wrote:

 Karen See below for fast connect... Make sure you are looking at the
 section that has the RemoteIPAddress with the IP address of the CM. Also
 did you enable F/S on the CM?

 VOIP:
 ConnectionId[0x43E821AF 0xB06B11E2 0x80020025 0x45D31A08]
 IncomingConnectionId[0x43E821AF 0xB06B11E2 0x80020025 0x45D31A08]
 CallID=7
 RemoteIPAddress=192.168.1.51
 RemoteUDPPort=17628
 RemoteSignallingIPAddress=192.168.1.51
 RemoteSignallingPort=1720
 RemoteMediaIPAddress=10.10.110.2
 RemoteMediaPort=17628
 RoundTripDelay=0 ms
 SelectedQoS=best-effort
 tx_DtmfRelay=h245-alphanumeric
 FastConnect=TRUE   fast Start


 Edgar


 On Wed, Jul 3, 2013 at 7:20 PM, Karen Johnson 
 karen.johnson...@yahoo.cawrote:

 experts,

 i have configure H323 GW to be  Fast start  by using Transcoder and
 enable fast start under service voip in IOS GW.

 however when i make call and verified using  sh call acti voice and
 looking at Fast Connect= False.

 - is there any thing else needed to make this work ?
 - what is the right steps to implement this  :  Inbound Fast Start to UCM
 and Outbound Fast Start from UCM ?

 tks
 K







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Re: [OSL | CCIE_Voice] fast start H323

2013-07-03 Thread Edgar Feliz
Karen See below for fast connect... Make sure you are looking at the
section that has the RemoteIPAddress with the IP address of the CM. Also
did you enable F/S on the CM?

VOIP:
ConnectionId[0x43E821AF 0xB06B11E2 0x80020025 0x45D31A08]
IncomingConnectionId[0x43E821AF 0xB06B11E2 0x80020025 0x45D31A08]
CallID=7
RemoteIPAddress=192.168.1.51
RemoteUDPPort=17628
RemoteSignallingIPAddress=192.168.1.51
RemoteSignallingPort=1720
RemoteMediaIPAddress=10.10.110.2
RemoteMediaPort=17628
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=h245-alphanumeric
FastConnect=TRUE   fast Start


Edgar


On Wed, Jul 3, 2013 at 7:20 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 experts,

 i have configure H323 GW to be  Fast start  by using Transcoder and
 enable fast start under service voip in IOS GW.

 however when i make call and verified using  sh call acti voice and
 looking at Fast Connect= False.

 - is there any thing else needed to make this work ?
 - what is the right steps to implement this  :  Inbound Fast Start to UCM
 and Outbound Fast Start from UCM ?

 tks
 K







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

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Re: [OSL | CCIE_Voice] IPCC extension not showing in CUCM a nd also not showing rmcm user

2013-07-01 Thread Edgar Feliz
It is a bug.
CLI to Pub and enter this command:

run sql update processconfig set paramvalue=T where paramname like
'%nstalled%'


Do this before you go into users I do it in the beginning of the labs since
it is always an issue with the proctor labs Racks

EF


On Mon, Jul 1, 2013 at 12:04 AM, Amit Sharma aryan231...@gmail.com wrote:

 guys,

 anyone using proctor lab and use this issue?

 iPCC extension not showing...

 rmcm not showing under cucm?


 how can fix it?

 --
 Thanks  Regard's
 Amit Sharma


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Re: [OSL | CCIE_Voice] proctorlabs replication

2013-05-23 Thread Edgar Feliz
Check DB is replication.

Edgar


On Thu, May 23, 2013 at 3:21 PM, CCIEing aboaz...@gmail.com wrote:

 Did you try to disable csa?


 On Thu, May 23, 2013 at 8:21 PM, Karen Johnson 
 karen.johnson...@yahoo.cawrote:

 hi all,

 I used auto phone register in proctorlabs. When UCM group start with SUB,
 phone never registered.
 But when I move PUB to 1st Server in Group, phone registered fine.

 And I also checked in Cisco CallManager Reporting for DB summary:
 replication look good 2 and CLI also look good

 any idea what wrong and what is command to check in this case?

 tks

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 For more information regarding industry leading CCIE Lab training, please
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Re: [OSL | CCIE_Voice] proctorlabs replication

2013-05-23 Thread Edgar Feliz
cli = util db status

gui = Cisco Unified reporting  DB replication status

Edgar


On Thu, May 23, 2013 at 4:20 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 u mean show utils service  ?

   *From:* CCIEing aboaz...@gmail.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Thursday, May 23, 2013 1:21:36 PM
 *Subject:* Re: [OSL | CCIE_Voice] proctorlabs replication

  Did you try to disable csa?


 On Thu, May 23, 2013 at 8:21 PM, Karen Johnson 
 karen.johnson...@yahoo.cawrote:

  hi all,

 I used auto phone register in proctorlabs. When UCM group start with SUB,
 phone never registered.
 But when I move PUB to 1st Server in Group, phone registered fine.

 And I also checked in Cisco CallManager Reporting for DB summary:
 replication look good 2 and CLI also look good

 any idea what wrong and what is command to check in this case?

 tks

 ___
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Re: [OSL | CCIE_Voice] DistinctiveRingList in SRST

2013-05-08 Thread Edgar Feliz
Have you tried adding that file to the SRST RTR and doing TFTP-server etc.
from that RTR.

Edgar


On Wed, May 8, 2013 at 11:15 AM, Suresh Bhandari bring...@gmail.com wrote:

 Hello Experts!

 Anyone tried using DistinctiveRingList.xml in CME SRST to give the IP
 Phone DN a specific list of ring lists?

 I am trying it, and everytime I am getting Ring List Unavailable message
 on my IP Phone in SRST mode.

 Simple config is used:

 ephone-dn tag
 ring internal

 copied DistinctiveRingList.xml to flash, and enabled tftp-server for the
 same file.

 Anything am I missing?

 TIA
 --
 Suresh Bhandari

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Re: [OSL | CCIE_Voice] phone association

2013-04-29 Thread Edgar Feliz
BAT file.

EJF


On Mon, Apr 29, 2013 at 12:05 AM, ikizoo hello ikiz...@hotmail.com wrote:

 Hi All
 is there any easy way to associate a phone to user, i am doing it
 manually, and it takes a bit of time.
 -ikizoo

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Re: [OSL | CCIE_Voice] Back to back session IPexperts racks

2013-04-16 Thread Edgar Feliz
I've done it many times just ask them to join your sessions, the support
team has always been great about doing it when you give them enough time
and if racks are available

Edgar


On Tue, Apr 16, 2013 at 6:58 AM, Bill whl...@gmail.com wrote:

 You need to verify as in the past this did not always work.  Not sure how
 the new scheduling works

 Sent from my iPad

 On Apr 15, 2013, at 10:27 PM, Mohamed Gazzaz mgaz...@hotmail.com wrote:

 The two sessions are linked automatically when you book B2B sessions and
 your configuration stays.

 You can ask support for more clarification.

 --
 Date: Tue, 16 Apr 2013 04:43:16 +0300
 From: aboaz...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Back to back session IPexperts racks

 Dears,,

 When I reserve a back to back racks sessions , does the pod number changed?

 does my configuration removed one I go from one time session to the next
 one

 Appreciated

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 regarding industry leading CCIE Lab training, please visit
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Re: [OSL | CCIE_Voice] QUESTION FOR THE LAB

2012-11-22 Thread Edgar Feliz
Unless it is specifically stated to use one or the other, use whatever you
feel comfortable with as long as the tasks are completed correctly.

Edgar

On Thu, Nov 22, 2012 at 9:05 AM, Chrysostomos Christofi 
ch.christ...@logicom.net wrote:

  Hi

 ** **

 If we have cue in the lab then we CAN proceed with the initialization
 through the web interface?

 Or we have to proceed through cli

 ** **

 Or you can go with both ways

 ** **

 I hope that my question is not violate  any policies !

 ** **

 Regards

 cc

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Re: [OSL | CCIE_Voice] No Services Configured

2012-11-17 Thread Edgar Feliz
Bill,

This works great for deleting services and changing the order that the
directories are listed in, but if you are give a picture that shows the No
Services Configured  just above the soft-keys that only appears when you
follow Vic's method, from what I have been able to see working this out
both ways.

Edgar

On Fri, Nov 16, 2012 at 7:44 PM, Bill Lake whl...@gmail.com wrote:

 Hello

 I think if you just have to have it say No Services Configured, it is OK
 to delete the Eternal URL as IPExpert says.  Set SA PH 1 to external and
 the others are internal.  Internal will just use the call manager and you
 shouldn't have issues with this solution.

 If you are doing it to make it the same as what is the default and perhaps
 block one service (like corporate directory) you could do it like this

 Log into the UCCX server.  Create a file as shown below but adjust your
 order as on my phones I know Corporate directory is last, I might have
 the others in the wrong order.


 CiscoIPPhoneMenu
 MenuItem
 NameMissed Calls/Name
 URLApplication:Cisco/MissedCalls/URL
 /MenuItem
 MenuItem
 NameReceived Calls/Name
 URLApplication:Cisco/ReceivedCalls/URL
 /MenuItem
 MenuItem
 NamePlaced Calls/Name
 URLApplication:Cisco/PlacedCalls/URL
 /MenuItem
 MenuItem
 NamePersonal Directory/Name
 URLApplication:Cisco/PersonalDirectory/URL
 /MenuItem
 MenuItem
 NameCorporateDirectory/Name
 URLhttp://10.10.210.5/directory.html/URL
 /MenuItem
 /CiscoIPPhoneMenu

 Save this file as directory.xml and store it in c:\inetpub\wwwroot

 Change the service URL in Enterprise to http://10.10.210.5/directory.xml

 now create a new file and just put whatever message you want it to say

 No Directory Services

  now save that as directory.html and store it in c:\inetpub\wwwroot



 On Fri, Nov 16, 2012 at 5:59 PM, Pixar Perfect pixarperf...@live.comwrote:


 I am working on the question from the lab experience handbook where SA
 phone 1 needs to see No Services Configured when it hits directory
 button. IPEXPERT suggests that we delete the Service URL from EP and then
 setup the SA Phone 1 device configuration page Service parameter to
 External ? I dont think it is acceptable solution as this could cause other
 phones to not see any services if they reboot and download new config file
 which has no directories on any phone !!!

 My approach as follows
 1) make a directory.xml file using the IP Phone Services document from
 cisco.com
 2) delete the Missed, Placed, Corporate, Placed and Received from the
 Phones  IP Phone Services page
 3) Setup the EP for Services to Both  Update the URL to
 http://x.x.x.x/directory.xml where x.x.x.x is UCCX IP
 4) Upload the directory.xml to the IIS server on the UCCX, restart IIS
 5) on SA Phone 1 page, setup the Services to Internal, Reset all phones


 Comments? Feedback? Experiences? TIA.


 directory.xml

 CiscoIPPhoneMenu
   MenuItem
 NameMissed Calls/Name
 URLApplication:Cisco/MissedCalls/URL
   /MenuItem
   MenuItem
 NameCorporateDirectory/Name
 URLApplication:Cisco/CorporateDirectory/URL
   /MenuItem
   MenuItem
 NameReceived Calls/Name
 URLApplication:Cisco/ReceivedCalls/URL
   /MenuItem
   MenuItem
 NamePlaced Calls/Name
 URLApplication:Cisco/PlacedCalls/URL
   /MenuItem
   MenuItem
 NamePersonal Directory/Name
 URLApplication:Cisco/PersonalDirectory/URL
   /MenuItem
 /CiscoIPPhoneMenu

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Re: [OSL | CCIE_Voice] No Services Configured

2012-11-17 Thread Edgar Feliz
Thanks Mohamed.

Edgar

On Sat, Nov 17, 2012 at 11:13 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote:

  Thanks for the clarification. Can't we achieve the same result by using
 these steps

 - Delete the Missed, Placed, Corporate, Placed and Received from the
 Phones  IP Phone Services page
 - From the below link, restore the 5 services with enterprise
 subscription -- f   (false)


 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/7_0_1/cucm-rel_notes-701.html#wp631159

 - Use BAT to subscribe the 5 services to the desired phones

 I tried this and it worked for me.

 Regards,
 Mohamed Gazzaz


 --
 Subject: Re: [OSL | CCIE_Voice] No Services Configured
 From: w...@netcraftsmen.net
 Date: Sat, 17 Nov 2012 11:01:46 -0500
 CC: ke...@kevinspicer.co.uk; pixarperf...@live.com;
 ccie_voice@onlinestudylist.com
 To: mgaz...@hotmail.com

 The OP was referencing one of the four labs in the One Week Lab
 Experience (OWLE) workbook. It is a different set of labs than the 5-Day
 Lab workbook.

 -Bill

 --
 William Bell
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla



 On Nov 17, 2012, at 5:29 AM, Mohamed Gazzaz wrote:

 Where is this solution ? which lab and question ?  I searched for it but
 could not find it in the 5 labs self study workbook.

 --
 Date: Sat, 17 Nov 2012 00:28:52 +
 From: ke...@kevinspicer.co.uk
 To: pixarperf...@live.com
 CC: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] No Services Configured

 The enterprise url parameters apply only to phones using external
 provisioning.  So the IP expert solution is fine.
 Be aware that some older phone models (e.g.7960) only support external
 provisioning so you might see different results depending on what hardware
 you have - but the Cisco lab itself is 7965 phones (this is public
 published info)
 On 17 Nov 2012 00:01, Pixar Perfect pixarperf...@live.com wrote:


 I am working on the question from the lab experience handbook where SA
 phone 1 needs to see No Services Configured when it hits directory
 button. IPEXPERT suggests that we delete the Service URL from EP and then
 setup the SA Phone 1 device configuration page Service parameter to
 External ? I dont think it is acceptable solution as this could cause other
 phones to not see any services if they reboot and download new config file
 which has no directories on any phone !!!

 My approach as follows
 1) make a directory.xml file using the IP Phone Services document from
 cisco.com
 2) delete the Missed, Placed, Corporate, Placed and Received from the
 Phones  IP Phone Services page
 3) Setup the EP for Services to Both  Update the URL to
 http://x.x.x.x/directory.xml where x.x.x.x is UCCX IP
 4) Upload the directory.xml to the IIS server on the UCCX, restart IIS
 5) on SA Phone 1 page, setup the Services to Internal, Reset all phones


 Comments? Feedback? Experiences? TIA.


 directory.xml

 CiscoIPPhoneMenu
   MenuItem
 NameMissed Calls/Name
 URLApplication:Cisco/MissedCalls/URL
   /MenuItem
   MenuItem
 NameCorporateDirectory/Name
 URLApplication:Cisco/CorporateDirectory/URL
   /MenuItem
   MenuItem
 NameReceived Calls/Name
 URLApplication:Cisco/ReceivedCalls/URL
   /MenuItem
   MenuItem
 NamePlaced Calls/Name
 URLApplication:Cisco/PlacedCalls/URL
   /MenuItem
   MenuItem
 NamePersonal Directory/Name
 URLApplication:Cisco/PersonalDirectory/URL
   /MenuItem
 /CiscoIPPhoneMenu

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Re: [OSL | CCIE_Voice] AAR Not Working

2012-11-15 Thread Edgar Feliz
A lot of times for me at least AAR would work after I reboot the routers.

Edgar

On Wed, Nov 14, 2012 at 10:05 PM, Bill whl...@gmail.com wrote:

 So you can see the call go out the pstn at HQ and arrive at SB but you get
 busy?

 What is the error or reason for busy?

 Normally, I do not put anything but aar pt in aar CSS.  I the assign the
 aar CSS to the device level.  I assign the aar group to the line level.  On
 the target I set the aar number, uncheck the box and put aar as its CSS.

 Bill


 On Nov 14, 2012, at 7:08 PM, Barrera, Hugo hugo.barr...@nexusis.com
 wrote:

  Hi,

 ** **

 I am doing AAR, Lab 6 Volume 1, but I just can’t get it to work correctly.
 The following is my programming task list and what I witnessed:

 ** **

 Call flow = HQ Phone 2 x5002 calls BR1 Phone 2 x1002 

 ** **

 1.  Enabled AAR globally in service parameters 

 2.  Created my AAR-PT and AAR-CSS

 o   The AAR-CSS contains the AAR-PT and Internal-PT

 3.  Both my phones have their corresponding Ext Phone Mask (10
 digits) 

 4.  Created my (1) AAR Group named = HQ-BR1-AARGroup

 o   Prefix for this AAR group is “91” 

 5.  Assigned the above AAR group at the line and device level of both
 phones 

 6.  Built out separate AAR Rout Patterns pointing to new Route List **
 **

 ** **

 Troubleshooting:

 I ran my debugs on both gateways and could see the call from CUCM to the
 HQ-RTR I could also see the incoming call from PSTN to BR1-RTR however I
 still received a busy?

 ** **

 Any ideas would be appreciated! 

 ** **

 ** **

 *- Hugo*

 ** **

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Re: [OSL | CCIE_Voice] CME Call List Presence

2012-10-05 Thread Edgar Feliz
Happens to me every time. It will not work unless I reboot and I have the
the create cnf and reset the phones too.

Edgar

On Fri, Oct 5, 2012 at 12:52 PM, Bruno Nonogaki brun...@gmail.com wrote:

 Hello,

 When I configure CME Call List presence, I always have to reboot the
 router to make it work. Does this also happen with you guys? Or is this a
 Proctor Lab's bug?

 Even after an ephone reset, I cannot see the presence indication in call
 lists.
 But right after a reboot, it works perfectly.

 The configuration I apply is as follow:

 presence
  presence call-list

 sip-ua
  presence enable

 ephone-dn X
  allow watch


 Thank you,

 Bruno


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Re: [OSL | CCIE_Voice] new 5 Lab 2 - CUPS not syncing with CUCM database

2012-08-12 Thread Edgar Feliz
Did you reset the trunk? That ussually works for me.

On Sun, Aug 12, 2012 at 2:29 PM, Rrcrumm rrcr...@yahoo.com wrote:

 Yes

 Sent from my iPhone

 On Aug 12, 2012, at 11:21 AM, Jeff S jes20...@gmail.com wrote:

 On CUPS, run the command “show tech network hosts” from the CLI.  Does
 CUPS have the correct hostname for CUCM?

 ** **

 --Jeff

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Randall Crumm
 *Sent:* Sunday, August 12, 2012 10:10 AM
 *To:* Online Study
 *Subject:* [OSL | CCIE_Voice] new 5 Lab 2 - CUPS not syncing with CUCM
 database

 ** **

 Hello,

 Here is the error on the CUPS server from the CUCM Publisher page.

 ** **

 Sync Completed, but currently failed to connect to the CUCM Database
 Monitor. Retrying... 2012-08-12 13:37:17

 ** **

 ** **

 I reset the AXL username and pw. It connects wit the credentials.

 ** **

 It seems like the hostname is not correct.

 ** **

 I changed it to UCMPub. I got this from the CUCM server in system/Cisco
 Unified CM. Also from the CLI, show status, I see the same hostname.

 ** **

 Any thoughts?

 Is this working for other people?

  

 Cheers,
 Randall


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Re: [OSL | CCIE_Voice] new 5 Lab 2 - CUPS not syncing with CUCM database

2012-08-12 Thread Edgar Feliz
Sometimes takes a couple of resets.

Edgar

On Sun, Aug 12, 2012 at 4:48 PM, Randall Crumm rrcr...@yahoo.com wrote:

 Thats what I did and it worked.

 Thought I reset it when I created it.



 Cheers,
 Randall

   --
 *From:* Edgar Feliz ejzi...@gmail.com
 *To:* Rrcrumm rrcr...@yahoo.com
 *Cc:* Jeff S jes20...@gmail.com; Online Study 
 ccie_voice@onlinestudylist.com
 *Sent:* Sunday, August 12, 2012 1:33 PM
 *Subject:* Re: [OSL | CCIE_Voice] new 5 Lab 2 - CUPS not syncing with
 CUCM database

 Did you reset the trunk? That ussually works for me.

 On Sun, Aug 12, 2012 at 2:29 PM, Rrcrumm rrcr...@yahoo.com wrote:

 Yes

 Sent from my iPhone

 On Aug 12, 2012, at 11:21 AM, Jeff S jes20...@gmail.com wrote:

 On CUPS, run the command “show tech network hosts” from the CLI.  Does
 CUPS have the correct hostname for CUCM?
 ** **
 --Jeff
 ** **
 ** **
 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Randall Crumm
 *Sent:* Sunday, August 12, 2012 10:10 AM
 *To:* Online Study
 *Subject:* [OSL | CCIE_Voice] new 5 Lab 2 - CUPS not syncing with CUCM
 database
 ** **
 Hello,
 Here is the error on the CUPS server from the CUCM Publisher page.
 ** **
 Sync Completed, but currently failed to connect to the CUCM Database
 Monitor. Retrying... 2012-08-12 13:37:17
 ** **
 ** **
 I reset the AXL username and pw. It connects wit the credentials.
 ** **
 It seems like the hostname is not correct.
 ** **
 I changed it to UCMPub. I got this from the CUCM server in system/Cisco
 Unified CM. Also from the CLI, show status, I see the same hostname.
 ** **
 Any thoughts?
 Is this working for other people?
  
 Cheers,
 Randall


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Re: [OSL | CCIE_Voice] 150 ms latency

2012-02-26 Thread Edgar Feliz
That's one way. 

Sent from my iPhone

On Feb 26, 2012, at 3:06 PM, donny f f.faraday...@gmail.com wrote:

 hi all,
  
 are the  150 ms   latency and  64 kpbs  for  one way  or 2 way ?
  
 tks
 d
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Re: [OSL | CCIE_Voice] Tragic news

2012-02-25 Thread Edgar Feliz
WoW!!! Sorry to hear that, so young too. Condolences to his family.

Edgar



On Fri, Feb 24, 2012 at 10:59 PM, Emanuel Damasceno
aedamasc...@gmail.comwrote:

 Hello brothers,

 I just like to let you know that my friend and study partner, Jeferson
 Guardia CCIE #28157, has passed away yesterday.

 He was a skateboarder and day before yesterday, he fell, hitting his head.
 He didn't want to go to the hospital, because he thought he was ok.
 According to his father he passed in his sleep due to a blood clot in his
 brain. This is a tragic moment for all his family and friends.

 I thought I should share this with you guys because he's been very active
 here on the list, and we were studying together for the CCIE Voice. He was
 a great motivator and helped me get out of my personal problems so I could
 focus on my studies. It's sad how life is, and what shocks everybody the
 most is that he was only 24 years old (soon to be 25 on March 20th).

 Mourning, but still on the fight... =(

 *Emanuel Damasceno*
 CCNP Voice




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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-11 Thread Edgar Feliz
Thanks I'll be working on this later this week.

Edgar

On Fri, Feb 10, 2012 at 4:45 PM, romain mullier romain.mull...@gmail.comwrote:

 I had tried with and without authentication with the same result.


 Thanks

 On Fri, Feb 10, 2012 at 3:38 PM, Vik Malhi vma...@ipexpert.com wrote:

 Can you try doing a CUC/UCM SIP integration with authentication required?
 I'm not able to test this right now but can do later on.

   Vik Malhi – CCIE #13890
 Managing Partner - IPexpert, Inc.

 Telephone: +1.810.326.1444 ext 420
 Fax: +1.810.454.0130
 Mailto: vma...@ipexpert.com




 On Feb 10, 2012, at 9:31 AM, romain mullier wrote:

 Edgar, William, Vik,

 I recreated this issue with the CUC/SIP integration and narrowed it down
 a bit more. It seems that the issue only occurs when the caller does not
 have a mailbox on CUC (Site C callers, PSTN callers..). If you add a VM box
 for your Site C guy for instance then he can successfully leave the
 voicemail to A and B with MWI and the whole nine yard. If you remove his
 mailbox then you are back to the point where the message never makes to its
 destination. Still haven't figured out the root cause. Anyone?

 Romain

 On Sun, Feb 5, 2012 at 9:42 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 I have the same issue in my own LAB, and as soon as I restart my
 CUC server, the MWI and Message start to work from PSTN for a while. but
 then again stops.

 And I make restart of CUC server every time.

 Thus I was using SCCP integration.

 On Sun, Feb 5, 2012 at 1:00 AM, Edgar Feliz ejzi...@gmail.com wrote:

 Also another issue I had was that it seemed like when I was leaving a
 VM and press # it was not recognizing that from any phone other then SA.

 Had most of the lab working except for the SIP/CUC.

 Thanks,

 Edgar

  On Sat, Feb 4, 2012 at 3:15 PM, Vik Malhi vma...@ipexpert.com wrote:

  Can you successfully leave SAP2 a new VM from any phone ? Another SA
 phone or PSTN or SB?



 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: www.ipexpert.com/chat
 http://www.ipexpert.com/chat

 IPexpert is a premier provider of Self-Study Workbooks, Video on
 Demand, Audio Tools, Online Hardware Rental and Classroom Training
 for the Cisco CCIE (RS, Voice, Wireless, Security  Service
 Provider) certification(s) with training locations throughout the
 United States, Europe, South Asia and Australia. Be sure to visit our
 online communities at www.ipexpert.com/communities 
 http://www.ipexpert.com/communities  and our public website at
 www.ipexpert.com http://www.ipexpert.com/


 On Feb 4, 2012, at 12:08 PM, William Affeldt 
 william.affe...@yahoo.com wrote:

   So I am good then?

 Sent from my iPhone

 On Feb 4, 2012, at 11:59 AM, Vik Malhi vma...@ipexpert.com wrote:

   Edgar- make sure that you do not have one way cRTP. Or there is any
 MTP being used no sccp/sccp.

 Bill-  there is a CUC bug when you leave a VM  and can press # and
 hear your message, but this message never gets sent to the mail box (from
 specific ip addresses). In this case you have to just rely on VM/MWI from
 an extension /gateway that does work.



 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.comvma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: http://www.ipexpert.com/chat
 www.ipexpert.com/chat
  http://www.ipexpert.com/chathttp://www.ipexpert.com/chat

 IPexpert is a premier provider of Self-Study Workbooks, Video on
 Demand, Audio Tools, Online Hardware Rental and Classroom Training
 for the Cisco CCIE (RS, Voice, Wireless, Security  Service
 Provider) certification(s) with training locations throughout the
 United States, Europe, South Asia and Australia. Be sure to visit our
 online communities at  http://www.ipexpert.com/communities
 www.ipexpert.com/communities  http://www.ipexpert.com/communities
 http://www.ipexpert.com/communities  and our public website at
 http://www.ipexpert.com/www.ipexpert.com http://www.ipexpert.com/
 http://www.ipexpert.com/


 On Feb 4, 2012, at 11:42 AM, William Affeldt william.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

   You are having one way audio issues then.

 Sent from my iPhone

 On Feb 4, 2012, at 11:27 AM, Edgar Feliz  ejzi...@gmail.com
 ejzi...@gmail.com wrote:

  I don't hear the message when I press # for more options from SC
 or PSTN nothing is happening but I am getting the options from SA/SB

 E

 On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt 
 william.affe...@yahoo.comwilliam.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

  I am currently having the same problem. I have been troubleshooting
 for a hour now. It forwards to the correct VM box and you can even
 play the message back to your self after you record it. It just never 
 makes
 it to the mailbox.

   *From:* Edgar Feliz  ejzi

Re: [OSL | CCIE_Voice] UCCX scripts

2012-02-05 Thread Edgar Feliz
OOPs!! Sorry Randall I gave you option to copy Vic's scripts to the
RDP/CCX. Not sure if it works the other way around, but there aren't any
script kept on the server because they rebuilt every session.

Edgar



On Sun, Feb 5, 2012 at 11:29 AM, Ken Wyan kew...@gmail.com wrote:

 That's illegal.

 share one directory of your computer  copy from UCCX server to shared
 directory.

 On Sun, Feb 5, 2012 at 9:21 PM, Randall Crumm rrcr...@yahoo.com wrote:

  HI,
 How can I copy the icd script to my computer from the proctor labs pod?

 Thanks,

 Cheers, Randall

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[OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread Edgar Feliz
I am currently working on new lab 5 and I have my CUCM-CUC integration
working, for voicemail left by SA  SB to SB phone and SB  SA Phone I get
MWI both directions. But for PSTN or SC While I can leave a VM MWI
does not work and the VM does not show up when I check the inbox for either
SA or SB phones for VM left from PSTN or SC. I have looked at the SIP trunk
setting and do not see anything there CSS/PTs all look correct any Ideas?

Thanks

E
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread Edgar Feliz
I don't hear the message when I press # for more options from SC or PSTN
nothing is happening but I am getting the options from SA/SB

E

On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt
william.affe...@yahoo.comwrote:

 I am currently having the same problem. I have been troubleshooting for a
 hour now. It forwards to the correct VM box and you can even play the
 message back to your self after you record it. It just never makes it to
 the mailbox.

   *From:* Edgar Feliz ejzi...@gmail.com
 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Saturday, February 4, 2012 10:15 AM
 *Subject:* [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

 I am currently working on new lab 5 and I have my CUCM-CUC integration
 working, for voicemail left by SA  SB to SB phone and SB  SA Phone I
 get MWI both directions. But for PSTN or SC While I can leave a VM MWIdoes 
 not work and the
 VM does not show up when I check the inbox for either SA or SB phones for
 VM left from PSTN or SC. I have looked at the SIP trunk setting and do
 not see anything there CSS/PTs all look correct any Ideas?

 Thanks

 E

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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com


___
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread Edgar Feliz
Yes I can leave VM from SB phones only SC and PSTN I get to vm prompt but
VM never appears on phones.

Edgar

On Sat, Feb 4, 2012 at 3:15 PM, Vik Malhi vma...@ipexpert.com wrote:

 Can you successfully leave SAP2 a new VM from any phone ? Another SA phone
 or PSTN or SB?



 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: www.ipexpert.com/chat
 http://www.ipexpert.com/chat

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio
 Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
 (RS, Voice, Wireless, Security  Service Provider) certification(s) with 
 training
 locations throughout the United States, Europe, South Asia and Australia.
 Be sure to visit our online communities at www.ipexpert.com/communities 
 http://www.ipexpert.com/communities  and our public website at
 www.ipexpert.com http://www.ipexpert.com/


 On Feb 4, 2012, at 12:08 PM, William Affeldt william.affe...@yahoo.com
 wrote:

 So I am good then?

 Sent from my iPhone

 On Feb 4, 2012, at 11:59 AM, Vik Malhi vma...@ipexpert.com wrote:

 Edgar- make sure that you do not have one way cRTP. Or there is any MTP
 being used no sccp/sccp.

 Bill-  there is a CUC bug when you leave a VM  and can press # and hear
 your message, but this message never gets sent to the mail box (from
 specific ip addresses). In this case you have to just rely on VM/MWI from
 an extension /gateway that does work.



 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.comvma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: http://www.ipexpert.com/chat
 www.ipexpert.com/chat
  http://www.ipexpert.com/chathttp://www.ipexpert.com/chat

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio
 Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
 (RS, Voice, Wireless, Security  Service Provider) certification(s) with 
 training
 locations throughout the United States, Europe, South Asia and Australia.
 Be sure to visit our online communities at 
 http://www.ipexpert.com/communities
 www.ipexpert.com/communities  http://www.ipexpert.com/communities
 http://www.ipexpert.com/communities  and our public website at
 http://www.ipexpert.comwww.ipexpert.com  http://www.ipexpert.com/
 http://www.ipexpert.com/


 On Feb 4, 2012, at 11:42 AM, William Affeldt  william.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

 You are having one way audio issues then.

 Sent from my iPhone

 On Feb 4, 2012, at 11:27 AM, Edgar Feliz  ejzi...@gmail.com
 ejzi...@gmail.com wrote:

 I don't hear the message when I press # for more options from SC or PSTN
 nothing is happening but I am getting the options from SA/SB

 E

 On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt 
 william.affe...@yahoo.comwilliam.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

 I am currently having the same problem. I have been troubleshooting for a
 hour now. It forwards to the correct VM box and you can even play the
 message back to your self after you record it. It just never makes it to
 the mailbox.

   *From:* Edgar Feliz  ejzi...@gmail.com ejzi...@gmail.com
 ejzi...@gmail.com
 *To:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 ccie_voice@onlinestudylist.com
 *Sent:* Saturday, February 4, 2012 10:15 AM
 *Subject:* [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

 I am currently working on new lab 5 and I have my CUCM-CUC integration
 working, for voicemail left by SA  SB to SB phone and SB  SA Phone I
 get MWI both directions. But for PSTN or SC While I can leave a VM MWIdoes 
 not work and the
 VM does not show up when I check the inbox for either SA or SB phones
 for VM left from PSTN or SC. I have looked at the SIP trunk setting and
 do not see anything there CSS/PTs all look correct any Ideas?

 Thanks

 E

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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.PlatinumPlacement.com http://www.PlatinumPlacement.com
 www.PlatinumPlacement.com


  ___
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 visit http://www.ipexpert.comwww.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread Edgar Feliz
Also another issue I had was that it seemed like when I was leaving a VM
and press # it was not recognizing that from any phone other then SA.

Had most of the lab working except for the SIP/CUC.

Thanks,

Edgar

On Sat, Feb 4, 2012 at 3:15 PM, Vik Malhi vma...@ipexpert.com wrote:

 Can you successfully leave SAP2 a new VM from any phone ? Another SA phone
 or PSTN or SB?



 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: www.ipexpert.com/chat
 http://www.ipexpert.com/chat

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio
 Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
 (RS, Voice, Wireless, Security  Service Provider) certification(s) with 
 training
 locations throughout the United States, Europe, South Asia and Australia.
 Be sure to visit our online communities at www.ipexpert.com/communities 
 http://www.ipexpert.com/communities  and our public website at
 www.ipexpert.com http://www.ipexpert.com/


 On Feb 4, 2012, at 12:08 PM, William Affeldt william.affe...@yahoo.com
 wrote:

 So I am good then?

 Sent from my iPhone

 On Feb 4, 2012, at 11:59 AM, Vik Malhi vma...@ipexpert.com wrote:

 Edgar- make sure that you do not have one way cRTP. Or there is any MTP
 being used no sccp/sccp.

 Bill-  there is a CUC bug when you leave a VM  and can press # and hear
 your message, but this message never gets sent to the mail box (from
 specific ip addresses). In this case you have to just rely on VM/MWI from
 an extension /gateway that does work.



 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.comvma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: http://www.ipexpert.com/chat
 www.ipexpert.com/chat
  http://www.ipexpert.com/chathttp://www.ipexpert.com/chat

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio
 Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
 (RS, Voice, Wireless, Security  Service Provider) certification(s) with 
 training
 locations throughout the United States, Europe, South Asia and Australia.
 Be sure to visit our online communities at 
 http://www.ipexpert.com/communities
 www.ipexpert.com/communities  http://www.ipexpert.com/communities
 http://www.ipexpert.com/communities  and our public website at
 http://www.ipexpert.comwww.ipexpert.com  http://www.ipexpert.com/
 http://www.ipexpert.com/


 On Feb 4, 2012, at 11:42 AM, William Affeldt  william.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

 You are having one way audio issues then.

 Sent from my iPhone

 On Feb 4, 2012, at 11:27 AM, Edgar Feliz  ejzi...@gmail.com
 ejzi...@gmail.com wrote:

 I don't hear the message when I press # for more options from SC or PSTN
 nothing is happening but I am getting the options from SA/SB

 E

 On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt 
 william.affe...@yahoo.comwilliam.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

 I am currently having the same problem. I have been troubleshooting for a
 hour now. It forwards to the correct VM box and you can even play the
 message back to your self after you record it. It just never makes it to
 the mailbox.

   *From:* Edgar Feliz  ejzi...@gmail.com ejzi...@gmail.com
 ejzi...@gmail.com
 *To:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 ccie_voice@onlinestudylist.com
 *Sent:* Saturday, February 4, 2012 10:15 AM
 *Subject:* [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

 I am currently working on new lab 5 and I have my CUCM-CUC integration
 working, for voicemail left by SA  SB to SB phone and SB  SA Phone I
 get MWI both directions. But for PSTN or SC While I can leave a VM MWIdoes 
 not work and the
 VM does not show up when I check the inbox for either SA or SB phones
 for VM left from PSTN or SC. I have looked at the SIP trunk setting and
 do not see anything there CSS/PTs all look correct any Ideas?

 Thanks

 E

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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.PlatinumPlacement.com http://www.PlatinumPlacement.com
 www.PlatinumPlacement.com


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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.PlatinumPlacement.comwww.PlatinumPlacement.com


___
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Re: [OSL | CCIE_Voice] 5-lab Workbook #4 - Unity Connection Notification

2012-01-24 Thread Edgar Feliz
Vic, have the updated guides been uploaded? We only get so many print
cycles and just want to make sure I am not wasting paper too.

Thanks

On Tue, Jan 24, 2012 at 2:38 AM, Vik Malhi vma...@ipexpert.com wrote:

 I don't think they are in beta mode. I accidentally skipped a question,
 it's been corrected and it should have been uploaded. I'll ensure it is
 tomorrow.

 Vik Malhi – CCIE #13890
 Managing Partner - IPexpert, Inc.

 Telephone: +1.810.326.1444 ext 420
 Fax: +1.810.454.0130
 Mailto: vma...@ipexpert.com




 On Jan 22, 2012, at 7:14 PM, Jurassic Labs wrote:

 !  ALL BE ADVISED   
 The new 5-lab handbooks are somewhat in beta mode I believe as I've ran
 into some minor errors and such.  But the biggest issue is that I'm finding
 the solution guide for Lab#5 is not available (the estimated timeline for
 it to published has passed) and I'm finding out in Lab#4 there a section
 completly skipped in the solution guide.  If anyone has some thoughts about
 making this work - please post.

 Ensure that everytime somebody leaves user [fill in name] a new voicemail
 in his account that a message notification call is made out to PSTN line
 2.  The call should be sent out of the SiteA gateway as a local call, with
 the SiteB gateway acting as a backup.  If the call goes out the SiteB
 gateway, it should be a long distance call.  Ensure that Unity Connection
 cannot place a call outbound to any other number except PSTN line 2
 (+12024678124).

 Setting up a subscriber's notification device is the easy part - getting
 Unity Connection to dial out that number to CUCM - where / how is that
 controlled?
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] 5-lab Workbook #4 - Unity Connection Notification

2012-01-24 Thread Edgar Feliz
OK great thanks.

On Tue, Jan 24, 2012 at 2:25 PM, Vik Malhi vma...@ipexpert.com wrote:

 They have been uploaded.


 Vik Malhi – CCIE #13890
 Managing Partner - IPexpert, Inc.

 Telephone: +1.810.326.1444 ext 420
 Fax: +1.810.454.0130
 Mailto: vma...@ipexpert.com




 On Jan 24, 2012, at 11:08 AM, Edgar Feliz wrote:

 Vic, have the updated guides been uploaded? We only get so many print
 cycles and just want to make sure I am not wasting paper too.

 Thanks

 On Tue, Jan 24, 2012 at 2:38 AM, Vik Malhi vma...@ipexpert.com wrote:

 I don't think they are in beta mode. I accidentally skipped a question,
 it's been corrected and it should have been uploaded. I'll ensure it is
 tomorrow.

   Vik Malhi – CCIE #13890
 Managing Partner - IPexpert, Inc.

 Telephone: +1.810.326.1444 ext 420
 Fax: +1.810.454.0130
 Mailto: vma...@ipexpert.com




 On Jan 22, 2012, at 7:14 PM, Jurassic Labs wrote:

 !  ALL BE ADVISED   
 The new 5-lab handbooks are somewhat in beta mode I believe as I've ran
 into some minor errors and such.  But the biggest issue is that I'm finding
 the solution guide for Lab#5 is not available (the estimated timeline for
 it to published has passed) and I'm finding out in Lab#4 there a section
 completly skipped in the solution guide.  If anyone has some thoughts about
 making this work - please post.

 Ensure that everytime somebody leaves user [fill in name] a new voicemail
 in his account that a message notification call is made out to PSTN line
 2.  The call should be sent out of the SiteA gateway as a local call, with
 the SiteB gateway acting as a backup.  If the call goes out the SiteB
 gateway, it should be a long distance call.  Ensure that Unity Connection
 cannot place a call outbound to any other number except PSTN line 2
 (+12024678124).

 Setting up a subscriber's notification device is the easy part - getting
 Unity Connection to dial out that number to CUCM - where / how is that
 controlled?
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com




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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] 5-lab SelfStudy NTP

2012-01-23 Thread Edgar Feliz
Your prefer method is the way i have done the labs.

E

On Mon, Jan 23, 2012 at 3:58 PM, Jurassic Labs jurassicl...@gmail.comwrote:

 In both LAB#4 and LAB#5 of the newer 5-lab series, there's an NTP section
 that I don't think is being fully addressed in the DSG (Solution Guides).
  Here's the question:

 Configure the UCM, Site A / B / C gateway to synchronize their clock with
 the clock source on the backbone NTP  server.  The IP Address of the NTP
 server is 10.10.100.2.  In all cases the backup NTP server is SiteA gateway
 loopback IP Address (10.10.110.1).

 The DSG just covers pointing the PUB, SiteA-RTR, SiteB-RTR, and SiteC-RTR
 to 10.10.100.2 via command ntp server 10.10.100.2.  But isn't that just
 half of the story??  Shouldn't the PUB have both 10.10.100.2 AND
 10.10.110.1 listed?  Then SiteA would have ntp master 6, then for both
 SiteB  SiteC we could enter the two commands, ntp server 10.10.100.2
 prefer   ntp server 10.10.110.1.

 Wouldn't that be the complete solution?

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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

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For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Five New Lab from IPExpert

2012-01-20 Thread Edgar Feliz
Hi Wayne,

The guide for lab 5 is still not up.

Edgar

On Sun, Jan 15, 2012 at 1:02 PM, Wayne Lawson
waynelawson-...@ipexpert.comwrote:

 Gang -

  I think the 5th DSG is going to be uploaded this week. Matt / Vik -
 can you verify?

 Regards,

 Wayne A. Lawson II - CCIE #5244
 Founder  President
 IPexpert, Inc., Proctor Labs, Inc., Masonic e-Institute of Technology,
 Inc.,  Platinum, Inc.

 Mobile: +1.810.334.1564
 eFax: +1.810.454.0130
 Email: wlaw...@ipexpert.com
 Connect @ www.WayneLawson.com

 :: Message sent from iPhone.

 On Jan 15, 2012, at 12:38 PM, Leslie Meade lme...@salientnetworks.com
 wrote:

 you got the last proctor guide for 5?
  I still only see up to 4



 --
 *From: *Rrcrumm rrcr...@yahoo.com
 *To: *John McGaughey (jomcgaug) jomcg...@cisco.com
 *Cc: *ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent: *Sunday, January 15, 2012 8:03:02 AM
 *Subject: *Re: [OSL | CCIE_Voice] Five New Lab from IPExpert

 Hi
 I have the new five labs and I think they are much better. They have alot
 of new things and really good explanations.

 I tho k they are worth the money

 HTH
 Randall

 Sent from my iPhone

 On Jan 15, 2012, at 6:53 AM, John McGaughey (jomcgaug) 
 jomcg...@cisco.com wrote:

 Has anyone purchased the five new labs from IPExpert?  If so, are they
 worth money?  What’s different about them versus the 10 lab workbook?
 Anything new or just more of the same?



 John



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com


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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com


 ___
 For more information regarding industry leading CCIE Lab training, please
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Re: [OSL | CCIE_Voice] CUCM Log Problem

2011-12-05 Thread Edgar Feliz
Are you looking at the Sub CUCM or whatever  you primary call processing
agent is. IF you are in the wrong CUCM you will not see any call
information.

E

On Mon, Dec 5, 2011 at 10:56 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 Hello,

 I have a problem with CUCM  - when trying to view Logs in real time either
 through RTMT or file tail command in CLI, the logs starts only for
 several lines, and then stops logging, there are no new information (Loggin
 messages) coming up any more.


 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com

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Re: [OSL | CCIE_Voice] Calling Name with H323 GW

2011-11-29 Thread Edgar Feliz
Does this help?

you have a customer who does not want to see the final transmitted number.
Rather he wants to see the number that was sent to the GW (e.g. in a e164
concept the one starting with a +).

this can be done by disabling the h225 CID update during the call.
This can be set either globally:
voice service voip
no supplementary-service h225-notify cid-update

or  on a individual dialpeer:
dial-peer voice 10
no supplementary-service h225-notify cid-update

On Tue, Nov 29, 2011 at 8:54 AM, ccielabrat ccielab...@gmail.com wrote:

 The problem was the check box on the gateway configuration.
 After looking at every available option on the gateway, I found out CUCM
 wasn't send the name in the first place :)

 I do have a follow up question though.

 I'm really confused about the logic around no supplementary-service
 h225-notify cid-update
 If the command is by default supplementary-service h225-notify
 cid-update, it seems to me this would ENABLE
 the CID to be updated.
 It doesn't make sense to me why DISABLING this with NO
 supplementary-service h225-notify cid-update actually allows the gateway
 to trigger an updated display on the phone.



 On Tue, Nov 29, 2011 at 5:06 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 Also you have to check the Display IE checkbox in CUCM H323 gateway
 configuration.


 On Tue, Nov 29, 2011 at 7:05 AM, Rrcrumm rrcr...@yahoo.com wrote:

 Do you have isdn outgoing display-ie under serial 0/0/0:23 or 15
 interface?

 Randall



 Sent from my iPhone

 On Nov 28, 2011, at 5:52 PM, ccielabrat ccielab...@gmail.com wrote:

  I must be missing something easy.
  I'm trying to get Calling Name to display on my PSTN phone when
 receiving a call from a IP phone going through a H323 gateway.
 
  I've found many links online suggesting it's not supported but then
 others suggesting it's possible.
 
  Can someone point me to a good link?
 
 
   ___
  For more information regarding industry leading CCIE Lab training,
 please visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/
 ___
 For more information regarding industry leading CCIE Lab training,
 please visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/




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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

___
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[OSL | CCIE_Voice] lab @ RTP

2011-11-29 Thread Edgar Feliz
Anyone have a lab scheduled at RTP in the next few days?

E
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Re: [OSL | CCIE_Voice] Calling Name with H323 GW

2011-11-28 Thread Edgar Feliz
Did you put in the

voice service voip
   no supplementary-service h225-notify cid-update


On Mon, Nov 28, 2011 at 8:52 PM, ccielabrat ccielab...@gmail.com wrote:

 I must be missing something easy.
 I'm trying to get Calling Name to display on my PSTN phone when
 receiving a call from a IP phone going through a H323 gateway.

 I've found many links online suggesting it's not supported but then others
 suggesting it's possible.

 Can someone point me to a good link?



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

___
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Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration

2011-11-27 Thread Edgar Feliz
Run this command on the CUCM Pub you do NOT have to reboot.

run sql update processconfig set paramvalue=T where paramname like
'%nstalled%'

E

On Sun, Nov 27, 2011 at 2:07 AM, CCIEVoiceKP ccievoic...@gmail.com wrote:

 Have you re-syched the data yet?  Subsystems -- Unified CM Telephony --
 Data Resync

 KP

 On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.com wrote:

  HI,
 I am working on the New Lab 1 UCCX portion. I am suppose to add an UCCX
 extension to each user page, but the field to add the UCCX exntension is
 not there. I also did not see a rmjtapi user in CUCM. This is the user in
 the intergration page on UCCX.

 Any thoughts?

 Randall


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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

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Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration

2011-11-27 Thread Edgar Feliz
That is not the issue there is something wrong with the way the initial
installations took place and you have to run this command. I have been
working on the first 3 labs from the new 5 7 days straight and it is
happening every time.

The T flag is not being set and it has nothing to do with resync  etc. I
spent 3-4 hours on this and the only thing the works is running this on the
CUCM Pub.

run sql update processconfig set paramvalue=T where paramname like
'%nstalled%'

It sets the flag to true

E


On Sun, Nov 27, 2011 at 7:50 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 First you need to integrate UCCX with CUCM fully, meaning that the JTAPI
 and RMCM users must be present in CUCM after successful integration.

 Please verify that you have configured the JTAPI and RMCM in UCCX, so that
 it will be replicated in CUCM automatically.

 Then make JTAPI resynch.

 When all that is done successfully, then you will be able to see the UCCX
 Extension in a Users page.



 On Sun, Nov 27, 2011 at 11:07 AM, CCIEVoiceKP ccievoic...@gmail.comwrote:

 Have you re-syched the data yet?  Subsystems -- Unified CM Telephony --
 Data Resync

 KP

   On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.comwrote:

HI,
 I am working on the New Lab 1 UCCX portion. I am suppose to add an UCCX
 extension to each user page, but the field to add the UCCX exntension is
 not there. I also did not see a rmjtapi user in CUCM. This is the user in
 the intergration page on UCCX.

 Any thoughts?

 Randall


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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

___
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Are you a CCNP or CCIE and looking for a job? Check out 
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Re: [OSL | CCIE_Voice] ISDN Cause i = 0x80AF - Resource unavailable

2011-11-27 Thread Edgar Feliz
Bill what lab are you working on?

On Sun, Nov 27, 2011 at 7:55 AM, Bill Lake whl...@gmail.com wrote:

 OK, if you are sure it is not a ISDN issue but it appears to be that the
 call is set up

 Nov 26 22:10:48.504: ISDN Se0/0/0:15 Q931: RX - ALERTING pd = 8  callref
 = 0x808D
 Progress Ind i = 0x8188 - In-band info or appropriate now availab

 you then get


 Nov 26 22:10:48.548: ISDN Se0/0/0:15 Q931: TX - DISCONNECT pd = 8
 callref = 0x008D
 Cause i = 0x80AF - Resource unavailable, unspecified

 so maybe it is an IOS bug but we don't know what IOS you are running or do
 we know running config or the status of any of your system.

 Just guessing here but can you complete any calls over the PSTN
 connection?  If so what type of calls are completed?


 On Sun, Nov 27, 2011 at 6:36 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 Bill, if you will take a dipper look at the Logs of ISDN, you will see
 the the Router send the ISD SETUP message, then received the Procceding and
 Alerting message.
 So the Voice port in operating normally.
 After the Alert message, the IP Side is starting - H245 is going for
 negotiation, and there is a problem in H323 IP Side, not in ISDN side.

 And I have already mentioned the Problem description in my earlier post.

 If there would the DSP issues, then the SETUP message will not be send
 even.

 On Sun, Nov 27, 2011 at 1:26 AM, Bill Lake whl...@gmail.com wrote:

 This looks like it could be an ISDN issue:

 Is your voice-port in an active (not shut down) state?

 Do you have the proper DSP resources for your configuration?

 Check these and post your configuration if this is not the issue.



 On Sat, Nov 26, 2011 at 1:24 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 That's because of Codec Problem between HQ router and the CUCM.

 Now look, as we know, when the CUCM sends the H323 SETUP message to GW,
 the GW, by default, begins the H245 Address announcement on Alert Message,
 and at that point the CUCM and GW have exchanged the H245 message, and the
 GW provides the Ring Back tone. Even when the CUCM is using the Slow Start.
 Whereas CUCME can detect CUCM and start the H245 only on Connect
 Message, while the default one idicates h245 address on Alert. CUCME IP
 Phones cannot provide the In Band Ring Back tones, so that CUCM needs
 always the Slow start to CUCME, so that CUCM will provide the Local
 Ringback to its IP Phones until the H245 address is received from CUCME
 (whereas CUCME will send the H245 address On Connect only to CUCM).

 Now look, as for you case, you get the error message after the Alerting
 Message. What does happen at that moment? GW sends the Alert message with
 H245 Address (and the TCS OLC negotiation begins). CUCM and GW have some
 problems with Codec negotation, and that's why you get that message.

 Try to ensure that you use the correct region to GW, and also the
 correct Codecs (or voice class codec) is configured on the GW.

 That's the problem of the Codec mismatch.






  On Sat, Nov 26, 2011 at 9:19 PM, Ricardo Palaver 
 ricardo.pala...@hotmail.com wrote:

   Hi Experts!,

 I 've already gave up to fix the problem below.

 I am calling from a HQ - phone #  to PSTN (BR2) region  but call
 fails  and it's returnning me the cause code: 0x80AF.

 In the Cisco community I found its an IOS bug,  so if it is,   someone
 here  have already faced the same issue.

 Can someone give me any idea on how to fix it ?


 Thanks a zillions !..   have a great weekend all !



 debug isdn q931 is  ON.
 BR2-RTR#
 Nov 26 22:10:48.460: ISDN Se0/0/0:15 Q931: Applying typeplan for
 sw-type 0x12 is 0x0 0x0, Calling num 2123945001
 Nov 26 22:10:48.464: ISDN Se0/0/0:15 Q931: Sending SETUP  callref =
 0x008D callID = 0x800E switch = primary-net5 interface = User
 Nov 26 22:10:48.464: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8  callref
 = 0x008D
 Bearer Capability i = 0x8090A3
 Standard = CCITT
 Transfer Capability = Speech
 Transfer Mode = Circuit
 Transfer Rate = 64 kbit/s
 Channel ID i = 0xA98381
 Exclusive, Channel 1
 Display i = 'HQ Phone 1'
 Calling Party Number i = 0x0081, '2123945001'
 Plan:Unknown, Type:Unknown
 Called Party Number i = 0x80, '32141891'
 Plan:Unknown, Type:Unknown
 Nov 26 22:10:48.500: ISDN Se0/0/0:15 Q931: RX - CALL_PROC pd = 8
 callref = 0x808D
 Channel ID i = 0xA98381
 Exclusive, Channel 1
 Nov 26 22:10:48.504: ISDN Se0/0/0:15 Q931: RX - ALERTING pd = 8
 callref = 0x808D
 Progress Ind i = 0x8188 - In-band info or appropriate now
 availab
 BR2-RTR#le
 Nov 26 22:10:48.548: ISDN Se0/0/0:15 Q931: TX - DISCONNECT pd = 8
 callref = 0x008D
 Cause i = 0x80AF - Resource unavailable, unspecified
 Nov 26 22:10:48.560: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8
 callref = 0x808D
 Nov 26 22:10:48.564: ISDN Se0/0/0:15 

Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration

2011-11-27 Thread Edgar Feliz
Larry after manually adding the rmjtapi user you have to restart the engine
if not it would not work.
This is the way it has been for the new labs.
Every single one you have to add the CUCM rmjtapi app-user  associate
phones add cti enable.
on the CCX make sure the pwd is the same I usually put in just in case
resync restart engine.

E

On Sun, Nov 27, 2011 at 10:49 AM, Larry Stern larry.st...@blackbox.comwrote:

 Randall
 I had the same exact problem with New Lab 1, after I ran the SQL command
 from the cucm cli, This fixed the not showing of the  IPCC extension field
 on my cucm enduser page, but after an hour or so of reverting, resyncing
 uccx I never got rmjtapi to show in CUCM as an application user. I tried
 adding it manually into CUCM , and then resyncing UCCX, still ng. Finally
 moved on.

 Sent from my iPad

 On Nov 27, 2011, at 10:13 AM, ccie_voice-requ...@onlinestudylist.com 
 ccie_voice-requ...@onlinestudylist.com wrote:

  Send CCIE_Voice mailing list submissions to
 ccie_voice@onlinestudylist.com
 
  To subscribe or unsubscribe via the World Wide Web, visit
 http://onlinestudylist.com/mailman/listinfo/ccie_voice
  or, via email, send a message with subject or body 'help' to
 ccie_voice-requ...@onlinestudylist.com
 
  You can reach the person managing the list at
 ccie_voice-ow...@onlinestudylist.com
 
  When replying, please edit your Subject line so it is more specific
  than Re: Contents of CCIE_Voice digest...
 
 
  Today's Topics:
 
1. Re: CUCM/UCCX Intergration (Pradeep Kumar Sharma)
2. Re: CUCM/UCCX Intergration (Edgar Feliz)
3. Re: CUCM/UCCX Intergration (Edgar Feliz)
4. Re: ISDN Cause i = 0x80AF - Resource unavailable (Edgar Feliz)
 
 
  --
 
  Message: 1
  Date: Sun, 27 Nov 2011 19:02:29 +0530
  From: Pradeep Kumar Sharma sonu.netwo...@gmail.com
  To: CCIEVoiceKP ccievoic...@gmail.com
  Cc: Online Study ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration
  Message-ID:
 CALWZLGczqjzTX0jq-JEOGOLG=jW+=
 v9epdffuxzk5wj5avs...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  It seems your UCCX server is not integrated properly with CUCM.
 
  Use http://ccie-musketeers.blogspot.com/search/label/UCCX and fix it.
 
  -SONU-
 
  On Sun, Nov 27, 2011 at 12:37 PM, CCIEVoiceKP ccievoic...@gmail.com
 wrote:
 
  Have you re-syched the data yet?  Subsystems -- Unified CM Telephony
 --
  Data Resync
 
  KP
 
  On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.com
 wrote:
 
  HI,
  I am working on the New Lab 1 UCCX portion. I am suppose to add an UCCX
  extension to each user page, but the field to add the UCCX exntension
 is
  not there. I also did not see a rmjtapi user in CUCM. This is the
 user in
  the intergration page on UCCX.
 
  Any thoughts?
 
  Randall
 
 
  ___
  For more information regarding industry leading CCIE Lab training,
 please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com http://www.platinumplacement.com/
 
 
 
  ___
  For more information regarding industry leading CCIE Lab training,
 please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com
 
  -- next part --
  An HTML attachment was scrubbed...
  URL:
 /archives/ccie_voice/attachments/2027/4a72770d/attachment-0001.html
 
  --
 
  Message: 2
  Date: Sun, 27 Nov 2011 08:36:53 -0500
  From: Edgar Feliz ejzi...@gmail.com
  To: CCIEVoiceKP ccievoic...@gmail.com
  Cc: Online Study ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration
  Message-ID:
 
 cabxks5qfl_jbyttuxstdw9+vxw0z-w0_8-fpebwtxackx_u...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  Run this command on the CUCM Pub you do NOT have to reboot.
 
  run sql update processconfig set paramvalue=T where paramname like
  '%nstalled%'
 
  E
 
  On Sun, Nov 27, 2011 at 2:07 AM, CCIEVoiceKP ccievoic...@gmail.com
 wrote:
 
  Have you re-syched the data yet?  Subsystems -- Unified CM Telephony
 --
  Data Resync
 
  KP
 
  On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.com
 wrote:
 
  HI,
  I am working on the New Lab 1 UCCX portion. I am suppose to add an UCCX
  extension to each user page, but the field to add the UCCX exntension
 is
  not there. I also did not see a rmjtapi user in CUCM. This is the
 user in
  the intergration page on UCCX.
 
  Any thoughts?
 
  Randall
 
 
  ___
  For more information regarding industry leading CCIE Lab training,
 please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com http://www.platinumplacement.com

[OSL | CCIE_Voice] 5 New labs what to watch for!!!

2011-11-27 Thread Edgar Feliz
Guys I have been working on the new (5) labs mostly 1-3 since they have the
guide. Here is what i have found to be things to watch for and fixes,

Please note i have been working on these for over 10 days and it's the
same every time it is just the way the integration was done for the VM
images purposely or not.



   1. There are some typos you will see reference to 011916745738931 but if
   you try to call that number it is not available true number
   is 011916745738931
   2. There are some others name of routers in guide not always right when
   you look for solutions so just make sure you note what router you are
   working on. I think this is mainly because of copy and pasting from one lab
   to another etc since some things are similar but devices are different.
   3. UCCX integration;


   1. rmjtapi app-user is not configured  add rmjtapi  add phones and cti
  enable
  2. make sure pwds match on CCX and CUCM for jtapi_1 and rmjtapi
  3. from CCX run sync and restart engine
  4. You may have to re-sync a couple of times


   1. the T flag is not set so the CUCM never sees that the CCX is
   integrated you have to do a manual step to get this up and running so you
   can add the UCCX extension to users,


   1. run this command on the CUCM Pub as stated in earleir emails  run
  sql update processconfig set paramvalue=T where paramname like
  '%nstalled%'
  2. NO need to reload it takes affect right away


   1. CUC - I have had to reload the CUC server every time for MWIs and to
   be able to leave message... This one I am not sure why I have no fix other
   than to restart the server.
   2. CUE - As noted in labs WB and Guide you will need to change licenses
   depending on the lab you are working on.
   3. Lab 2 CCXquestion there is a step missing in the script. a generated
   prompt of zero calls ahead of you when you are the only one  in queue = you
   do need the decrement step (thanks larry)

If i recall or run into any more items of interest i will post back i am in
last 5 days of prep and will be concentrating on the labs

E
___
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Re: [OSL | CCIE_Voice] 5 New labs what to watch for!!!

2011-11-27 Thread Edgar Feliz
Sorry had a copy/paste issue myself 2nd number in first point should be
01191674573893 0119167457389312

On Sun, Nov 27, 2011 at 12:12 PM, Edgar Feliz ejzi...@gmail.com wrote:

 Guys I have been working on the new (5) labs mostly 1-3 since they have
 the guide. Here is what i have found to be things to watch for and fixes,

 Please note i have been working on these for over 10 days and it's the
 same every time it is just the way the integration was done for the VM
 images purposely or not.



1. There are some typos you will see reference to 011916745738931 but
if you try to call that number it is not available true number is
011916745738931
2. There are some others name of routers in guide not always right
when you look for solutions so just make sure you note what router you are
working on. I think this is mainly because of copy and pasting from one lab
to another etc since some things are similar but devices are different.
3. UCCX integration;


1. rmjtapi app-user is not configured  add rmjtapi  add phones and
   cti enable
   2. make sure pwds match on CCX and CUCM for jtapi_1 and rmjtapi
   3. from CCX run sync and restart engine
   4. You may have to re-sync a couple of times


1. the T flag is not set so the CUCM never sees that the CCX is
integrated you have to do a manual step to get this up and running so you
can add the UCCX extension to users,


1. run this command on the CUCM Pub as stated in earleir emails  run
   sql update processconfig set paramvalue=T where paramname like
   '%nstalled%'
   2. NO need to reload it takes affect right away


1. CUC - I have had to reload the CUC server every time for MWIs and
to be able to leave message... This one I am not sure why I have no fix
other than to restart the server.
2. CUE - As noted in labs WB and Guide you will need to change
licenses depending on the lab you are working on.
3. Lab 2 CCXquestion there is a step missing in the script. a
generated prompt of zero calls ahead of you when you are the only one  in
queue = you do need the decrement step (thanks larry)

 If i recall or run into any more items of interest i will post back i am
 in last 5 days of prep and will be concentrating on the labs

 E


___
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www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 69, Issue 191

2011-11-27 Thread Edgar Feliz
I could not find it in the docs I found it in Cisco forum. Someone had
posted a similar issue as we have had and someone from Cisco posted the
command.

E

On Sun, Nov 27, 2011 at 6:41 PM, Jason Langenfeld
jlangenf...@prosysis.comwrote:

 Where is this sql command found in documentation?



 1. run sql update processconfig set paramvalue=T where paramname
 like '%nstalled%'







 Thanks,
 Jason Langenfeld  |  Systems Engineer
 Prosys
 6550 Carothers Parkway, Suite 420
 Franklin, TN 37067
 O 615.435.1540
 Email jlangenf...@prosysis.com






 On 11/27/11 9:49 AM, Larry Stern larry.st...@blackbox.com wrote:

 Randall
 I had the same exact problem with New Lab 1, after I ran the SQL command
 from the cucm cli, This fixed the not showing of the  IPCC extension
 field on my cucm enduser page, but after an hour or so of reverting,
 resyncing uccx I never got rmjtapi to show in CUCM as an application
 user. I tried adding it manually into CUCM , and then resyncing UCCX,
 still ng. Finally moved on.
 
 Sent from my iPad
 
 On Nov 27, 2011, at 10:13 AM, ccie_voice-requ...@onlinestudylist.com
 ccie_voice-requ...@onlinestudylist.com wrote:
 
  Send CCIE_Voice mailing list submissions to
 ccie_voice@onlinestudylist.com
 
  To subscribe or unsubscribe via the World Wide Web, visit
 http://onlinestudylist.com/mailman/listinfo/ccie_voice
  or, via email, send a message with subject or body 'help' to
 ccie_voice-requ...@onlinestudylist.com
 
  You can reach the person managing the list at
 ccie_voice-ow...@onlinestudylist.com
 
  When replying, please edit your Subject line so it is more specific
  than Re: Contents of CCIE_Voice digest...
 
 
  Today's Topics:
 
1. Re: CUCM/UCCX Intergration (Pradeep Kumar Sharma)
2. Re: CUCM/UCCX Intergration (Edgar Feliz)
3. Re: CUCM/UCCX Intergration (Edgar Feliz)
4. Re: ISDN Cause i = 0x80AF - Resource unavailable (Edgar Feliz)
 
 
  --
 
  Message: 1
  Date: Sun, 27 Nov 2011 19:02:29 +0530
  From: Pradeep Kumar Sharma sonu.netwo...@gmail.com
  To: CCIEVoiceKP ccievoic...@gmail.com
  Cc: Online Study ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration
  Message-ID:
 
 CALWZLGczqjzTX0jq-JEOGOLG=jW+=v9epdffuxzk5wj5avs...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  It seems your UCCX server is not integrated properly with CUCM.
 
  Use http://ccie-musketeers.blogspot.com/search/label/UCCX and fix it.
 
  -SONU-
 
  On Sun, Nov 27, 2011 at 12:37 PM, CCIEVoiceKP ccievoic...@gmail.com
 wrote:
 
  Have you re-syched the data yet?  Subsystems -- Unified CM Telephony
 --
  Data Resync
 
  KP
 
  On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.com
 wrote:
 
  HI,
  I am working on the New Lab 1 UCCX portion. I am suppose to add an
 UCCX
  extension to each user page, but the field to add the UCCX exntension
 is
  not there. I also did not see a rmjtapi user in CUCM. This is the
 user in
  the intergration page on UCCX.
 
  Any thoughts?
 
  Randall
 
 
  ___
  For more information regarding industry leading CCIE Lab training,
 please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com http://www.platinumplacement.com/
 
 
 
  ___
  For more information regarding industry leading CCIE Lab training,
 please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com
 
  -- next part --
  An HTML attachment was scrubbed...
  URL:
 /archives/ccie_voice/attachments/2027/4a72770d/attachment-0001.html
 
  --
 
  Message: 2
  Date: Sun, 27 Nov 2011 08:36:53 -0500
  From: Edgar Feliz ejzi...@gmail.com
  To: CCIEVoiceKP ccievoic...@gmail.com
  Cc: Online Study ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration
  Message-ID:
 
 cabxks5qfl_jbyttuxstdw9+vxw0z-w0_8-fpebwtxackx_u...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  Run this command on the CUCM Pub you do NOT have to reboot.
 
  run sql update processconfig set paramvalue=T where paramname like
  '%nstalled%'
 
  E
 
  On Sun, Nov 27, 2011 at 2:07 AM, CCIEVoiceKP ccievoic...@gmail.com
 wrote:
 
  Have you re-syched the data yet?  Subsystems -- Unified CM Telephony
 --
  Data Resync
 
  KP
 
  On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.com
 wrote:
 
  HI,
  I am working on the New Lab 1 UCCX portion. I am suppose to add an
 UCCX
  extension to each user page, but the field to add the UCCX exntension
 is
  not there. I also did not see a rmjtapi user in CUCM. This is the
 user in
  the intergration page on UCCX.
 
  Any thoughts?
 
  Randall
 
 
  ___
  For more information regarding

Re: [OSL | CCIE_Voice] Live Record

2011-11-27 Thread Edgar Feliz
I think Ash means you have to have a CTI route point with the # you will
call to record and have that forward all to VM/CUC. When you are on a call
then you can hit the conference soft-key and put in the number mentioned
above from the route-point and conference soft-key again to start recording
but there is not single key option...not that I am aware off.

E

PS

Amit Questions are allowed to be asked here if you don't want to
participate you don't have too mate.


On Sun, Nov 27, 2011 at 11:49 PM, Ashraf Ayyash ash.ayy...@gmail.comwrote:

 NO , you have to create conferance call with the Live record number

 Ash

 On Sun, Nov 27, 2011 at 9:52 PM, Ken Wyan kew...@gmail.com wrote:
  Is Live Record Softkey (LiveRcd) available with CUCM ( Call Manager
 Server
  based system ) ?
  ___
  For more information regarding industry leading CCIE Lab training, please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com
 
 ___
 For more information regarding industry leading CCIE Lab training, please
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Re: [OSL | CCIE_Voice] Live Record

2011-11-27 Thread Edgar Feliz
Hi ken,

Don't pay attention to A Mitstake...

You can ask all the questions you want here.. he is not the boss of us...
If he does not want to participate in a discussion he should just keep his
trap shut.

E

On Mon, Nov 28, 2011 at 12:20 AM, Ken Wyan kew...@gmail.com wrote:

 Dear  Sir (Amit),

 Thanks for your encouragement.

 Keep it going.

 Ken
 ( before asking forum , I saw LiveRcd available in CUCME , but in none of
 CUCM softkey templates had it or couldn't add for a new softkey template  )
 On Mon, Nov 28, 2011 at 10:25 AM, Amit Singh batraji...@yahoo.com wrote:

 Mate I guess u can login into cucm and have a quick look.

 I am 100% sure that is much faster than typing this email.

 Mate seems like ur not putting any efforts in studying from your side.



 Regards
 Amit

 Sent from my iPad

 On 28/11/2011, at 4:52 PM, Ken Wyan kew...@gmail.com wrote:

  Is Live Record Softkey (LiveRcd) available with CUCM ( Call Manager
 Server based system ) ?
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Re: [OSL | CCIE_Voice] live-record no bip

2011-11-26 Thread Edgar Feliz
No that is not it I had tried what you said below.

I found it in the CUCM service parameters.

Party Entrance Tone
https://10.10.210.10:8443/ccmadmin/serviceParamSave.do?showall=false#[image:
Required Field] True False True

Set this to false and you do not get a tone at the other side.

E

On Sat, Nov 26, 2011 at 8:10 AM, Gregg Malcolm gregg.malc...@gmail.comwrote:

 I haven't tried this but from the CUC help pages, you should be able to
 disable the beep completely by changing the timeout to zero.

 The following procedure is optional. It adjusts the interval between beeps
 while Cisco Unity Connection is recording a phone conversation.
  To Adjust the Live Record Beep Interval (Optional)
 Step 1 In Cisco Unity Connection Administration, expand *System Settings*,
 then click *Advanced  Telephony*.

1. On the Telephony Configuration page, in the Live Record Beep
Interval in Milliseconds field, enter the interval (in milliseconds)
between beeps when a phone conversation is being recorded by using the live
record feature.
If the setting is blank, the interval is 15,000 milliseconds. If the
setting is 0, the beep is disabled.
 2. Click *Save*.



 On Fri, Nov 25, 2011 at 9:52 PM, Edgar Feliz ejzi...@gmail.com wrote:

 If given the task of configuring live-record but part of the requirement
 states that the TONE NOT be played back to the users when recording has
 begun, how is this done? I have looked in CUC and CCM and don't see a way
 to stop this.

 E

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Re: [OSL | CCIE_Voice] CUC problems

2011-11-26 Thread Edgar Feliz
This is a VM setup you are talking about right? Cause I have seen things on
the VMs I have setup for labs that I normally don't see in hardware based.

E

On Sat, Nov 26, 2011 at 1:59 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 I had several CUC installations in Production Environment, and did not
 have any problems like that.

 I will try to collect the Traces for that call from CUCM and see what will
 happen

 On Sat, Nov 26, 2011 at 6:31 AM, Edgar Feliz ejzi...@gmail.com wrote:

 CUC reboot worked for me. I use PTs and CSSs. for whatever reason this
 has been pretty much a daily ting for me I can't figure out what it is.

 E


 On Fri, Nov 25, 2011 at 9:12 PM, Edgar Feliz ejzi...@gmail.com wrote:

 I am having the same problem today!!! I just rebooted the CUC and am
 waiting to see.

 E

 On Fri, Nov 25, 2011 at 6:02 PM, Gurpreet Singh Kukreja 
 tycoononway1...@gmail.com wrote:

 Hi,

 Try this:

 1) Make sure that under the Voice Mail Profile configuration, Voice
 Mail Box Mask has a mask of  if you're using four digit port numbers.
 2) Second, remove all the CSS and partitions if applied on the VM Port
 Group, the Hunt Pilot  the MWIs.

 Did you pick up Call Manager SDI traces yet? What do you see in them?
 Does this work fine internally if you dial from an IP Phone registered with
 the CM?


 - Gurpreet




 On Fri, Nov 25, 2011 at 2:21 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 I have tried to Restart CUC, as I do not have any issues the CUCM DB
 replication. But still not success :(

  On Fri, Nov 25, 2011 at 7:40 PM, Edgar Feliz ejzi...@gmail.comwrote:

 I have had this 2 x and in one I had a DB issue and had to do a db
 sync restart sub, the other I had to restart the CUC.

 E


 On Fri, Nov 25, 2011 at 9:01 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 As I know, the CUC has the G729 Support as well, So I do not think
 that the issue is because of Tarsncoder.

 My local Phones with G729 can leave a voice mail without a problem

   On Fri, Nov 25, 2011 at 5:17 PM, Ax julian76...@yahoo.com wrote:

  Hi,

 I had simila issue,for on-net calls is working fine but for off-net
 incoming call not so I had to insert an transcoder resource in mrgl of
 gateway.
 You can do another test without transcoder but be sure you are
 using g711 from pstn  gateway -- ccm-- ip phone.




 Ion


  --
 *From:* datucha123 datucha123 datucha...@gmail.com
 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Friday, November 25, 2011 12:35 PM
 *Subject:* [OSL | CCIE_Voice] CUC problems

  Hello,

 I have a problem with CUC.

 When the external (PSTN) caller leaves a VM for CUCM IP phones, the
 MWI does not lights up, and even the recorded message is not present 
 in the
 users mailbox. WHile the CUC tells the PSTN caller that the message has
 been received. But actually, I cannot have it in my Phones inbox, nor 
 the
 MWI is turned On.

 Why is it so?



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/




 ___
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 please visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/






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Re: [OSL | CCIE_Voice] CUC problems

2011-11-25 Thread Edgar Feliz
I have had this 2 x and in one I had a DB issue and had to do a db sync
restart sub, the other I had to restart the CUC.

E

On Fri, Nov 25, 2011 at 9:01 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 As I know, the CUC has the G729 Support as well, So I do not think that
 the issue is because of Tarsncoder.

 My local Phones with G729 can leave a voice mail without a problem

 On Fri, Nov 25, 2011 at 5:17 PM, Ax julian76...@yahoo.com wrote:

  Hi,

 I had simila issue,for on-net calls is working fine but for off-net
 incoming call not so I had to insert an transcoder resource in mrgl of
 gateway.
 You can do another test without transcoder but be sure you are using g711
 from pstn  gateway -- ccm-- ip phone.




 Ion


  --
 *From:* datucha123 datucha123 datucha...@gmail.com
 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Friday, November 25, 2011 12:35 PM
 *Subject:* [OSL | CCIE_Voice] CUC problems

  Hello,

 I have a problem with CUC.

 When the external (PSTN) caller leaves a VM for CUCM IP phones, the MWI
 does not lights up, and even the recorded message is not present in the
 users mailbox. WHile the CUC tells the PSTN caller that the message has
 been received. But actually, I cannot have it in my Phones inbox, nor the
 MWI is turned On.

 Why is it so?



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

___
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[OSL | CCIE_Voice] Documentation Access and Where to Go

2011-11-25 Thread Edgar Feliz
This is a list of what we know could be required urls and features for any
given lab and were to go to find it if needed. Please add to the list below
as this was just a starting point. My intent is to have the urls needed as
part of the 20-30 minute initial read and documentation of the lab and than
have a better idea of where to go for things that may come out of the blue.


SRND7x

   1. IPMA =  p936
   2. EM  = p929
   3. CUPS = p897

CCX Doc
ICD + 1button = Support  voice  CUCM  Contact Center Express

MVA = CUCM  help  search  mobile voice access  look for H323/PRI

CME Feature and configuration = Documentation  CME tech notes and config
guide

E
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Re: [OSL | CCIE_Voice] CUC problems

2011-11-25 Thread Edgar Feliz
I am having the same problem today!!! I just rebooted the CUC and am
waiting to see.

E

On Fri, Nov 25, 2011 at 6:02 PM, Gurpreet Singh Kukreja 
tycoononway1...@gmail.com wrote:

 Hi,

 Try this:

 1) Make sure that under the Voice Mail Profile configuration, Voice
 Mail Box Mask has a mask of  if you're using four digit port numbers.
 2) Second, remove all the CSS and partitions if applied on the VM Port
 Group, the Hunt Pilot  the MWIs.

 Did you pick up Call Manager SDI traces yet? What do you see in them? Does
 this work fine internally if you dial from an IP Phone registered with the
 CM?


 - Gurpreet




 On Fri, Nov 25, 2011 at 2:21 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 I have tried to Restart CUC, as I do not have any issues the CUCM DB
 replication. But still not success :(

 On Fri, Nov 25, 2011 at 7:40 PM, Edgar Feliz ejzi...@gmail.com wrote:

 I have had this 2 x and in one I had a DB issue and had to do a db sync
 restart sub, the other I had to restart the CUC.

 E


 On Fri, Nov 25, 2011 at 9:01 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 As I know, the CUC has the G729 Support as well, So I do not think that
 the issue is because of Tarsncoder.

 My local Phones with G729 can leave a voice mail without a problem

   On Fri, Nov 25, 2011 at 5:17 PM, Ax julian76...@yahoo.com wrote:

  Hi,

 I had simila issue,for on-net calls is working fine but for off-net
 incoming call not so I had to insert an transcoder resource in mrgl of
 gateway.
 You can do another test without transcoder but be sure you are using
 g711 from pstn  gateway -- ccm-- ip phone.




 Ion


  --
 *From:* datucha123 datucha123 datucha...@gmail.com
 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Friday, November 25, 2011 12:35 PM
 *Subject:* [OSL | CCIE_Voice] CUC problems

  Hello,

 I have a problem with CUC.

 When the external (PSTN) caller leaves a VM for CUCM IP phones, the
 MWI does not lights up, and even the recorded message is not present in 
 the
 users mailbox. WHile the CUC tells the PSTN caller that the message has
 been received. But actually, I cannot have it in my Phones inbox, nor the
 MWI is turned On.

 Why is it so?



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/




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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CUC problems

2011-11-25 Thread Edgar Feliz
CUC reboot worked for me. I use PTs and CSSs. for whatever reason this has
been pretty much a daily ting for me I can't figure out what it is.

E

On Fri, Nov 25, 2011 at 9:12 PM, Edgar Feliz ejzi...@gmail.com wrote:

 I am having the same problem today!!! I just rebooted the CUC and am
 waiting to see.

 E

 On Fri, Nov 25, 2011 at 6:02 PM, Gurpreet Singh Kukreja 
 tycoononway1...@gmail.com wrote:

 Hi,

 Try this:

 1) Make sure that under the Voice Mail Profile configuration, Voice
 Mail Box Mask has a mask of  if you're using four digit port numbers.
 2) Second, remove all the CSS and partitions if applied on the VM Port
 Group, the Hunt Pilot  the MWIs.

 Did you pick up Call Manager SDI traces yet? What do you see in them?
 Does this work fine internally if you dial from an IP Phone registered with
 the CM?


 - Gurpreet




 On Fri, Nov 25, 2011 at 2:21 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 I have tried to Restart CUC, as I do not have any issues the CUCM DB
 replication. But still not success :(

 On Fri, Nov 25, 2011 at 7:40 PM, Edgar Feliz ejzi...@gmail.com wrote:

 I have had this 2 x and in one I had a DB issue and had to do a db sync
 restart sub, the other I had to restart the CUC.

 E


 On Fri, Nov 25, 2011 at 9:01 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 As I know, the CUC has the G729 Support as well, So I do not think
 that the issue is because of Tarsncoder.

 My local Phones with G729 can leave a voice mail without a problem

   On Fri, Nov 25, 2011 at 5:17 PM, Ax julian76...@yahoo.com wrote:

  Hi,

 I had simila issue,for on-net calls is working fine but for off-net
 incoming call not so I had to insert an transcoder resource in mrgl of
 gateway.
 You can do another test without transcoder but be sure you are using
 g711 from pstn  gateway -- ccm-- ip phone.




 Ion


  --
 *From:* datucha123 datucha123 datucha...@gmail.com
 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Friday, November 25, 2011 12:35 PM
 *Subject:* [OSL | CCIE_Voice] CUC problems

  Hello,

 I have a problem with CUC.

 When the external (PSTN) caller leaves a VM for CUCM IP phones, the
 MWI does not lights up, and even the recorded message is not present in 
 the
 users mailbox. WHile the CUC tells the PSTN caller that the message has
 been received. But actually, I cannot have it in my Phones inbox, nor the
 MWI is turned On.

 Why is it so?



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/




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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com




___
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www.ipexpert.com

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[OSL | CCIE_Voice] live-record no bip

2011-11-25 Thread Edgar Feliz
If given the task of configuring live-record but part of the requirement
states that the TONE NOT be played back to the users when recording has
begun, how is this done? I have looked in CUC and CCM and don't see a way
to stop this.

E
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Re: [OSL | CCIE_Voice] User Import to Unity Connection from CallManager

2011-11-24 Thread Edgar Feliz
I've done 6-7 CUCM/CUC intergration in the practice labs and had only to do
a Sync when I had a DB issue on the CUCMs.

E

On Thu, Nov 24, 2011 at 3:59 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 I think you can do both as nothing will be ruined after the synch.

 On Thu, Nov 24, 2011 at 10:01 AM, Ken Wyan kew...@gmail.com wrote:

 Hi Guys,

 I have a very basic question with manually created users in CUCM. ( *No*AD 
 or LDAP integration)

 When integrating CUCM with CUC in the exam , is it sufficient to import
 users from CUCM to CUC  or  need to Sync users after importing ?

 I saw in a CCIE video ,  user import  sync both to be done in order.

 As mentioned in cisco docs , sync is required if any later change is done
 to CUCM users.

 Do you normally do both user import  user sync both or  do import only?

 Thanks

 Ken



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 visit www.ipexpert.com

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 ___
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[OSL | CCIE_Voice] Type/Plan not as should be.

2011-11-24 Thread Edgar Feliz
I am working a a lab that requires TEHO SiteA calls to siteB/PSTN = calls
are routed SB-RTR and if not available SG-SLRG.


   1. SB=H323
   2. Calls out of SB should have plan/type as ISDN/National
   3. calling number = 10
   4. called number = 7

My problem is that as seen on the output below my calling # plan/type are
not showing up as ISDN/National but rather as U/U. Called Plan/Type are
correct.

 Display i = 'SiteA phone 2'
Calling Party Number
SiteB-RTR# i = 0x0081, '2025552002'
Plan:Unknown, Type:Unknown *** Should
be ISDN/National.
Called Party Number i = 0x80, '8397263'
Plan:Unknown, Type:Unknown


Any Ideas??? I am including the SB translation part of the config.

voice translation-rule 10
 rule 1 /^3...$/ /408387\0/ type any subscriber plan any isdn
 rule 2 /^2...$/ /202555\0/ type any national plan any isdn
!
!
voice translation-profile 10
 translate calling 10
!
!
dial-peer voice 7 pots
 translation-profile outgoing 10 calls on translation profile
10 above
 destination-pattern 9[2-9]..
 port 0/0/0:23
 forward-digits 7
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Re: [OSL | CCIE_Voice] Type/Plan not as should be.

2011-11-24 Thread Edgar Feliz
OK I found the problem...Operator Error!!!

I had this under the s0/0/0.0.1 int,

*isdn map address ^[2-9] plan unknown type unknown *than i tried this isdn
map address *^[2-9].. plan unknown type unknown* but if you don't put
the the $ at the end it will not put the correct plat/type for the calling
number. so at last i put this which i normally had been doing but had a
brain fart i guess. So fixing the ISDN map fixed the calling plan/type.

This works!!

isdn map address ^[2-9]..$ plan unknown type unknown

On Thu, Nov 24, 2011 at 1:28 PM, Edgar Feliz ejzi...@gmail.com wrote:

 I am working a a lab that requires TEHO SiteA calls to siteB/PSTN =
 calls are routed SB-RTR and if not available SG-SLRG.


1. SB=H323
2. Calls out of SB should have plan/type as ISDN/National
3. calling number = 10
4. called number = 7

 My problem is that as seen on the output below my calling # plan/type are
 not showing up as ISDN/National but rather as U/U. Called Plan/Type are
 correct.

  Display i = 'SiteA phone 2'
 Calling Party Number
 SiteB-RTR# i = 0x0081, '2025552002'
 Plan:Unknown, Type:Unknown *** Should
 be ISDN/National.
 Called Party Number i = 0x80, '8397263'
 Plan:Unknown, Type:Unknown


 Any Ideas??? I am including the SB translation part of the config.

 voice translation-rule 10
  rule 1 /^3...$/ /408387\0/ type any subscriber plan any isdn
  rule 2 /^2...$/ /202555\0/ type any national plan any isdn
 !
 !
 voice translation-profile 10
  translate calling 10
 !
 !
 dial-peer voice 7 pots
  translation-profile outgoing 10 calls on translation profile
 10 above
  destination-pattern 9[2-9]..
  port 0/0/0:23
  forward-digits 7

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Re: [OSL | CCIE_Voice] Afterhours Support

2011-11-14 Thread edgar feliz
Honestly!!! 


You guys have to be kidding me. I want my money back for these labs. you sold 
us something you weren't prepared to support I have been fighting these for 2 
weeks now and fixes are coming rack are being prepped is what I have heard, 
and I am 2 weeks away from by lab date and I can't prepare the way I would 
like because you sold us BS.
 A very frustrated and pissed off


Edgar



From: Matt Just mj...@ipexpert.com
To: ejfeli...@yahoo.com ejfeli...@yahoo.com
Cc: supp...@proctorlabs.com supp...@proctorlabs.com
Sent: Monday, November 14, 2011 6:39 PM
Subject: Re: Afterhours Support


Edgar,

The  5 lab handbook doesn't work on any pods currently and wont until next week.

Regards,


Matt Just – IPexpert, Inc.

CTO – Chief Technical Officer
Email: mj...@ipexpert.com
Linkedin: http://www.linkedin.com/in/mattjust
Twitter: matt_just
Facebook: http://www.facebook.com/mattmjust
Office: (810) 326-1444 Ext: 331
Mobile +1 (810) 406-7218

Sent from my iPhone

On Nov 14, 2011, at 6:32 PM, Edgar Feliz nore...@ipexpert.com wrote:



           N: Edgar Feliz
           E: ejfeli...@yahoo.com
           Pod: CCIE  Voice 28
           UN: ejfeliz65
           D: 2011-11-14
           T: 06:31 PM Servers are NOT coming up!! 
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[OSL | CCIE_Voice] Modify button items vs Phone button template

2011-11-11 Thread edgar feliz
I am posing this questions because I have worked on the same lab consecutive 
days and had different results on this one item today from yesterday.

The question is form the 5 new labs, lab 1 Q-2.3. which to summarize, asks you 
to configure blf for phn1 ext 2001 on phn2. Today I configure this same as 
yesterday using the Modify Button Items on the phone by the lines. last night 
everything worked no issues, today I did the same thing while I could use the 
button (2) like a speed-dial to dial 2001 the blf part did not work the lamp 
never lit up when 2001 was on a call/ringing etc.

So my question is has anyone had issues with using this method Modify Button 
Items? the other option is a phone button template it would take a little 
longer but I know it works 100% of the time.

Any thoughts?

Thanks

Edgar___
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Re: [OSL | CCIE_Voice] Workbook 2 typos

2011-11-10 Thread edgar feliz
Did you configure the calling party transform pattern?  \.+44.! predot/prefix 
0? At least in other labs with similar question this was part of the 
configuration. because the number you see when ringing does not have the plus 
but in the missed calls you should see the plus and be able to dial out from 
the missed calls with + dialing.

Edgar



From: John McGaughey (jomcgaug) jomcg...@cisco.com
To: ccie_voice@onlinestudylist.com
Sent: Thursday, November 10, 2011 11:07 PM
Subject: [OSL | CCIE_Voice] Workbook 2 typos


Is it just me or is Workbook 2 full of mistakes?  For example, I’m in lab 5 
question 2.3.  It says calls from phone button 2 should appear in the history 
logs as +44 020 5943 2785.
 
The answer states to set the Subscriber Number Prefix to +44.  That’s incorrect 
since the call comes in like this.
 
Nov 11 03:46:14.567: ISDN Se0/2/0:23 Q931: RX - SETUP pd = 8  callref = 0x0089
    Bearer Capability i = 0x8090A2
    Standard = CCITT
    Transfer Capability = Speech
    Transfer Mode = Circuit
    Transfer Rate = 64 kbit/s
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Progress Ind i = 0x8583 - Origination address is non-ISDN
    Calling Party Number i = 0x4180, '2059432785'
    Plan:ISDN, Type:Subscriber(local)
    Called Party Number i = 0xA1, '2059434002'
    Plan:ISDN, Type:National
Nov 11 03:46:14.587: ISDN Se0/2/0:23 Q931: TX - CALL_PROC pd = 8  callref = 
0x8089
    Channel ID i = 0xA98381
    Exclusive, Channel 1
 
The correct answer in my opinion is that the Subscriber Number Prefix should be 
+440.
 
Also, it shows the National Prefix as +44:1.  The only call that comes across 
as National is from the 900 number.  Which comes in like this.
 
Nov 11 03:37:26.540: ISDN Se0/2/0:23 Q931: RX - SETUP pd = 8  callref = 0x0088
    Bearer Capability i = 0x8090A2
    Standard = CCITT
    Transfer Capability = Speech
    Transfer Mode = Circuit
    Transfer Rate = 64 kbit/s
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Progress Ind i = 0x8583 - Origination address is non-ISDN
    Calling Party Number i = 0x2180, '9004522138'
    Plan:ISDN, Type:National
    Called Party Number i = 0xA1, '2059434002'
    Plan:ISDN, Type:National
 
If I use that prefix the calling number shows up as +44004522138.
 
I’m using my own rack but using the configs provided by ipexpert’s website.
 
That’s just one example.  I’ve ran into many of these.  Anyone else notice 
similar things?
 
John
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Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600

2011-10-30 Thread edgar feliz
OK, I thought I had done the CUCM first but i may have been working both at the 
same time. The configs all look right APP/USER ports etc. I guess I may have 
gotten ahead of myself. I am reworking that lab.

Thanks

Edgar



From: Vik Malhi vma...@ipexpert.com
To: edgar feliz ejfeli...@yahoo.com
Cc: CCIE ccie_voice@onlinestudylist.com
Sent: Sunday, October 30, 2011 12:24 PM
Subject: Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600


Well if the CTI Ports are down that is why it is not working. You HAVE to do 
the UCM work before the CUE side of the integration. You should add the CTI 
Ports- and supply the jtapi user credentials. This jtapi user should be 
associated to the cti rp and cti ports and have standard cti enabled.

ccn subsystem jtapi
 ctiport 3601 3602 
 ccm-manager address 10.10.210.11 10.10.210.10
 ccm-manager credentials ..
 end subsystem

Vik Malhi – CCIE #13890 
Managing Partner - IPexpert, Inc.

Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130 
Mailto: vma...@ipexpert.com




On Oct 28, 2011, at 7:31 PM, edgar feliz wrote:

Hi Vic, thanks for the response,
 
here is the some of the output fomr the show command, The VM (3600) port shows 
registered the other two (3601/3602) do not. Also all I get is a fast busy 
when I try to dial the 3600 or press the vm service button.
 
ccn trigger jtapi phonenumber 3600
 application voicemail
 enabled
 locale en_US
 maxsessions 6
 end trigger
 
ccn subsystem jtapi
 ccm-manager address 10.10.210.11 10.10.210.10 
 ccm-manager credentials hidden 
OSGGx1TGBhk3jDxWVhfWhUnfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP
 end subsystem



From: Vik Malhi vma...@ipexpert.com
To: edgar feliz ejfeli...@yahoo.com
Cc: CCIE ccie_voice@onlinestudylist.com
Sent: Friday, October 28, 2011 10:02 PM
Subject: Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600


Are CTI Rps and CTI ports registered?


When you sh run inside the cue cli do you see the CTI ports under ccn 
subsystem jtapi? Also check the ccn trigger jtapi phonenumber


Do you hear annunciatior or fast busy?


More info pls


-- 
Vik Malhi – CCIE #13890
Managing Partner / Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130 

On Oct 28, 2011, at 5:29 PM, edgar feliz ejfeli...@yahoo.com wrote:


   I have set up CUE to CCM configured CTI ports, vm profile etc. double 
checked with all the IE guide and can't get BR2 phone to dial into VM? Any 
Ideas? Users imported int0 CUE OK, CCM can see the 10.10.202.2 CUE when I look 
at CTI RPs.
 
EJF
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[OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600

2011-10-28 Thread edgar feliz
   I have set up CUE to CCM configured CTI ports, vm profile etc. double 
checked with all the IE guide and can't get BR2 phone to dial into VM? Any 
Ideas? Users imported int0 CUE OK, CCM can see the 10.10.202.2 CUE when I look 
at CTI RPs.
 
EJF___
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Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600

2011-10-28 Thread edgar feliz
Hi Vic, thanks for the response,
 
here is the some of the output fomr the show command, The VM (3600) port shows 
registered the other two (3601/3602) do not. Also all I get is a fast busy when 
I try to dial the 3600 or press the vm service button.
 
ccn trigger jtapi phonenumber 3600
 application voicemail
 enabled
 locale en_US
 maxsessions 6
 end trigger
 
ccn subsystem jtapi
 ccm-manager address 10.10.210.11 10.10.210.10 
 ccm-manager credentials hidden 
OSGGx1TGBhk3jDxWVhfWhUnfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP
 end subsystem




From: Vik Malhi vma...@ipexpert.com
To: edgar feliz ejfeli...@yahoo.com
Cc: CCIE ccie_voice@onlinestudylist.com
Sent: Friday, October 28, 2011 10:02 PM
Subject: Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600


Are CTI Rps and CTI ports registered?

When you sh run inside the cue cli do you see the CTI ports under ccn 
subsystem jtapi? Also check the ccn trigger jtapi phonenumber

Do you hear annunciatior or fast busy?

More info pls


-- 
Vik Malhi – CCIE #13890
Managing Partner / Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130 

On Oct 28, 2011, at 5:29 PM, edgar feliz ejfeli...@yahoo.com wrote:


   I have set up CUE to CCM configured CTI ports, vm profile etc. double 
checked with all the IE guide and can't get BR2 phone to dial into VM? Any 
Ideas? Users imported int0 CUE OK, CCM can see the 10.10.202.2 CUE when I look 
at CTI RPs.
 
EJF
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Re: [OSL | CCIE_Voice] Router with problem

2011-10-09 Thread edgar feliz
Yeah there is a couple of routers on the vrack with that problem I have one on 
rack 15 and have had rack 20 also. You can still work it won't bother the 
configuration. If you want turn logging to console off and it won't be so 
annoying.

EJF




From: Emanuel Damasceno aedamasc...@gmail.com
To: ccie_voice@onlinestudylist.com
Sent: Sunday, October 9, 2011 6:54 PM
Subject: [OSL | CCIE_Voice] Router with problem


The router that represents Branch 2, on Voice vRack pod 20 is warning me of a 
malfunction:
Oct 10 02:52:19.588: %ENVMON-3-FAN_FAILED: Fan 2 not rotating
Just to let you know. 
Antonio Emanuel Damasceno
CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written)
CompTIA Network+



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[OSL | CCIE_Voice] Overlapping lab questions

2011-10-09 Thread edgar feliz
I came across this today when doing labs and had not paid attention to this 
before but today I did a lot of things from memory even though the order was 
not same as lab book. I have a question that requires that the PSTN phone line 
2 ( +12123942123) call hqphn2 (2123945002), in the missed calls I should see 
+12123945002 and I do but later on there is a mobility question with hqohn2 
extension 5002 as the user's extension so now when I dial 2123945002 from the 
PSTN phone I do not see missed called from +12123942123 but from 5002 as it 
correct  for this question but because of the overlap both questions answers 
are not seen. 


What is the best or any way to handle this? lets say in the real labs if you 
read as suggested the entire lab book and configure like things in same time 
frame as  did today both work but the result of one is not visible.


EJF
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Re: [OSL | CCIE_Voice] Overlapping lab questions

2011-10-09 Thread edgar feliz
And another question would be do we bring this up to the proctor, or how do 
they grade this???

EJF




From: edgar feliz ejfeli...@yahoo.com
To: CCIE ccie_voice@onlinestudylist.com
Sent: Sunday, October 9, 2011 9:04 PM
Subject: [OSL | CCIE_Voice] Overlapping lab questions


I came across this today when doing labs and had not paid attention to this 
before but today I did a lot of things from memory even though the order was 
not same as lab book. I have a question that requires that the PSTN phone line 
2 ( +12123942123) call hqphn2 (2123945002), in the missed calls I should see 
+12123945002 and I do but later on there is a mobility question with hqohn2 
extension 5002 as the user's extension so now when I dial 2123945002 from the 
PSTN phone I do not see missed called from +12123942123 but from 5002 as it 
correct  for this question but because of the overlap both questions answers 
are not seen. 


What is the best or any way to handle this? lets say in the real labs if you 
read as suggested the entire lab book and configure like things in same time 
frame as  did today both work but the result of one is not visible.


EJF

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Re: [OSL | CCIE_Voice] Router with problem

2011-10-09 Thread edgar feliz
Been there done that...But as you can see not fixed!!!

EJF



From: Rrcrumm rrcr...@yahoo.com
To: Emanuel Damasceno aedamasc...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; IPexpert 
Tech Support Team supp...@ipexpert.com
Sent: Sunday, October 9, 2011 10:30 PM
Subject: Re: [OSL | CCIE_Voice] Router with problem


This should be addressed to support.

Randall

Sent from my iPhone

On Oct 9, 2011, at 3:54 PM, Emanuel Damasceno aedamasc...@gmail.com wrote:


The router that represents Branch 2, on Voice vRack pod 20 is warning me of a 
malfunction:
Oct 10 02:52:19.588: %ENVMON-3-FAN_FAILED: Fan 2 not rotating
Just to let you know. 
Antonio Emanuel Damasceno
CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written)
CompTIA Network+



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Re: [OSL | CCIE_Voice] Why not set CSS in the Device Pool?

2011-10-08 Thread edgar feliz
You can have DPs for devices in that will have similar behavior and set the 
CSSs accordingly in the DPs.

EJF




From: Ken Wyan kew...@gmail.com
To: Mark Reed marklr...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Sent: Saturday, October 8, 2011 3:47 PM
Subject: Re: [OSL | CCIE_Voice] Why not set CSS in the Device Pool?


Normally different devices have different calling privileges ( local only , 
international , premium)  that's why a single CSS is not used per DP.
 
One exception is Line Device approach where CSS can be applied to DP  call 
block RPs are included in line CSS of individual phones. But with Local RGs , 
this approach has gone obsolete
 

 
On Sat, Oct 8, 2011 at 10:25 PM, Mark Reed marklr...@gmail.com wrote:

I never see this done for some reason.  With the number of phones were talking 
about it isn't a huge time saver,  but I'll take every second I can get at this 
point.  I did my entire mock lab this way yesterday and everything worked great 
except the aar-group which I still needed on the line itself.  But everything 
worked exactly like I had set it per device.  Am I missing something or am I 
just out of the loop and people are doing it this way?

-- 
Thanks,

Mark L Reed

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[OSL | CCIE_Voice] Can't Connect to PL VRACK Servers via Hardware VPN

2011-10-07 Thread edgar feliz
Is anyone else having issues connecting to the Servers on their VRACK sessions?

EJF
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[OSL | CCIE_Voice] Proctor labs issues AGAIN!!??

2011-10-01 Thread edgar feliz
Is anybody else having issues getting to their session, been fighting this for 
over an hour. I actually had a session from 12 AM and hoped it would continue 
but I went to sleep for a couple of hours and return to not being able to 
access anything. I can login now but so far that's as far as I have been able 
to get, and even that took almost an hour.

EJF___
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Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

2011-10-01 Thread edgar feliz
Its not you it's them I am having same issues, they say they are looking at it.

Now I can get to the login page and start my session but it is slow routers are 
not always responding.

Edgar



From: pithog...@yahoo.com pithog...@yahoo.com
To: ccie_voice@onlinestudylist.com
Sent: Saturday, October 1, 2011 6:58 AM
Subject: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

I am on a current session now and I can't access proctorlabs via hardware vpn, 
my router can ping internet but my laptop connected to the router can't access 
the internet so as to initiate hardware VPN .

Please help, you can even connect to my laptop via team viewer .

Experts kindly assist

Pithohoil
Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed internet 
service with Etisalat easy net, available at all our experience centres
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Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

2011-10-01 Thread edgar feliz
There is nothing we can do until Proctorlabs resolves the ISP issue they are 
having. I was told anyone who had issues would get a replacement voucher.

EJF



From: pithog...@yahoo.com pithog...@yahoo.com
To: edgar feliz ejfeli...@yahoo.com; ccie_voice@onlinestudylist.com 
ccie_voice@onlinestudylist.com
Sent: Saturday, October 1, 2011 11:10 AM
Subject: Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now


I use a 881 router, I entered the crypto ...connect command but I did not get 
the xauth request, that was an indication that there was a problem.

Another challenge I had was that I could not access the internet from my laptop 
 despite the fact that my router was assigning ip address to my laptop, my 
laptop could not resolve dns , I will appreciate assistance please.

Thanks for timely response.

Regards

Pithogoil
Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed internet 
service with Etisalat easy net, available at all our experience centres


From:  edgar feliz ejfeli...@yahoo.com 
Date: Sat, 1 Oct 2011 07:37:59 -0700 (PDT)
To: pithog...@yahoo.compithog...@yahoo.com; 
ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
ReplyTo:  edgar feliz ejfeli...@yahoo.com 
Subject: Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

Its not you it's them I am having same issues, they say they are looking at it.

Now I can get to the login page and start my session but it is slow routers are 
not always responding.

Edgar



From: pithog...@yahoo.com pithog...@yahoo.com
To: ccie_voice@onlinestudylist.com
Sent: Saturday, October 1, 2011 6:58 AM
Subject: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

I am on a current session now and I can't access proctorlabs via hardware vpn, 
my router can ping internet but my laptop connected to the router can't access 
the internet so as to initiate hardware VPN .

Please help, you can even connect to my laptop via team viewer .

Experts kindly assist

Pithohoil
Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed internet 
service with Etisalat easy net, available at all our experience centres
___
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Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

2011-10-01 Thread edgar feliz
To many people are having issue for it to be operator error it's definitely 
something on their end ISP etc.

EJF 



From: Chris Martin clm.c...@gmail.com
To: pithog...@yahoo.com
Cc: ccie_voice@onlinestudylist.com
Sent: Saturday, October 1, 2011 10:31 AM
Subject: Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now


You can open a case with support, that would be the quickest way to get their 
help.  Did you use IPE's configuration for hardware vpn?  What type of router 
are you using?  Did you enter the commands to initiate the xauth?

Chris


On Sat, Oct 1, 2011 at 5:58 AM, pithog...@yahoo.com wrote:

I am on a current session now and I can't access proctorlabs via hardware vpn, 
my router can ping internet but my laptop connected to the router can't access 
the internet so as to initiate hardware VPN .

Please help, you can even connect to my laptop via team viewer .

Experts kindly assist

Pithohoil
Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed internet 
service with Etisalat easy net, available at all our experience centres
___
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Re: [OSL | CCIE_Voice] proctorlabs website ?

2011-09-25 Thread edgar feliz
I just tried again and can't get to site.

EJF



From: William Affeldt william.affe...@yahoo.com
To: edgar feliz ejfeli...@yahoo.com
Cc: Rrcrumm rrcr...@yahoo.com; Ken Wyan kew...@gmail.com; 
ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sunday, September 25, 2011 6:48 PM
Subject: Re: [OSL | CCIE_Voice] proctorlabs website ?


Is anyone else having problems with proctor labs right now? 

Sent from my iPhone

On Sep 24, 2011, at 6:27 PM, edgar feliz ejfeli...@yahoo.com wrote:


NO I trade this morning for a while and it was not available.


EJF




From: Rrcrumm rrcr...@yahoo.com
To: Ken Wyan kew...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Saturday, September 24, 2011 5:28 PM
Subject: Re: [OSL | CCIE_Voice] proctorlabs website ?


If it is you should send an email to supp...@proctorlabs.com  and someone 
should be able to help you


Randall

Sent from my iPhone

On Sep 24, 2011, at 8:34 AM, Ken Wyan kew...@gmail.com wrote:


Today proctorlabs.com website is down ??? 
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Re: [OSL | CCIE_Voice] proctorlabs website ?

2011-09-24 Thread edgar feliz
NO I trade this morning for a while and it was not available.

EJF



From: Rrcrumm rrcr...@yahoo.com
To: Ken Wyan kew...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Saturday, September 24, 2011 5:28 PM
Subject: Re: [OSL | CCIE_Voice] proctorlabs website ?


If it is you should send an email to supp...@proctorlabs.com  and someone 
should be able to help you

Randall

Sent from my iPhone

On Sep 24, 2011, at 8:34 AM, Ken Wyan kew...@gmail.com wrote:


Today proctorlabs.com website is down ??? 
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Re: [OSL | CCIE_Voice] ntp

2011-09-19 Thread edgar feliz
Ray,

I have never put the NTP master command on my routers and I have not had an 
issue with the CUCM sync to the (HQ) router. I use mainly the VRACKs from 
proctor, and in my little lab at home it's the same. As long as I have set the 
time I want in the PSTN router for the HQ router to sync off I have not had an 
issue.

EJF



From: Ray jonha...@yahoo.com
To: Justin Barksdale jus...@barksdale.net; ccie_voice@onlinestudylist.com 
ccie_voice@onlinestudylist.com
Sent: Monday, September 19, 2011 1:12 AM
Subject: Re: [OSL | CCIE_Voice] ntp


Ntp is not slow when i put the ntp master command on HQ router and the cucm syn 
right the way. i took off the ntp master from the hq and only have ntp server 
and ntp source lo0. and reboot the cucm and the hq. and let it set for 4hrs to 
5hrs and the cucm would not syn, but the moment i put the ntp master it syns 
within 2 mins... you can try it in you lab and let me know.. make sure you 
reboot the cucm and the hq router...



From: Justin Barksdale jus...@barksdale.net
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sunday, September 18, 2011 9:14 PM
Subject: Re: [OSL | CCIE_Voice] ntp

NTP is by nature slow.  An earlier post pointed out the correct method.  If the 
question wants you to sync hq with the pstn and then ccm with hq then the ntp 
master command will break this requirement.  All that is required on the HQ 
router is ntp server x.x.x.x

This is also mentioned in Vic's class class as well as Cisco documentation.  
The important key here is that the ntp master command can cause you to override 
a valid external time source and thus can be very dangerous.  As long as the 
ntp server command points to a valid and reachable ntp source then the hq 
router will sync and you WILL be able to sync ucm without the ntp master 
command.

Justin
 Barksdale
CCIE# 29866
Voice

Sent from my iPhone 4.  Please excuse any typos.

On Sep 18, 2011, at 7:39 PM, ccie_voice-requ...@onlinestudylist.com wrote:

 Re: [OSL | CCIE_Voice] ntp
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Re: [OSL | CCIE_Voice] MVA unable to match RD

2011-09-18 Thread edgar feliz
I just had similar issue and it was that I had not checked the box in the RDP 
to be associated with the users line 5002 in my case.

EJF



From: Sanoj Thomas san...@yahoo.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sunday, September 18, 2011 3:48 AM
Subject: [OSL | CCIE_Voice] MVA unable to match RD


Hi All,

CUCM does not recognize the Remote destination based on the caller ID and 
prompt for a pin number, but instead the system prompts to enter the remote 
destination number.
Call manager is configured via the Service Parameter to match (partial match) 
Remote destination based on the last 10 digits of the caller ID.

System configuration.

PSTNH323GW--CUCM
RD (config in CUCM)  - 912012517295
CUCM v 7.0.2

I can see the RD in the attached trace file, but i believe it fails the match 
and give a java.lang.NullPointerException. Has anyone seen a similar issue. 
Or is it misconfig.

CUCM MVA trace file


2011-09-17 11:30:25,263 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile - [CCM_IVR]:: IVR get Filename 
2011-09-17 11:30:25,264 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() 
2011-09-17 11:30:25,265 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 1.au and  
locale =
 en_US
2011-09-17 11:30:25,265 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 1.au and  
locale =
 en_US
2011-09-17 11:30:25,267 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile - [CCM_IVR]:: getting the file now
2011-09-17 11:30:25,270 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::execute(): action start
2011-09-17 11:30:25,270 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::srcdir: en_US
2011-09-17 11:30:25,271 DEBUG [http-8080-Processor22]
 controller.IVRCalleridLookup - 
[CCM_IVR]::-new-code-got-callerid-as-remotedest: 2012517295
2011-09-17 11:30:25,271 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::srcdir: en_US   
sessionid:6D51E867730917739018955E44680948
2011-09-17 11:30:25,272 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::remotedest: 2012517295   
sessionid:6D51E867730917739018955E44680948
2011-09-17 11:30:25,272 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::Host: 10.10.3.10:8080
2011-09-17 11:30:25,273 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::Content-Type 
application/x-www-form-urlencoded
2011-09-17 11:30:25,273 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::Connection: close
2011-09-17 11:30:25,273 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::Accept: text/vxml, text/x-vxml, 
application/vxml,
 application/x-vxml, application/voicexml, application/x-voicexml, text/plain, 
text/html, audio/basic, audio/wav, multipart/form-data, application/octet-stream
2011-09-17 11:30:25,274 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::User-Agent: Cisco-IOS-C3745/12.4
2011-09-17 11:30:25,274 DEBUG [http-8080-Processor22] util.IVRDBInterface - 
[CCM_IVR]--- getUserIdFromRemoteDestination- DB Query --   remoteDest  = 
2012517295
2011-09-17 11:30:25,277 DEBUG [http-8080-Processor22] util.IVRDBInterface - 
[CCM_IVR]--- isPartialMatchEnabled --DB Query --   sql:  = select paramValue 
from processconfig where paramName ='RemDestCallerIDMatchType' and tkService 
='0' 
2011-09-17 11:30:25,314 DEBUG [http-8080-Processor22] util.IVRDBInterface - 
[CCM_IVR]--- isPartialMatchEnabled --DB Query --   RemDestCallerIDMatchType:  
= 1
2011-09-17 11:30:25,315 DEBUG [http-8080-Processor22] util.IVRDBInterface -
 [CCM_IVR]--- getUserIdFromRemoteDestination()  -- partial match enabled
2011-09-17 11:30:25,315 DEBUG [http-8080-Processor22] util.IVRDBInterface - 
[CCM_IVR]--- getNumDigitsPartialMatch --DB Query --   sql:  = select 
paramValue from processconfig where paramName ='RemDestCallerIDMatchDigits' and 
tkService ='0' 
2011-09-17 11:30:25,316 DEBUG [http-8080-Processor22] util.IVRDBInterface - 
[CCM_IVR]IVRDBInterface.getUserIdFromRemoteDestination() 
--java.lang.NullPointerException
2011-09-17 11:30:25,316 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::ccmusername: null
2011-09-17 11:30:25,317 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]::ccmusername:is Null ?? null
2011-09-17 11:30:25,317 DEBUG [http-8080-Processor22] 
controller.IVRCalleridLookup - [CCM_IVR]IVRCallerIdLookup::execute(): 
ccmusername is Null and action forwarding IVRUserid 
2011-09-17 11:30:28,779 DEBUG
 [http-8080-Processor25] controller.IVRGetAudioFile - [CCM_IVR]:: IVR get 
Filename 
2011-09-17 11:30:28,779 DEBUG [http-8080-Processor25] 
controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() 
2011-09-17 11:30:28,780 DEBUG [http-8080-Processor25] 

[OSL | CCIE_Voice] CUPC Certificate error

2011-09-01 Thread edgar feliz
I am getting the following error when I try to login with CUPC Ver 7.1.

Login failed. Certificate verification failed
Make sure you have a valid certificate.

Has anyone else had this issue?

EJ___
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Re: [OSL | CCIE_Voice] CUPC Certificate error

2011-09-01 Thread edgar feliz
Great, thanks for the reply Ash.

EJ



From: Ashraf Ayyash ash.ayy...@gmail.com
To: edgar feliz ejfeli...@yahoo.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Thursday, September 1, 2011 4:59 AM
Subject: Re: [OSL | CCIE_Voice] CUPC Certificate error

this error mean that you have install/run CUPC version not compatible
with your CUPS Version ,
CUPC 7.1 is only compatible with CUPS version 7.0.5

http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html#wp40258

downgrade your CUPC or upgrade your CUPS

Ash



On Thu, Sep 1, 2011 at 7:04 AM, edgar feliz ejfeli...@yahoo.com wrote:
 I am getting the following error when I try to login with CUPC Ver 7.1.
 Login failed. Certificate verification failed
 Make sure you have a valid certificate.
 Has anyone else had this issue?
 EJ
 ___
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 visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] CUPC Certificate error

2011-09-01 Thread edgar feliz
I downgraded to 7.02(a) CUPC but now I am getting unable to connect to login 
server message, but I place the presence service on two phone and I can login 
on those no problem.

EJ


From: Ashraf Ayyash ash.ayy...@gmail.com
To: edgar feliz ejfeli...@yahoo.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Thursday, September 1, 2011 4:59 AM
Subject: Re: [OSL | CCIE_Voice] CUPC Certificate error

this error mean that you have install/run CUPC version not compatible
with your CUPS Version ,
CUPC 7.1 is only compatible with CUPS version 7.0.5

http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html#wp40258

downgrade your CUPC or upgrade your CUPS

Ash



On Thu, Sep 1, 2011 at 7:04 AM, edgar feliz ejfeli...@yahoo.com wrote:
 I am getting the following error when I try to login with CUPC Ver 7.1.
 Login failed. Certificate verification failed
 Make sure you have a valid certificate.
 Has anyone else had this issue?
 EJ
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

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 www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] CUPC Certificate error

2011-09-01 Thread edgar feliz
Ash yes hosts file was OK...I ended up having to go back to CUPC version 
7.0(1)...not a lot of backward compatibility is there!!!

EJ



From: Ashraf Ayyash ash.ayy...@gmail.com
To: edgar feliz ejfeli...@yahoo.com
Sent: Thursday, September 1, 2011 9:31 AM
Subject: Re: [OSL | CCIE_Voice] CUPC Certificate error

hosts file on the PC ?

On Thu, Sep 1, 2011 at 3:23 PM, edgar feliz ejfeli...@yahoo.com wrote:
 I downgraded to 7.02(a) CUPC but now I am getting unable to connect to login
 server message, but I place the presence service on two phone and I can
 login on those no problem.
 EJ
 
 From: Ashraf Ayyash ash.ayy...@gmail.com
 To: edgar feliz ejfeli...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Sent: Thursday, September 1, 2011 4:59 AM
 Subject: Re: [OSL | CCIE_Voice] CUPC Certificate error

 this error mean that you have install/run CUPC version not compatible
 with your CUPS Version ,
 CUPC 7.1 is only compatible with CUPS version 7.0.5

 http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html#wp40258

 downgrade your CUPC or upgrade your CUPS

 Ash



 On Thu, Sep 1, 2011 at 7:04 AM, edgar feliz ejfeli...@yahoo.com wrote:
 I am getting the following error when I try to login with CUPC Ver 7.1.
 Login failed. Certificate verification failed
 Make sure you have a valid certificate.
 Has anyone else had this issue?
 EJ
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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Re: [OSL | CCIE_Voice] Vol 2 WB 6 - QoS question 7.1

2011-08-28 Thread edgar feliz
Well since the WB questions on based on a whole lab and just broken down by 
areas for the individual labs I would say that yes the PG answer was based on 
the prior questions's configuration with the new 7.1 requirements.

EJ



From: Geoghegan, Stuart stuart.geoghe...@ngbailey.co.uk
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sunday, August 28, 2011 11:10 AM
Subject: [OSL | CCIE_Voice] Vol 2 WB 6 - QoS question 7.1


 
Hi All,
 
I have been working through the QoS question in 7.1 in Vol 2
WB 6 – specifically the configuration required between HQ  BR2.
 
The portion of the question for HQ   Br2 states
that :
 
The PVC speed is 1024 kbps
 
Configure LLQ between HQ  BR2.  Compression is
not required on the Link. 
 
Between HQ  BR2 provision enough bandwidth in the
priority queue for 8 g729 calls
 
Use 5% of the unused BW for signalling traffic
 
 
Using the QoS SRND guide I have based my L2 overhead per call
as 4 bytes  (page 291 table 6-1)
 
 
This works out to be 25.6 kbps per call or rounded up to 26
kbps per call. 
 
[4+40+20]/20 x 8000
 
 Multiply this by 8 for the question requirement and my
LLQ bandwidth is configured as 208 kbps
 
 
In the proctor guide the figure of 224 kbps is used. 
 
If you reverse engineer the calculation this works out to be
10 bytes of Layer 2 overhead – which is the L2 bandwidth required for
MLP.
 
Although the question for BR2  HQ doesn’t specify
MLP, the previous portion for HQ to BR1 does.
 
So am I being stupid here   this should imply that
simply because HQ router has 2 frame-relay sub-interface/ PVCs on one physical
interface – you assume the MLP overhead for both PVCs and as such use
that in the calculation?
 
Kind regards
 
StuartThe information contained in this e-mail (and attachments) is 
confidential. It must not be read, copied, disclosed, printed, forwarded, 
relied upon or used by any person other than the intended recipient. 
Unauthorised use, disclosure or copying is strictly prohibited. If you have 
received this e-mail in error immediately contact the sender and 
postmas...@ngbailey.co.uk then permanently delete it and any attachments. NG 
Bailey may monitor email where it is considered proportionate to any perceived 
risk. This email has been sent on behalf of an NG Bailey company, if in doubt 
ask the sender to clarify which company.  The NG Bailey companies include NG 
Bailey Limited (342778), NG Bailey IT Services Limited (2338401), NG Bailey 
Facilities Services Limited (5472032), Kedington (Northern Ireland) Limited (NI 
31145) and NG Bailey Group Limited (1490238).  All of the companies, except 
Kedington (Northern Ireland) Limited, are registered in England with
 registered office at Denton Hall Ilkley West Yorkshire LS29 0HH.  Kedington 
(Northern Ireland) Limited is registered in Northern Ireland and its registered 
office is c/o Carson McDowell, Murray House, Murray Street, Belfast, BT1 6DN. 
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Re: [OSL | CCIE_Voice] (no subject)

2011-08-27 Thread edgar feliz
Also make sure under Remote Destination Configuration that you [/] the line 
associating box. I just went through this and that was my issue.

EJ



From: Alex Goh ncsalex@gmail.com
To: Mini Me cciev.min...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Saturday, August 27, 2011 8:50 PM
Subject: Re: [OSL | CCIE_Voice] (no subject)


check also did the mva was enable for the user under end user page.


On Sat, Aug 27, 2011 at 1:00 AM, Mini Me cciev.min...@gmail.com wrote:

Did you enable it in Service Parameters?


HTH

From:  Ray jonha...@yahoo.com
Reply-To:  Ray jonha...@yahoo.com
Date:  Fri, 26 Aug 2011 08:32:19 -0700 (PDT)
To:  ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject:  [OSL | CCIE_Voice] (no subject)



when i dial 3033300 from 525 pstn 2nd and put the pin 12345, MVA IVR ask 
me to 
press 2 to turn on the remote dest or press 3 to turn it off.. it doesnt let me 
dial any extension.. help

 
SB#sho run
Building configuration...




Current configuration : 5001 bytes
!
! Last configuration change at 05:10:01 CST Fri Aug 26 2011
! NVRAM config last updated at 23:36:37 CST Mon Aug 22 2011
!
version 15.0
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname BR1
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
!
!
clock timezone CST -6
clock summer-time CST recurring
network-clock-participate wic 2 
!
dot11 syslog
ip source-route
!
!
ip cef
!
ip dhcp pool phone
   network 177.2.11.0 255.255.255.0
   default-router 177.2.11.1 
   option 150 ip 10.11.11.19 
!
!
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!         
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class h323 1
  h225 timeout tcp establish 3
  call start slow
!
!
!
voice translation-rule 1
 rule 1 // // type any unknown plan any isdn
!
voice translation-rule 2
 rule 1 // // type any subscriber plan any isdn
!
voice translation-rule 3
 rule 1 // // type any national plan any isdn
!
voice translation-rule 4
 rule 1 // // type any international plan any isdn
!         
voice translation-rule 5
 rule 1 /.*\(3...\)/ /\1/
!
voice translation-rule 11
 rule 1 /^3...$/ /972303/ type any unknown plan any isdn
!
voice translation-rule 12
 rule 1 /^3...$/ /303/ type any subscriber plan any isdn
!
voice translation-rule 13
 rule 1 /^3...$/ /972303/ type any national plan any isdn
!
voice translation-rule 14
 rule 1 /^3...$/ /+1972303/ type any international plan any isdn
 rule 2 /^4...$/ /+8522404/ type any international plan any isdn
!
!
voice translation-profile 911
 translate calling 11
 translate called 1
!
voice translation-profile INC
 translate called 5
!
voice translation-profile intl
 translate calling 14
 translate called 4
!
voice translation-profile local
 translate calling 12
 translate called 2
!
voice translation-profile nat
 translate calling 13
 translate called 3
!
!
voice-card 0
!
!
application
 service mva http://10.11.11.19:8080/ccmivr/pages/IVRMainpage.vxml
 !
!
!
!         
!
!
license udi pid CISCO2811 sn FTX0902D1ZG
!
redundancy
!
!
controller T1 0/2/0
 pri-group timeslots 1-4,24
!
! 
!
!
!
!
!
!
!
interface Loopback0
 ip address 18.1.1.1 255.255.255.0
 ip ospf network point-to-point
 h323-gateway voip interface
 h323-gateway voip bind srcaddr 18.1.1.1
 !
!
interface FastEthernet0/0
 no ip address
 shutdown
 duplex auto
 speed auto
 !
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 !
!
interface FastEthernet0/1/0
 switchport trunk native vlan 12
 switchport mode trunk
 switchport voice vlan 11
 spanning-tree portfast
 !
!         
interface FastEthernet0/1/1
 switchport trunk native vlan 12
 switchport mode trunk
 switchport voice vlan 11
 spanning-tree portfast
 !        
!         
interface FastEthernet0/1/2
 switchport trunk native vlan 12
 switchport mode trunk
 switchport voice vlan 11
 spanning-tree portfast
 !        
!         
interface FastEthernet0/1/3
 switchport trunk native vlan 12
 switchport mode trunk
 switchport voice vlan 11
 spanning-tree portfast
 !        
!         
interface Serial0/0/0
 no ip address
 encapsulation frame-relay
 no frame-relay inverse-arp
 !        
!         
interface Serial0/0/0.101 point-to-point
 ip address 177.0.101.2 255.255.255.0
 frame-relay interface-dlci 602   
!         
interface Serial0/0/1
 no ip address
 shutdown 
 clock rate 200
 !        
!         
interface Serial0/2/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing ie facility 
 isdn outgoing display-ie
 no cdp enable
 !        
!         
interface Vlan1
 no ip address
 !        
!         
interface Vlan11
 ip address 177.2.11.1 255.255.255.0
 ip helper-address 10.11.11.19
 !        
!         
interface Vlan12
 ip address 177.2.12.1 255.255.255.0
 !        
!         
!