Re: [OSL | CCIE_Voice] GK failover call display

[ShinGei Yong]

What Alex describe is one of the reason that the "+" sign doesn't appear due
to the H323 native behavior.

Another reason is the GW ios version and the phone load. You may refer to
the below link on why proctor lab PSTN can display the "+" sign.
http://onlinestudylist.com/archives/ccie_voice/2010-March/066917.html

For your CME question, do a debug on isdn and ccapi inout, to confirm that
the number send by PSTN.
If the "+" doesn't tag along with the ANI, then you probably need to do what
Alex has mentioned earlier.

TIA
Shingei

On Wed, Jun 29, 2011 at 8:20 AM, Adil Shaikh  wrote:

> Hi all,
>
>
> I have configured route list with 1st choice as gatekeeper and 2nd choice
> as local PSTN.
> When I shut down the Gatekeeper, the call goes out from PSTN and back into
> branch gateway via PSTN as expected.
>
> debug isdn q931 shows the 'Calling Party Number' in +E164 format but the
> phone display calling party number without plus. The phone is 7965.
>
> Is this what you are getting on your phone? Is this normal behaviour?
>
> One branch site is H323 and other is CME.
>
>
> Thanks
> -adil
>
>
> --
>   .. . .
> _7___|___|_|_|adil.sha...@gmail.com
>
> . .
>
>
>
> ___
> 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] MVA hairpin @ BR1, call drop if call from HQ or PSTN

[ShinGei Yong]

hi Adil,

Apologize for the questions, because never try your setup before.

How do you translate the mgcp incoming 1999 to 1998? Do you have any
translation?
If not, how are you going to match the DNIS#

My understanding was:

PSTN IN --> mgcp gw --> route pattern --> h323 gw --> match incoming mva#
(service mva configure) dp-->match outgoing mva# dp --> UCM

TIA
Shingei

On Wed, Jun 15, 2011 at 7:32 PM, Adil Shaikh  wrote:

> hi ShinGei,
>
> The MVA number is 1998.
> Rather than using same number for the inbound to application service mva
> and outbound to MVA number, two seperate numbers are used.
>
> So, inbound dialpeer to application service mva is 1999 and after
> hairpinning on Gateway the outbound number is 1998 which is the Mobile Voice
> Access number configured on the Call Manager.
>
> i hope this clarifies what i am trying to do.
>
> thanks
> -adil
>
> On Wed, Jun 15, 2011 at 9:27 PM, ShinGei Yong wrote:
>
>> Hi,
>>
>> May i know where is your outbound dp for mva#1999?
>> What is the purpose of 1998 ?
>>
>> TIA
>> Shingei
>>
>>
>> On Wed, Jun 15, 2011 at 2:26 PM, Adil Shaikh wrote:
>>
>>> Forgot to add that HQ and BR1 phones are sccp.
>>>
>>>
>>> On Wed, Jun 15, 2011 at 4:21 PM, Adil Shaikh wrote:
>>>
>>>> Hi all,
>>>>
>>>> I have configured MVA hairpin on BR1 router which is an MGCP gateway.
>>>> If the call is made to MVA route-pattern number [1999] from BR1 phone
>>>> then i hear "Welcome to Cisco Unfiied Communications"
>>>> If i call to MVA route-pattern number [1999] from HQ phone or PSTN phone
>>>> then i do not hear "Welcome to Cisco Unfied Communications". debug shows 
>>>> the
>>>> correct MVA dial-peer hits on BR1-GW.
>>>>
>>>> There is no transcoder configured on HQ or BR1 device pool.  To resolve
>>>> my issue, if transcoder needs to be configured on HQ or BR1 device pool 
>>>> then
>>>> i would like to know why.
>>>>
>>>>
>>>> Here is the configuration:
>>>> MVA directory number: 1998
>>>> Route-pattern: 1999  > BR1-H323-GW in BR1 device pool
>>>> BR1-MGCP-GW in BR1 devicepool
>>>> BR1 and HQ region talks g729
>>>>
>>>> Configuration from BR1-RTR:
>>>>
>>>> interface Loopback0
>>>>  ip address 10.10.110.2 255.255.255.255
>>>>  ip ospf network point-to-point
>>>>  h323-gateway voip bind srcaddr 10.10.110.2
>>>>
>>>>
>>>> voice service voip
>>>>  allow-connections h323 to h323
>>>> !
>>>> !
>>>> !
>>>> voice class codec 1
>>>>  codec preference 1 g711ulaw bytes 160
>>>>  codec preference 2 g729r8 bytes 20
>>>> !
>>>> !
>>>> !
>>>> voice class h323 1
>>>>  h225 timeout tcp establish 3
>>>> !
>>>>
>>>>
>>>> voice translation-rule 1002
>>>>  rule 1 /^1002$/ /4158884343/
>>>> !
>>>> !
>>>> voice translation-profile 1002
>>>>  translate calling 1002
>>>> !
>>>> !
>>>> !
>>>> application
>>>>   service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
>>>>   !
>>>>
>>>> dial-peer voice 1997 voip
>>>>  destination-pattern 1998
>>>>  voice-class codec 1
>>>>  voice-class h323 1
>>>>  session target ipv4:10.10.210.11
>>>>  dtmf-relay h245-alphanumeric
>>>>  no vad
>>>> !
>>>> dial-peer voice 1998 voip
>>>>  preference 1
>>>>  destination-pattern 1998
>>>>  voice-class codec 1
>>>>  voice-class h323 1
>>>>  session target ipv4:10.10.210.10
>>>>  dtmf-relay h245-alphanumeric
>>>>  no vad
>>>> !
>>>> dial-peer voice 1999 voip
>>>>  translation-profile incoming 1002
>>>>  service mva
>>>>  voice-class codec 1
>>>>  incoming called-number 1999
>>>>  dtmf-relay h245-alphanumeric
>>>> !
>>>> -
>>>> output of debug voip dial-p
>>>>
>>>> Following output when called 1999 from BR1 extenstion 1002. I hear the
>>>> MVA message:
>>>> R1-RTR#
>>>> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPee

Re: [OSL | CCIE_Voice] MVA hairpin @ BR1, call drop if call from HQ or PSTN

[ShinGei Yong]

Hi,

May i know where is your outbound dp for mva#1999?
What is the purpose of 1998 ?

TIA
Shingei


On Wed, Jun 15, 2011 at 2:26 PM, Adil Shaikh  wrote:

> Forgot to add that HQ and BR1 phones are sccp.
>
>
> On Wed, Jun 15, 2011 at 4:21 PM, Adil Shaikh wrote:
>
>> Hi all,
>>
>> I have configured MVA hairpin on BR1 router which is an MGCP gateway.
>> If the call is made to MVA route-pattern number [1999] from BR1 phone then
>> i hear "Welcome to Cisco Unfiied Communications"
>> If i call to MVA route-pattern number [1999] from HQ phone or PSTN phone
>> then i do not hear "Welcome to Cisco Unfied Communications". debug shows the
>> correct MVA dial-peer hits on BR1-GW.
>>
>> There is no transcoder configured on HQ or BR1 device pool.  To resolve my
>> issue, if transcoder needs to be configured on HQ or BR1 device pool then i
>> would like to know why.
>>
>>
>> Here is the configuration:
>> MVA directory number: 1998
>> Route-pattern: 1999  > BR1-H323-GW in BR1 device pool
>> BR1-MGCP-GW in BR1 devicepool
>> BR1 and HQ region talks g729
>>
>> Configuration from BR1-RTR:
>>
>> interface Loopback0
>>  ip address 10.10.110.2 255.255.255.255
>>  ip ospf network point-to-point
>>  h323-gateway voip bind srcaddr 10.10.110.2
>>
>>
>> voice service voip
>>  allow-connections h323 to h323
>> !
>> !
>> !
>> voice class codec 1
>>  codec preference 1 g711ulaw bytes 160
>>  codec preference 2 g729r8 bytes 20
>> !
>> !
>> !
>> voice class h323 1
>>  h225 timeout tcp establish 3
>> !
>>
>>
>> voice translation-rule 1002
>>  rule 1 /^1002$/ /4158884343/
>> !
>> !
>> voice translation-profile 1002
>>  translate calling 1002
>> !
>> !
>> !
>> application
>>   service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
>>   !
>>
>> dial-peer voice 1997 voip
>>  destination-pattern 1998
>>  voice-class codec 1
>>  voice-class h323 1
>>  session target ipv4:10.10.210.11
>>  dtmf-relay h245-alphanumeric
>>  no vad
>> !
>> dial-peer voice 1998 voip
>>  preference 1
>>  destination-pattern 1998
>>  voice-class codec 1
>>  voice-class h323 1
>>  session target ipv4:10.10.210.10
>>  dtmf-relay h245-alphanumeric
>>  no vad
>> !
>> dial-peer voice 1999 voip
>>  translation-profile incoming 1002
>>  service mva
>>  voice-class codec 1
>>  incoming called-number 1999
>>  dtmf-relay h245-alphanumeric
>> !
>> -
>> output of debug voip dial-p
>>
>> Following output when called 1999 from BR1 extenstion 1002. I hear the MVA
>> message:
>> R1-RTR#
>> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
>> Type=PEER_TYPE_VOICE,
>>Peer Info Type=DIALPEER_INFO_SPEECH
>> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
>> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
>> Type=PEER_TYPE_VOICE,
>>Peer Info Type=DIALPEER_INFO_SPEECH
>> BR1-RTR#
>> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
>> BR1-RTR#
>> BR1-RTR#
>> BR1-RTR#
>> BR1-RTR#
>> BR1-RTR#
>> Following output when called 1999 from HQ extenstion 5002. I do not hear
>> the MVA message:
>> BR1-RTR#
>> BR1-RTR#
>> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>>Calling Number=5002, Called Number=1999, Voice-Interface=0x0,
>>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
>> Type=PEER_TYPE_VOICE,
>>Peer Info Type=DIALPEER_INFO_SPEECH
>> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
>> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>>Calling Number=5002, Called Number=1999, Voice-Interface=0x0,
>>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
>> Type=PEER_TYPE_VOICE,
>>Peer Info Type=DIALPEER_INFO_SPEECH
>> BR1-RTR#
>> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
>> BR1-RTR#
>> BR1-RTR#
>> BR1-RTR#
>>
>> Following output when called 4158881999 from PSTN phone 4158884343. I do
>> not hear the MVA message:
>> BR1-RTR#
>> *Jun 15 11:25:44.078: //-1/808792E10D00/DPM/dpAssociateIncomingPeerCore:
>>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
>> Type=PEER_TYPE_VOICE,
>>Peer Info Type=DIALPEER_INFO_SPEECH
>> *Jun 15 11:25:44.078: //-1/808792E10D00/DPM/dpAssociateIncomingPeerCore:
>>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
>> *Jun 15 11:25:44.078: /

Re: [OSL | CCIE_Voice] CME background with only 2 files

[ShinGei Yong]

You better to be familiar on how to write the List.xml script without
referring to any doc during ur attempt.

Not that too difficult, get yourself typing the below stuff 10times per day
till the exam.





Shingei.

On Wed, Jun 8, 2011 at 10:22 AM, Chris Green  wrote:

> Hi All,
>
> For CME Background we need 3 files, 2 images and one List XML file.
>
> Is there any way to complete this task with only with "2 images" and not
> "List XML" file?
>
> Basically what would be the solution if you provided with only 2 images and
> NOT List XML file.
>
> Chris
>
> ___
> 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] IPPM on cisco7961 didn't alert

[ShinGei Yong]

Ya, i believe assigned the "standard CTI control of all devices" + "standard
CTI enabled" to phone messenger apps user will do the trick.

For phone user,a combination of "standard CCM end user" + "standard cti
enabled" should be sufficient.

Shingei.

On Wed, Jun 1, 2011 at 6:11 AM, Ki Wi  wrote:

> Finally found the problem, I didn't give enough right to IPPM user. Seems
> like it need CCM Super user right instead!
>
>
> On Wed, Jun 1, 2011 at 6:07 AM, Ki Wi  wrote:
>
>> For me, everything is fine. I just do everything again from fresh. Same
>> problem.
>>
>> Something must be missing along the way.
>>
>> Is there any IPPM guide online? The one i found from
>> http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_Configuring_Cisco_IP_Phone_Messenger_on_Cisco_Unified_Presenceis
>>  pretty useless.
>>
>> On Wed, Jun 1, 2011 at 1:21 AM, Alex Goh  wrote:
>>
>>> Hi Ki Wi,
>>>
>>> I've encounter the same issue also, and I solved it by changing the
>>> Enterprise Parameters Services URL to IP instead of hostname (Apparently, I
>>> miss that part when I reverted my VMware snapshot), remember I saw this
>>> solution from OSL discussion before.
>>>
>>> HTH
>>>
>>> Cheers,
>>> Alex
>>>
>>> On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong 
>>> wrote:
>>>
>>>> Frens,
>>>>
>>>> If i can recall correctly, that was due to that i missed associate the
>>>> phone with
>>>> application user "Phone Messenger". You need the phone messenger
>>>> application user
>>>> to control the IPPM user.
>>>> Without the association, the messaging will still work but funny stuff
>>>> come out, if not wrong
>>>>
>>>> Shingei.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, May 31, 2011 at 4:52 AM, Ki Wi  wrote:
>>>>
>>>>> Hey,
>>>>> Do you still remember how did you resolve this alert issue? I'm still
>>>>> trying to train myself up in CUPS. Last night, my alert was working, my 
>>>>> IPPM
>>>>> login wasn't. Today my IPPM is working but no alert. =( All other 
>>>>> components
>>>>> are working.
>>>>>
>>>>>
>>>>> On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong >>>> > wrote:
>>>>>
>>>>>> Guys,
>>>>>> Pls ignore this mail, has managed to figured out the caused.
>>>>>>
>>>>>> thanks
>>>>>> Shingei.
>>>>>>
>>>>>>
>>>>>> On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong >>>>> > wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>> I've configure the IPPM on cisco 7961 phone,
>>>>>>> everything works smooth other that the message receive alert.
>>>>>>> It doesn't "ring" when there is a mgs come in from CIPC or
>>>>>>> other IPPM.i've set the "audible alert" to ON but still got
>>>>>>> no luck.
>>>>>>>
>>>>>>> Another IPPM phone encounter the same issue, so don't think
>>>>>>> is the phone problem. Any idea?
>>>>>>>
>>>>>>>
>>>>>>> Thanks
>>>>>>> Shingei.
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ___
>>>>>> For more information regarding industry leading CCIE Lab training,
>>>>>> please visit www.ipexpert.com
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
>>>> please visit www.ipexpert.com
>>>>
>>>> Are you a CCNP or CCIE and looking for a job? Check out
>>>> www.PlatinumPlacement.com
>>>>
>>>
>>>
>>
>
___
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Re: [OSL | CCIE_Voice] IPPM on cisco7961 didn't alert

[ShinGei Yong]

Frens,

If i can recall correctly, that was due to that i missed associate the phone
with
application user "Phone Messenger". You need the phone messenger application
user
to control the IPPM user.
Without the association, the messaging will still work but funny stuff come
out, if not wrong

Shingei.




On Tue, May 31, 2011 at 4:52 AM, Ki Wi  wrote:

> Hey,
> Do you still remember how did you resolve this alert issue? I'm still
> trying to train myself up in CUPS. Last night, my alert was working, my IPPM
> login wasn't. Today my IPPM is working but no alert. =( All other components
> are working.
>
>
> On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong wrote:
>
>> Guys,
>> Pls ignore this mail, has managed to figured out the caused.
>>
>> thanks
>> Shingei.
>>
>>
>> On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong wrote:
>>
>>> Hi,
>>> I've configure the IPPM on cisco 7961 phone,
>>> everything works smooth other that the message receive alert.
>>> It doesn't "ring" when there is a mgs come in from CIPC or
>>> other IPPM.i've set the "audible alert" to ON but still got
>>> no luck.
>>>
>>> Another IPPM phone encounter the same issue, so don't think
>>> is the phone problem. Any idea?
>>>
>>>
>>> Thanks
>>> Shingei.
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
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Re: [OSL | CCIE_Voice] CME presence - BLF speed dial vs. monitor button

[ShinGei Yong]

Hi All,

Adom provided a useful link indicated that BLF SD didn't work with DND prior
to CME 7.1
"In versions earlier than Cisco Unified CME 7.1, BLF monitoring does not
provide notification
of status changes when a monitored directory number becomes DND-enabled."

What i tested today was, i've 2 sccp phones, phone 1 with ext 3001 while
phone 2 with ext 3002
Phone 2 configured with blf-speed-dial to monitor line 1 of phone 1,3001.

When DND disable on phone 1, phone 2 can monitor the status of  3001 via
BLF, as below:

  Watcher   : 3002@142.102.66.50
  Presentity: 3001@142.102.66.254
  Expires   : 3600 seconds
  Subscription Duration : 2483 seconds
  line status   : idle
  watcher type  : local
  presentity type   : local
 Watcher phone type: SCCP [BLF Speed Dial]

phone 1 offhook, as below:
  Watcher   : 3002@142.102.66.50
  Presentity: 3001@142.102.66.254
  Expires   : 3600 seconds
  Subscription Duration : 2469 seconds
  line status   : busy
  watcher type  : local
  presentity type   : local
Watcher phone type: SCCP [BLF Speed Dial]

But when DND is enabled on phone 1, phone 2 was still received line status
update via BLF.
UCME is running on version
=
Version 7.0(1)
Cisco Unified Communications Manager Express

I've reloaded both the phones and CME.
Any reason why it works? Or i've mis-configured?

Shingei



On Thu, May 19, 2011 at 2:40 AM, Nirvair Sahota <
nirvair.sah...@sbcglobal.net> wrote:

> Thanks for clarification, BLF Speed Dial did not work for me with DND.
>
> --- On *Wed, 5/18/11, Adam Thompson * wrote:
>
>
> From: Adam Thompson 
>
> Subject: Re: [OSL | CCIE_Voice] CME presence - BLF speed dial vs. monitor
> button
> To: ccie_voice@onlinestudylist.com
> Date: Wednesday, May 18, 2011, 11:10 AM
>
>
>  BLF speed dials do not work with DND in CUCME 7.0. However, It does work
> with DND in CUCME 7.1. So if you get something like this in the lab, using
> only a BLF speed dial will not work. If you need DND status in CUCME 7.0,
> you need to use 'watch'.
>
> source:
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmepres.html#wp1022428
>
> HTH
> -Adam
> ccie #28928 (voice)
>
> On Wed, May 18, 2011 at 1:27 PM, adam compton 
> http://us.mc829.mail.yahoo.com/mc/compose?to=com...@gmail.com>
> > wrote:
>
> I think BLF speed dial can monitor a phone at a different location.
> Monitor can't.  You still have to program Presence on the CME gateway to get
> that to work though.
>
>   On Wed, May 18, 2011 at 11:30 AM, Michael Luo 
> http://us.mc829.mail.yahoo.com/mc/compose?to=hout...@gmail.com>
> > wrote:
>
>  On CME phone, we have different ways to monitor another phone's status
> (Ringing, Offhook, DND).
>
> I was trying to understand the difference between BLF speed dial and the
> monitor button (e.g. "button 3m1").  They seem to have the same feature.
> What are the difference?
>
> Thanks!
> Michael
>
> ___
> 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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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com 
>
>
>
> -Inline Attachment Follows-
>
>
> ___
> 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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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] DND on UCME SIP Phone

[ShinGei Yong]

Guys,

Facing problem on sip phone DND on UCME.
Has configured dnd on individual pool,has enabled "dnd-control" under ephone
template,
but DND feature still unavailable through Device Configuration >> Call
Preference >> DND
and of course DND softkey not appear.

Mac address is 001E.F728.2C0C
  Type is 7961
  Number list 1 : DN 1
  Proxy Ip address is 0.0.0.0
  DTMF Relay is enabled, rtp-nte, sip-notify
  Call Waiting is enabled
  DnD is enabled
  Description is 32143001
  keep-conference is enabled
  username 3001 password 1234
  template is 1
  kpml signal is enabled
  blf call list is enabled

Phone firmware is running 8.3.3, ios ver 12.4(22)T4

Any idea?

TIA
Shingei
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Re: [OSL | CCIE_Voice] MGCP BR1 registration problem

[ShinGei Yong]

My Fren,

Your PSTN rtr, the isdn port that connected to your BR1 site, "isdn
emulate-network" are missing from
serial interface 0/1/0:23.

regards
Shingei

On Mon, May 2, 2011 at 9:04 AM, Ahmed Ellboudy
wrote:

> Still did not solve it .
>
> Any idea?
>
>
>
> Thanks,
>
>
>
>
>
> Ahmed Ellboudy | CCNP, CCVP.
>
> * *
>
> *Networking Team Leader*
>
> Raya IT - Professional Networking Services
>
> Mobile: +20100770837
>
> Tel  : +20238276000 Ext. 2338
>
> Fax : +20238372930
>
> Email  : ahmed_ellbo...@rayacorp.com 
> Address : El Motamayez District - 6th of October
>
> *[image: cid:image001.jpg@01CB8A26.89E6B660]*
>
>
>
> *From:* CCIE Voice [mailto:cc...@corb.net]
> *Sent:* Monday, May 02, 2011 2:34 AM
> *To:* Ahmed Ellboudy
> *Cc:* 
> *Subject:* Re: [OSL | CCIE_Voice] MGCP BR1 registeration problem
>
>
>
> Try reloading the router.
>
> --
>
>
>
>
> On May 1, 2011, at 17:43, "Ahmed Ellboudy" 
> wrote:
>
> Dear All ,
>
> I am facing a problem to register br1 as MGCP on the CUCM
>
> The problem is when I use the T1 as H323 is working normally but when I
> need to use it as MGCP there is a problem in q921
>
> As L2 be TEI Assigned forever .
>
>
>
> I tried no isdn bind-l3 ccm
>
> The output of the debug isdn q921 is below :
>
> *May  1 21:09:19.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:19.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:25.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:25.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:26.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:26.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:27.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:27.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:28.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:28.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:34.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:34.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:35.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:35.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:36.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:36.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
> *May  1 21:09:37.431: ISDN Se0/1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>
> *May  1 21:09:37.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED &
> vsc_wants_L2_up = FALSE
>
>
>
> Please find attached the PSTN ,BR1 configuration and image of the call
> manager for this GW.
>
>
>
> Can anyone help me to fulfill this problem?
>
>
>
>
>
> Thanks,
>
>
>
>
>
> Ahmed Ellboudy | CCNP, CCVP.
>
> * *
>
>
>
> Disclaimer: NOTICE The information contained in this message is
> confidential and is intended for the addressee(s) only. If you have received
> this message in error or there are any problems please notify the originator
> immediately. The unauthorized use, disclosure, copying or alteration of this
> message is strictly forbidden. Raya will not be liable for direct, special,
> indirect or consequential damages arising from alteration of the contents of
> this message by a third party or as a result of any malicious code or virus
> being passed on. Views expressed in this communication are not necessarily
> those of Raya.If you have received this message in error, please notify the
> sender immediately by email, facsimile or telephone and return and/or
> destroy the original message.
>
> 
>
> 
>
> 
>
> ___
> 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
>
>
> Disclaimer: NOTICE The information contained in this message is
> confidential and is intended for the addressee(s) only. If you have received
> this message in error or there are any problems please notify the originator
> immediately. The unauthorized use, disclosure, copying or alteration of this
> message is strictly forbidden. Raya will not be liable for direct, special,
> indirect or consequential damages arising from alteration of the contents of
> this message by a third party or as a result of any malicious code or virus
> being passed on. Views expressed in this communication are not necessarily
> those of Raya.If you have received this message in error, please notify the
> sender immediately by email, facsimile or telephone and return and/or
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>
> ___
> For more informa

Re: [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST

[ShinGei Yong]

Hi Roger,

That's what i'm wondering too,but the fact is that, this is what the
question request.
The PGuide doesn't given too much of explanation of this.

Anyone who completed this question may provide some hints?

Thanks
Shingei.


On Mon, Apr 11, 2011 at 2:26 PM, Rogers Ochieng wrote:

> That's why i had wondered what the purpose is. I don't see a way unless you
> want to match all PSTN except HQ/BR1 numbers then translate those to go to
> VM and let thw HQ/BR1 go through the normal CFB and CFNA
>
>
> On 11 April 2011 09:16, ShinGei Yong  wrote:
>
>> Hi Roger,
>>
>> So what you trying to do is that
>> PSTN Caller --> ephone-dn(x3006) --> cfa vm --> outgoing xlation-rule
>>
>> Why do we need to change redirecting-number? Since its already 3006.
>> Also,if perform cfa, what about caller from HQ/BR1?
>> As BR2 is under SRST, HQ/BR1 will reach BR2 via PSTN as well, this will
>> cause them to be
>> directed to vm directly also right?
>>
>> Thanks
>> Shingei
>>
>>
>>
>>
>> On Mon, Apr 11, 2011 at 1:30 PM, Rogers Ochieng 
>> wrote:
>>
>>> Off the top of my head am thinking do call forward all and apply outgoing
>>> voice translation rule under the ephone-dn to change your redirecting
>>> number.
>>>
>>> On 11 April 2011 07:47, ShinGei Yong  wrote:
>>>
>>>> Hi all,
>>>>
>>>> I'm following up the same question that i posted last month, regarding
>>>> the PSTN call diversion
>>>> to BR2 CUE Voicemail during SRST.I was got confuse by the question,but
>>>> looking at it to question & PGuide,
>>>> i think my understanding is correct.
>>>>
>>>> The question is asking,
>>>> Ensure that the caller from PSTN who have dialed BR2 phone can be routed
>>>> to Voicemail when
>>>> there is a WAN outage at the BR2 site.
>>>>
>>>> In PGuide, the BR2 DN has CFUR external checked.So my understanding is
>>>> that,when PSTN caller
>>>> call BR2 phone,it should be routed to voicemail immediately,which CUE is
>>>> locally located.
>>>> My question is,how do achieve that PSTN caller enter Voicemail
>>>> immediately without ringing the BR2 phone but
>>>> allowing call from HQ/BR1 ring the destination and enter voicemail when
>>>> CFNA or CFB.
>>>>
>>>> Anyone complete this section successfully?May share the idea?
>>>>
>>>>
>>>> Shingei
>>>>
>>>>
>>>>
>>>> On Tue, Mar 8, 2011 at 11:16 PM, Rogers Ochieng <
>>>> rogersochi...@gmail.com> wrote:
>>>>
>>>>> Looking at that question the wording there does not specify that you
>>>>> need to send calls immediately to voicemail but that PSTN calls to BR2 can
>>>>> be routed to VM, it doesn't say at what state so to send VM so PSTN calls 
>>>>> to
>>>>> BR2 busy and no answer states should meet the requirement.
>>>>>
>>>>>
>>>>> On 8 March 2011 18:05, ShinGei Yong  wrote:
>>>>>
>>>>>> Hi Rogers,
>>>>>>
>>>>>> Yes, and again, as stated below, i'm able to achieved CFB and CFNA
>>>>>> during SRST,
>>>>>> so in other words,the required dial-peer and setting to route call to
>>>>>> CUE is already done,right?
>>>>>> And HQ/BR1 phone is able to call BR2 phone via PSTN during SRST.
>>>>>>
>>>>>> Maybe put the question in this way:
>>>>>>
>>>>>> PSTN caller which originate from HQ/BR1, ring the destination during
>>>>>> SRST.(I've done this)
>>>>>> PSTN caller which originate from PSTN, to Voicemail without ringing
>>>>>> BR2 phones during SRST
>>>>>>
>>>>>> Is the second requirement possible?
>>>>>>
>>>>>> Shingei.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 8, 2011 at 10:26 PM, Rogers Ochieng <
>>>>>> rogersochi...@gmail.com> wrote:
>>>>>>
>>>>>>> AS you've stated you are using CUE which in normal operations you've
>>>>>>> integrated using jtapi CUE integration, i assuem the CUE module is on 
>>>>>>> the
>>>>>>>

Re: [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST

[ShinGei Yong]

Hi Roger,

So what you trying to do is that
PSTN Caller --> ephone-dn(x3006) --> cfa vm --> outgoing xlation-rule

Why do we need to change redirecting-number? Since its already 3006.
Also,if perform cfa, what about caller from HQ/BR1?
As BR2 is under SRST, HQ/BR1 will reach BR2 via PSTN as well, this will
cause them to be
directed to vm directly also right?

Thanks
Shingei



On Mon, Apr 11, 2011 at 1:30 PM, Rogers Ochieng wrote:

> Off the top of my head am thinking do call forward all and apply outgoing
> voice translation rule under the ephone-dn to change your redirecting
> number.
>
> On 11 April 2011 07:47, ShinGei Yong  wrote:
>
>> Hi all,
>>
>> I'm following up the same question that i posted last month, regarding the
>> PSTN call diversion
>> to BR2 CUE Voicemail during SRST.I was got confuse by the question,but
>> looking at it to question & PGuide,
>> i think my understanding is correct.
>>
>> The question is asking,
>> Ensure that the caller from PSTN who have dialed BR2 phone can be routed
>> to Voicemail when
>> there is a WAN outage at the BR2 site.
>>
>> In PGuide, the BR2 DN has CFUR external checked.So my understanding is
>> that,when PSTN caller
>> call BR2 phone,it should be routed to voicemail immediately,which CUE is
>> locally located.
>> My question is,how do achieve that PSTN caller enter Voicemail immediately
>> without ringing the BR2 phone but
>> allowing call from HQ/BR1 ring the destination and enter voicemail when
>> CFNA or CFB.
>>
>> Anyone complete this section successfully?May share the idea?
>>
>>
>> Shingei
>>
>>
>>
>> On Tue, Mar 8, 2011 at 11:16 PM, Rogers Ochieng 
>> wrote:
>>
>>> Looking at that question the wording there does not specify that you need
>>> to send calls immediately to voicemail but that PSTN calls to BR2 can be
>>> routed to VM, it doesn't say at what state so to send VM so PSTN calls to
>>> BR2 busy and no answer states should meet the requirement.
>>>
>>>
>>> On 8 March 2011 18:05, ShinGei Yong  wrote:
>>>
>>>> Hi Rogers,
>>>>
>>>> Yes, and again, as stated below, i'm able to achieved CFB and CFNA
>>>> during SRST,
>>>> so in other words,the required dial-peer and setting to route call to
>>>> CUE is already done,right?
>>>> And HQ/BR1 phone is able to call BR2 phone via PSTN during SRST.
>>>>
>>>> Maybe put the question in this way:
>>>>
>>>> PSTN caller which originate from HQ/BR1, ring the destination during
>>>> SRST.(I've done this)
>>>> PSTN caller which originate from PSTN, to Voicemail without ringing BR2
>>>> phones during SRST
>>>>
>>>> Is the second requirement possible?
>>>>
>>>> Shingei.
>>>>
>>>>
>>>>
>>>> On Tue, Mar 8, 2011 at 10:26 PM, Rogers Ochieng <
>>>> rogersochi...@gmail.com> wrote:
>>>>
>>>>> AS you've stated you are using CUE which in normal operations you've
>>>>> integrated using jtapi CUE integration, i assuem the CUE module is on the
>>>>> BR2 router. So for SRST create a voip dial-peer using sip protocol and 
>>>>> codec
>>>>> g711ulaw, dtmf sip-notify, to route calls to CUE and set CFB and CFNA, 
>>>>> setup
>>>>> CUE sip settings. For HQ and BR1 to access BR2 setup CFUR
>>>>>
>>>>> On 8 March 2011 16:05, ShinGei Yong  wrote:
>>>>>
>>>>>> Hi Roger,
>>>>>>
>>>>>> As stated below, i'm able to achieved that PSTN caller routed to VM
>>>>>> when CFB and CFNA.
>>>>>> Alsothere's no CUC in this lab.
>>>>>>
>>>>>> how to achieve that PSTN caller will be route to VM while allowing HQ
>>>>>> or BR1 ring the destination in SRST site?
>>>>>> TIA
>>>>>> Shingei
>>>>>>
>>>>>> 2011/3/8 Roger Källberg 
>>>>>>
>>>>>>   You need to setup CFB & CFNA in an SRST situation, so that it sends
>>>>>>> the call over PSTN to CUC VM.
>>>>>>>
>>>>>>> Sincerely
>>>>>>>
>>>>>>>  *Roger Källberg*
>>>>>>> CCIE #26199 (Voice)
>>>>>>> Consultant
>>>>>>> Cygate AB
>&

Re: [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST

[ShinGei Yong]

Hi all,

I'm following up the same question that i posted last month, regarding the
PSTN call diversion
to BR2 CUE Voicemail during SRST.I was got confuse by the question,but
looking at it to question & PGuide,
i think my understanding is correct.

The question is asking,
Ensure that the caller from PSTN who have dialed BR2 phone can be routed to
Voicemail when
there is a WAN outage at the BR2 site.

In PGuide, the BR2 DN has CFUR external checked.So my understanding is
that,when PSTN caller
call BR2 phone,it should be routed to voicemail immediately,which CUE is
locally located.
My question is,how do achieve that PSTN caller enter Voicemail immediately
without ringing the BR2 phone but
allowing call from HQ/BR1 ring the destination and enter voicemail when CFNA
or CFB.

Anyone complete this section successfully?May share the idea?


Shingei


On Tue, Mar 8, 2011 at 11:16 PM, Rogers Ochieng wrote:

> Looking at that question the wording there does not specify that you need
> to send calls immediately to voicemail but that PSTN calls to BR2 can be
> routed to VM, it doesn't say at what state so to send VM so PSTN calls to
> BR2 busy and no answer states should meet the requirement.
>
>
> On 8 March 2011 18:05, ShinGei Yong  wrote:
>
>> Hi Rogers,
>>
>> Yes, and again, as stated below, i'm able to achieved CFB and CFNA during
>> SRST,
>> so in other words,the required dial-peer and setting to route call to CUE
>> is already done,right?
>> And HQ/BR1 phone is able to call BR2 phone via PSTN during SRST.
>>
>> Maybe put the question in this way:
>>
>> PSTN caller which originate from HQ/BR1, ring the destination during
>> SRST.(I've done this)
>> PSTN caller which originate from PSTN, to Voicemail without ringing BR2
>> phones during SRST
>>
>> Is the second requirement possible?
>>
>> Shingei.
>>
>>
>>
>> On Tue, Mar 8, 2011 at 10:26 PM, Rogers Ochieng 
>> wrote:
>>
>>> AS you've stated you are using CUE which in normal operations you've
>>> integrated using jtapi CUE integration, i assuem the CUE module is on the
>>> BR2 router. So for SRST create a voip dial-peer using sip protocol and codec
>>> g711ulaw, dtmf sip-notify, to route calls to CUE and set CFB and CFNA, setup
>>> CUE sip settings. For HQ and BR1 to access BR2 setup CFUR
>>>
>>> On 8 March 2011 16:05, ShinGei Yong  wrote:
>>>
>>>> Hi Roger,
>>>>
>>>> As stated below, i'm able to achieved that PSTN caller routed to VM when
>>>> CFB and CFNA.
>>>> Alsothere's no CUC in this lab.
>>>>
>>>> how to achieve that PSTN caller will be route to VM while allowing HQ or
>>>> BR1 ring the destination in SRST site?
>>>> TIA
>>>> Shingei
>>>>
>>>> 2011/3/8 Roger Källberg 
>>>>
>>>>   You need to setup CFB & CFNA in an SRST situation, so that it sends
>>>>> the call over PSTN to CUC VM.
>>>>>
>>>>> Sincerely
>>>>>
>>>>>  *Roger Källberg*
>>>>> CCIE #26199 (Voice)
>>>>> Consultant
>>>>> Cygate AB
>>>>> Eric Perssons väg 21, SE-217 62 MALMÖ
>>>>>
>>>>>  --
>>>>> *Från:* ShinGei Yong [shingei.y...@gmail.com]
>>>>> *Skickat:* den 8 mars 2011 11:00
>>>>> *Till:* ccie_voice@onlinestudylist.com
>>>>> *Ämne:* [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST
>>>>>
>>>>>  Hi,
>>>>>
>>>>> The question stated,caller from PSTN "CAN BE" routed to VM when there's
>>>>> WAN outage at BR2.
>>>>> Internal caller from HQ or BR1 must be able to reach BR2 phone and
>>>>> forward to VM if no answer.
>>>>>
>>>>> To me,there are two meaning of the sentense
>>>>>
>>>>> 1. PSTN caller routed to VM immediately when there's WAN outage at BR2,
>>>>>
>>>>>
>>>>> 2. PSTN caller routed to VM when CFB or CFNA.
>>>>>
>>>>> What confuse me is that,how to achieve that the PSTN caller routed to
>>>>> VM immediately
>>>>> when there's a WAN outage at BR2?I'm able to achieved that PSTN caller
>>>>> router to VM
>>>>> when CFB and CFNA.
>>>>>
>>>>> In proctor guide, Forward Unregisterd Int and External been
>>>>> checked(VM),but how the UCM instruct PSTN call
>>>>> to VM?The PSTN call will hitting the BR2 GW directly due to SRST.
>>>>>
>>>>> Am i thinking of too much?
>>>>>
>>>>> Shingei
>>>>>
>>>>
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
>>>> please visit www.ipexpert.com
>>>>
>>>>
>>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)

[ShinGei Yong]

Hi Alex@ncs,

While observing your config,i noticed that you've 3 zone defined under GK,
which are UCM,UCME& VIA.

If i remember correctly,ur R3 which is ur CME site should registered to UCME
instead of zone VIA right?
Also, what is your region configuration on that pointed to GK?

Thanks
Shingei.

On Sun, Apr 10, 2011 at 1:42 AM, Alex Goh  wrote:

> Hi Guys,
>
> I'm trying to get the solutions for question 4.2 to work, but
> apparently I'm missing something and hope someone can help.
> I've search thru the list but doesn't really found a solution work for my
> case.
>
> The issue I've encounter are when HQ phone 5001 calling BR2 phone
> 3003, 3003 ring, but when i tried to answered, the call drop.
> I know it might be related to codec issue, but I've my HQ-RTR
> configured with Xcoder which it is up and active but the call
> still failing. I also did have the trunk in cucm "Wait for Far End
> H.245 Terminal Capability Set" unchecked.
>
> once things I notice is that, my call doesn't seems get re-originated
> on the cube router to BR2 router, what I see during ringing state
> my "show gatekeeper endpoint" show the call is directly from the CUCM
> to BR2 It is only 2 call legs instead of 4 (see below).
>
> hm, what have I missed?
>
> Some Info:
> HQ Router (R1)
>
> interface Loopback0
>  ip address 172.1.254.1 255.255.255.255
>  h323-gateway voip interface
>  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
>  h323-gateway voip h323-id R1
>  h323-gateway voip bind srcaddr 172.1.254.1
>
> gatekeeper
>  zone local UCM 172.1.254.1
>  zone local UCME outvia VIA
>  zone local VIA
>  zone prefix UCME 3...
>  gw-type-prefix 1#* default-technology
>  no shutdown
>
> dial-peer voice 30 voip
>  destination-pattern 3...
>  session target ras
>  codec g711ulaw
> !
> dial-peer voice 31 voip
>  incoming called-number 3...
>
> Total number of active calls = 1.
> GATEKEEPER CALL INFO
> 
> LocalCallIDAge(secs)   BW
> 511-32797  6   16(Kbps)
>  Endpt(s): Alias E.164Addr
>   src EP: gk_trunk_25001
>   CallSignalAddr  Port  RASSignalAddr   Port
>   172.1.10.20 38233 172.1.10.20 32795
>  Endpt(s): Alias E.164Addr
>   dst EP: R33003
>   CallSignalAddr  Port  RASSignalAddr   Port
>   172.3.254.1 1720  172.3.254.1 49395
>
>GATEKEEPER ENDPOINT REGISTRATION
>
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
> --- - --- - - -
> 172.1.10.10 47142 172.1.10.10 32838 UCM   VOIP-GW
>H323-ID: gk_trunk_1
>Voice Capacity Max.=  Avail.=  Current.= 0
> 172.1.10.20 38233 172.1.10.20 32795 UCM   VOIP-GW
>H323-ID: gk_trunk_2
>Voice Capacity Max.=  Avail.=  Current.= 0
> 172.1.254.1 1720  172.1.254.2 56974 VIA   H323-GW
>H323-ID: R1
>Voice Capacity Max.=  Avail.=  Current.= 0
> 172.3.254.1 1720  172.3.254.1 49395 VIA   H323-GW
>H323-ID: R3
>Voice Capacity Max.=  Avail.=  Current.= 0
> Total number of active registrations = 4
>
> R1(config-if)#do sh gatek gw
> GATEWAY TYPE PREFIX TABLE
> =
> Prefix: 1#*(Default gateway-technology)
>  Zone UCM master gateway list:
>172.1.10.20:38233 gk_trunk_2
>172.1.10.10:47142 gk_trunk_1
>  Zone VIA master gateway list:
>172.3.254.1:1720 R3
>172.1.254.2:1720 R1
>
> BR2 Router (R2)
>
> interface Loopback0
>  ip address 172.3.254.1 255.255.255.255
>  h323-gateway voip interface
>  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
>  h323-gateway voip h323-id R3
>  h323-gateway voip tech-prefix 1#
>  h323-gateway voip bind srcaddr 172.3.254.1
>
> dial-peer voice 10 voip
>  incoming called-number 3...
>  dtmf-relay rtp-nte
>  codec g711ulaw
> !
>
> CUCM Trunk
> the trunk was assign a separate DP with a region that using G729 when
> calling HQ and BR2.
>
>
>
> Regards,
> Alex
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] IPMA Manager does not get info window

[ShinGei Yong]

Hi,
Has anyone encounter this weird behavior, the IPMA manager icons
doesn't appear on the phone display via EM. The same phone is able to
display the icons if that ext is configured as "manager".
Phone firmware all downloaded from UCM 7.01,and model is 7961G,should not
have any compatibility issue.

In non working scenario,the Manager is configured on device profile (EM),and
login into another ip phone.
I can login the assistant via IPMA service.
>From the assistant,i couldn't see the manager online/appear.

Action taken:
1.Restarted TomCat service
2.Restarted IPMA && CTI services
2.Deleted and recreate user "manager"

After a few hours troubleshooting,i found an identical bugs ID from Cisco:

Bug 
iD:CSCdz39967
Title: IPMA manager on ExtMobility does not get the info window on
telecaster

*IPMA manager on ExtMobility do not get the info window on telecaster **
Symptom*: IPMA is not supported for a EM/Mobile Manager.
*Condition*: The EM/Mobile manager logs into the same device for the 2nd
time.
it works fine the first time.
*Workaround*: Restart the Cisco IPMA Service by restarting the Cisco Tomcat
service from Windows Admistrative tools - services menu.According to bug
toolkit, it has been fixed since earlier release of UCM.
Any setting i'm configure wrongly? I can make the info window work if not
thru EM.

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

Re: [OSL | CCIE_Voice] Volume 2 Lab 4 - Dial plan question

[ShinGei Yong]

Hi,

Another "0" is your outbound PSTN dialing access code.Something like ur HQ
and BR1, used 9 as a PSTN access code.

Make sense?

Thanks
Shingei

On Tue, Apr 5, 2011 at 2:12 AM, Randall Crumm  wrote:

> HI I was just looking over V2 L4 and have a question
>
>
>
> In 2 .3 it says LD calls in Spain should use a TP 00.[1-9]xxx and Intl
> 000.!
>
>
>
> Is this right? I think the LD call would use an access code 0 and the intl
> is 00.
>
>
>
> Why the extra 0
>
>
>
> Thanks,
>
>
>
> Best Regards,
>
>
>
> Randall Crumm
>
> Voice and Video Architect
>
> Global Networking & Telecom
>
> *[image: ccvp_voice_sm]*
>
>
>
> [image: logo]
>
>
>
> 1007 Gibraltar Drive
>
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>
>
>
> Direct +1 408.576.7344
>
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>
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[ShinGei Yong]

Hi Guys,

Glad to see you all here at this time while now is time for "dinner"for
asia,are u guys sleepless?
I fire up my home lab after received the suggestion from Roger. Testing out
now...

20mins later...

Hi Roger,
Your magic works! As you mentioned, i changed the mva# partition from
pt-phones to pt-snr-3002,
the internal call successful,but just don't understand why RDP CSS can't see
pt-phones via a translation profile,
and why the mva# should be part of pt-snr-3002 instead of pt-phones? The
xlation pattern did have a css-phones
which comprised of pt-phones

Any reasons? OR RDP CSS can't view wild card expression?[15]XXX?

ps:George, you may try it out by just change the partition of mva# to
pt-xlate-mva(in ur case).

Thanks
Shingei.




2011/3/24 George Goglidze 

> Hi Roger,
>
> I don't see that as a requirement nowhere... I will try it out though next
> time I do it.
>
> Anyway, just to let you know, that while internal calls didn't work, I had
> an RP to 90014158884343 on PT-XLATE-MVA and that did work!!! so I'm not sure
> I need MVA DN's PT in CSS-MVA.
>
> From what I understand, MVA DN's PT has to be in a CSS of h323 gateway,
>
> Please correct me if I'm wrong.
>
> As I said, I'll try it out anyway!
>
> Thanks,
>
> 2011/3/24 Roger Källberg 
>
>>  Hi George,
>> What I ment was that you need to see the MVA DN set under Media Resources
>> in the RDP CSS. But I never said that you need to put that DN in
>> PT-INTERNAL, put that in PT-XLATE-MVA and try it out.
>>
>> Sincerely
>>
>>  *Roger Källberg*
>> CCIE #26199 (Voice)
>> Consultant
>> Cygate AB
>> Eric Perssons väg 21, SE-217 62 MALMÖ
>>
>>  --
>> *Från:* George Goglidze [gogli...@gmail.com]
>> *Skickat:* den 24 mars 2011 11:09
>> *Till:* Roger Källberg
>> *Kopia:* ShinGei Yong; stden...@cisco.com; ccie_voice@onlinestudylist.com
>>
>> *Ämne:* Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA
>>
>>  Hi Roger,
>>
>>  I think you have mis-read the question. The problem we are having is the
>> following:
>>
>>  PT-INTERNAL --- extensions ,500X, 100X
>> PT-XLATE-MVA - [15]XXX -> translate calling to +44..   -> APPLIED
>> CSS-INTERNAL -- contains PT-INTERNAL
>> CSS-MVA contains PT-XLATE-MVA
>>
>>  CSS-MVA is a CSS applied to RDP, and in service parameters we set MVA
>> uses Line CSS and RDP CSS combined.
>> (left out on purpose but the rest of requierements for MVA is set
>> correctly)
>>
>>  Now user calls in to MVA number, enters PIN#, then 1, then 5002# -> call
>> fails
>>
>>  Now if we change slightly the CSS-MVA and include PT-INTERNAL directly,
>> instead of only PT-XLATE-MVA, then the same example succeeds.
>> but by doing so, we break requirement of the prior task, stating that
>> incoming calls from mobile to 500X, 100X should see +44. in
>> missed/received calls too.
>>
>>  Regards,
>>
>>
>> 2011/3/24 Roger Källberg 
>>
>>>  Hi Shingei,
>>> Have you set the MVA number under Media Resources in CUCM and is this
>>> number in a PT that can be seen by the RDP CSS and possibly also incoming
>>> CSS of the MVA GW?
>>>
>>> You also need to be able to route to this number from the MVA GW with
>>> appropriate dialpeers, if the MVA DN is in the same number range as other
>>> internal DN's then you don't need any extra DP's.
>>>
>>>
>>> I assume that you also already have changed the service parameter “Enable
>>> Mobile Voice Access” and “Mobile Voice Access Number”. The later should be
>>> set to the same as MVA DN under Media Resources.
>>>
>>>
>>>  Sincerely
>>>
>>>  *Roger Källberg*
>>> CCIE #26199 (Voice)
>>> Consultant
>>> Cygate AB
>>> Eric Perssons väg 21, SE-217 62 MALMÖ
>>>
>>>  --
>>> *Från:* ShinGei Yong [shingei.y...@gmail.com]
>>> *Skickat:* den 23 mars 2011 19:01
>>> *Till:* George Goglidze; stden...@cisco.com
>>> *Kopia:* ccie_voice@onlinestudylist.com
>>> *Ämne:* Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA
>>>
>>>Mates,
>>>
>>> I believe quite a number of us hitting this funny issue(if you search
>>> thru entire list since early last year)
>>> No sure it was caused of softbug or the question itself,unable to get
>>> further explanation or solution after all posts.
>>>
>>> Maybe as Steve 

Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[ShinGei Yong]

Mates,

I believe quite a number of us hitting this funny issue(if you search thru
entire list since early last year)
No sure it was caused of softbug or the question itself,unable to get
further explanation or solution after all posts.

Maybe as Steve suggested, just give it (mobility/ucm service) a lucky
reset,then the problem gone.
Else,just wait for next unlucky fellow to hit this issue,maybe he'll give us
the solution.Who know?

Thanks all
Shingei.


On Thu, Mar 24, 2011 at 1:27 AM, George Goglidze  wrote:

> I had the same problem last time, and I wasn't able to resolve this without
> having an internal partition in the CSS that I applied on RDP.
> but then obviously it breaks task 3.1 requierement to show the number:
> +447976852817
>
> If you find out what the problem is, let us know mate,
>
> On Mon, Mar 21, 2011 at 11:33 AM, ShinGei Yong wrote:
>
>> Hi,
>>
>> I've gone thru the entire OSL list regarding the problem as i'm facing,
>> but unfortunately didn't manage to get the answer.
>>
>> MVA has configured on BR2 gw,the remote destination manage to call in
>> to MVA# and authenticated successfully. When press option#1, which make a
>> call to either internal or external, it just failed.
>>
>> RDP CSS:css-snr-3002/pt-snr-3002
>> Rerouting
>> CSS:css-br2-unrestricted/pt-uk-emer,pt-uk-national,pt-uk-international
>>
>> Mobility Service Parameters:
>> Partial Match 10Digits, RDP + Line CSS
>>
>> ** As per question required,we need xlation rule to display mobile ANI
>> instead of internal DN **
>> Translation Pattern
>> /pt-snr-3002
>> CSS:css-phones/pt-phones
>> Use Calling Party EPNM: Checked
>>
>> 1002/pt-phones
>> 5002/pt-phones
>>
>> I can call up the internal extension [15]002, IF i include the pt-phones
>> into
>> RDP CSS,but that will caused ANI display become internal DN instead of
>> mobile ANI
>> because of closest match.
>>
>> Any idea why internal calling doesn't work?
>>
>> Thanks
>> Shingei
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Voiec View Authentication issue

[ShinGei Yong]

Can you change your authentication URL to the below:

*url authentication http://cme-ip-address/CCMCIP/authenticate.asp

*Thanks
Shingei

On Tue, Mar 22, 2011 at 5:22 AM, Jimmy  wrote:

> Hi
>
>   I have tried that as well. After doing that create cnf. Even restarted
> the routers and power off on phones as well. Still no luck.
>
> Regards
>
>
> Sent from my iPad
>
> On Mar 22, 2011, at 3:04 AM, ShinGei Yong  wrote:
>
> Can you change your authentication path?
>
> Instead of pointing to your 202.128, changed it to  your UCME ip address,
> which is your 202.1
>
> Thanks
> Shingei
>
> On Mon, Mar 21, 2011 at 9:18 PM, Prashant Patel <
> prashantpatel...@gmail.com> wrote:
>
>> Hi Amit,
>>
>> Do you have the following in the CME config
>>
>> no ip http secure-server
>> ip http server
>> ip http path flash: (or /gui - path where u have the cme files)
>> ip http auth local
>>
>> HTH
>> Prashant
>>
>> On Mon, Mar 21, 2011 at 7:19 AM, amit batra < 
>> batraji...@yahoo.com> wrote:
>>
>>>   Hello Everyone ..
>>>
>>>I have searched everywhere and tried all commands but this thing isnt
>>> working..CUE is integrated with CME (BR2). everything is working as
>>> expected.. I have configured Voiceview .
>>> When i press the services button. I can login. i can see messages in the
>>> inbox. but when i press to listen ,  i get this error message .. "
>>> Authentication error. Report this error to your system administrator"
>>>
>>> I have all the required license on the CUE ..
>>>
>>>  show  software licenses
>>> Installed license files:
>>>  - voicemail_lic.sig : 12 MAILBOX LICENSE
>>> Core:
>>>  - Application mode: CCME
>>>  - Total usable system ports: 8
>>> Voicemail/Auto Attendant:
>>>  - Max system mailbox capacity time: 6000
>>>  - Default # of general delivery mailboxes: 5
>>>  - Default # of personal mailboxes: 12
>>>  - Max # of configurable mailboxes: 17
>>> Interactive Voice Response:
>>>  - Max # of IVR sessions: Not Available
>>> Languages:
>>>  - Max installed languages: 5
>>>  - Max enabled languages: 5
>>>
>>> Here is  the CME configuration
>>>
>>> telephony-service
>>>  sdspfarm units 1
>>>  sdspfarm transcode sessions 2
>>>  sdspfarm tag 1 BR2TRANS
>>>  no auto-reg-ephone
>>>  authentication credential admin 1234
>>>  load 7960-7940 P00308000500
>>>  max-ephones 3
>>>  max-dn 10
>>>  ip source-address 10.10.202.1 port 2000
>>>  timeouts interdigit 3
>>>  system message BR2 CME
>>>  url services <http://10.10.202.128/voiceview/common/login.do>
>>> http://10.10.202.128/voiceview/common/login.do
>>>  url authentication
>>> <http://10.10.202.128/voiceview/authentication/authenticate.do>
>>> http://10.10.202.128/voiceview/authentication/authenticate.do
>>>  date-format dd-mm-yy
>>>  live-record 3609
>>>  voicemail 3600
>>>  max-conferences 8 gain -6
>>>  moh <http://music-on-hold.au>music-on-hold.au
>>>  web admin system name admin password 1234
>>>  dn-webedit
>>>  time-webedit
>>>  transfer-system full-consult
>>>  secondary-dialtone 0
>>>  create cnf-files version-stamp 7960 Mar 22 2011 00:56:56
>>>
>>> Any ideas ???
>>>
>>> Thanks in advance ..
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit <http://www.ipexpert.com/>www.ipexpert.com
>>>
>>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit <http://www.ipexpert.com>www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Troubleshooting remote bandwidth

[ShinGei Yong]

Hi Friderich,

You may also give a shoot on "debug ras", which provide the information
about the type of msg send/receive and the source and destination ip
address.
In your case, hopefully a BRJ should seen from ur remote GK.

Thanks
Shingei.

On Mon, Mar 21, 2011 at 11:03 PM, Friderich Claude wrote:

>  Hello Guys,
>
>
>
> Trying to put on  the remote GK a  BW that don’t  let me place a  call
> from my call manager, I just wonder how to debug this famous lack of
> bandwidth on my hq router.
>
>
>
> I put  bandwidth remote *15* on the pstn-wan router in order to not have
> enough bandwidth
>
>
>
> After debug, hereafter the results …… Just a normal DCF …… But how to find
> out that’s a remote bandwidth failure ??? L
>
>
>
> Mar 21 14:41:09.928: ////GK/gk_process: QUEUE_EVENT
> (minor 0) wakeup
>
> Mar 21 14:41:09.928:  RecvUDP_IPSockData  successfully rcvd message of
> length 120 from 10.10.210.11:32790
>
> Mar 21 14:41:09.928: RAS INCOMING ENCODE BUFFER::=
> 2690006103C00034003900450037003700300032003000300030003000300030003000300032010700344C49A78A6BC650010500494AB96433400A0AD20B8FD200A600F64241856371D806000B020A0AC93209642010018011F64241856371D806000B020A0AC932070400550043004D01000100
>
> Mar 21 14:41:09.928:
>
> Mar 21 14:41:09.928: RAS INCOMING PDU ::=
>
>
>
> *value RasMessage ::= admissionRequest *:
>
> {
>
>   requestSeqNum 98
>
>   callType pointToPoint : NULL
>
>   endpointIdentifier {"49E77022"}
>
>   destinationInfo
>
>   {
>
> dialedDigits : "011916745738932"
>
>   }
>
>   srcInfo
>
>   {
>
> dialedDigits : "16178631001"
>
>   }
>
>   srcCallSignalAddress ipAddress :
>
>   {
>
> ip '0A0AD20B'H
>
> port 36818
>
>   }
>
>   bandWidth 160
>
>   callReferenceValue 6
>
>   conferenceID '00F64241856371D806000B020A0AC932'H
>
>   activeMC FALSE
>
>   answerCall FALSE
>
>   canMapAlias TRUE
>
>   callIdentifier
>
>   {
>
> guid '00F64241856371D806000B020A0AC932'H
>
>   }
>
>   gatekeeperIdentifier {"UCM"}
>
>   willSupplyUUIEs FALSE
>
> }
>
>
>
>
>
>
>
> Mar 21 14:41:09.928: ARQ (seq# 98) rcvd
>
> Mar 21 14:41:09.928: ////GK/gk_rassrv_arq:
> arqp=0x4806E608,crv=0x6, answerCall=0
>
> Mar 21 14:41:09.928: ////GK/gk_rassrv_sep_arq: ARQ
> Didn't use GK_AAA_PROC
>
> Mar 21 14:41:09.928: ////GK/gk_call_new:
> src_endptp=0x49E77020, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160,
> crv=6, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x48B6C990
>
> Mar 21 14:41:09.928: ////GK/gk_call_find_endpts:
> NOT_FOUND
>
> Mar 21 14:41:09.928: ////GK/gk_call_new: checking
> for default (CLI) carrier for sep endpt 0x49E77020
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/gk_dns_query: No Name
> servers
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/rassrv_get_addrinfo:
> (011916745738932) Tech-prefix match failed.
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/rassrv_get_addrinfo:
> (011916745738932) Matched zone prefix 01191 and remainder 6745738932
>
> Mar 21 14:41:09.928:
> ////GK/gk_rassrv_get_ingress_network: returning
> default ingress network = 1
>
> Mar 21 14:41:09.928:
> //00F642410600/00F642410600/GK/rassrv_arq_select_viazone: about to check the
> source side, src_zonep=0x4A65F6EC
>
> Mar 21 14:41:09.928:
> //00F642410600/00F642410600/GK/rassrv_arq_select_viazone: matched zone is
> UCM, and z_invianamelen=0
>
> Mar 21 14:41:09.928:
> //00F642410600/00F642410600/GK/rassrv_arq_select_viazone: about to check the
> destination side, dst_zonep=0x49ABD714
>
> Mar 21 14:41:09.928:
> //00F642410600/00F642410600/GK/rassrv_arq_select_viazone: matched zone is
> PSTN-WAN, and z_outvianamelen=0
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/rassrv_get_addrinfo: No
> tech prefix
>
>
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/rassrv_get_addrinfo:
> Alias not found
>
>
>
> Mar 21 14:41:09.928:
> //00F642410600/00F642410600/GK/rassrv_put_remote_zones_from_zone_list: zone
> PSTN-WAN
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/send_lrq: seq_lrq 1,
> use_be 0, rzone_cnt 1
>
> Mar 21 14:41:09.928: //00F642410600/00F642410600/GK/send_lrq: lrq array
> index 7, lap 4A8BC340
>
> Mar 21 14:41:09.932: H225 NONSTD OUTGOING PDU ::=
>
>
>
> value LRQnonStandardInfo ::=
>
> {
>
>   ttl 6
>
>   nonstd-callIdentifier
>
>   {
>
> guid '00F64241856371D806000B020A0AC932'H
>
>   }
>
>   gatewaySrcInfo
>
>   {
>
> e164 : "16178631001"
>
>   }
>
>   h225NonStdSrcCallSignalAddress h225NonStdIpAddress :
>
>   {
>
> ip '0A0AD20B'H
>
> port 36818
>
>   }
>
>   h225NonStdSrcendpointIdentifier {"49E77022"}
>
> }
>
>
>
>
>
>
>
> Mar 21 14:41:09.932: H225 NONSTD OUTGOING ENCODE BUF

Re: [OSL | CCIE_Voice] Voiec View Authentication issue

[ShinGei Yong]

Can you change your authentication path?

Instead of pointing to your 202.128, changed it to  your UCME ip address,
which is your 202.1

Thanks
Shingei

On Mon, Mar 21, 2011 at 9:18 PM, Prashant Patel
wrote:

> Hi Amit,
>
> Do you have the following in the CME config
>
> no ip http secure-server
> ip http server
> ip http path flash: (or /gui - path where u have the cme files)
> ip http auth local
>
> HTH
> Prashant
>
> On Mon, Mar 21, 2011 at 7:19 AM, amit batra  wrote:
>
>>   Hello Everyone ..
>>
>>I have searched everywhere and tried all commands but this thing isnt
>> working..CUE is integrated with CME (BR2). everything is working as
>> expected.. I have configured Voiceview .
>> When i press the services button. I can login. i can see messages in the
>> inbox. but when i press to listen ,  i get this error message .. "
>> Authentication error. Report this error to your system administrator"
>>
>> I have all the required license on the CUE ..
>>
>>  show  software licenses
>> Installed license files:
>>  - voicemail_lic.sig : 12 MAILBOX LICENSE
>> Core:
>>  - Application mode: CCME
>>  - Total usable system ports: 8
>> Voicemail/Auto Attendant:
>>  - Max system mailbox capacity time: 6000
>>  - Default # of general delivery mailboxes: 5
>>  - Default # of personal mailboxes: 12
>>  - Max # of configurable mailboxes: 17
>> Interactive Voice Response:
>>  - Max # of IVR sessions: Not Available
>> Languages:
>>  - Max installed languages: 5
>>  - Max enabled languages: 5
>>
>> Here is  the CME configuration
>>
>> telephony-service
>>  sdspfarm units 1
>>  sdspfarm transcode sessions 2
>>  sdspfarm tag 1 BR2TRANS
>>  no auto-reg-ephone
>>  authentication credential admin 1234
>>  load 7960-7940 P00308000500
>>  max-ephones 3
>>  max-dn 10
>>  ip source-address 10.10.202.1 port 2000
>>  timeouts interdigit 3
>>  system message BR2 CME
>>  url services http://10.10.202.128/voiceview/common/login.do
>>  url authentication
>> http://10.10.202.128/voiceview/authentication/authenticate.do
>>  date-format dd-mm-yy
>>  live-record 3609
>>  voicemail 3600
>>  max-conferences 8 gain -6
>>  moh music-on-hold.au
>>  web admin system name admin password 1234
>>  dn-webedit
>>  time-webedit
>>  transfer-system full-consult
>>  secondary-dialtone 0
>>  create cnf-files version-stamp 7960 Mar 22 2011 00:56:56
>>
>> Any ideas ???
>>
>> Thanks in advance ..
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[ShinGei Yong]

Hi,

I've gone thru the entire OSL list regarding the problem as i'm facing,
but unfortunately didn't manage to get the answer.

MVA has configured on BR2 gw,the remote destination manage to call in
to MVA# and authenticated successfully. When press option#1, which make a
call to either internal or external, it just failed.

RDP CSS:css-snr-3002/pt-snr-3002
Rerouting
CSS:css-br2-unrestricted/pt-uk-emer,pt-uk-national,pt-uk-international

Mobility Service Parameters:
Partial Match 10Digits, RDP + Line CSS

** As per question required,we need xlation rule to display mobile ANI
instead of internal DN **
Translation Pattern
/pt-snr-3002
CSS:css-phones/pt-phones
Use Calling Party EPNM: Checked

1002/pt-phones
5002/pt-phones

I can call up the internal extension [15]002, IF i include the pt-phones
into
RDP CSS,but that will caused ANI display become internal DN instead of
mobile ANI
because of closest match.

Any idea why internal calling doesn't work?

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

Re: [OSL | CCIE_Voice] CUE/CME MWI config

[ShinGei Yong]

Hi Michael,

Please refer to the link below

Integration CUE and UCME 3.0 and
later
Troubleshooting CUE MWI
Problem

Thanks
Shingei

On Mon, Mar 21, 2011 at 12:38 PM, Michael Luo  wrote:

> Can anyone send me a link to CUE/CME MWI configuration example?
>
> Whenever I left a message, CUE VM pilot will call the phone instead of
> lighting up the phone.
>
> Attached are the config and debug.
>
> Thanks!
> Michael
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] MOH across GK-controlled trunk?

[ShinGei Yong]

Hi Michael,

I believe you may just ignored that part"The br2 phone need to be registered
with cucm,not with cme"

This condition is valid only when you have centralized multisite deployment
in non-fallback/fallback mode.

As per your configuration,obviously this is a distributed multisite
deployment,which you have GK to controlled
the 2 different clusters.

I'm not too sure you're meet this criteria:

Distributed Multisite Deployment
source:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/moh.html#wp1043948

Cisco Unified CM 7.1(2) and later releases support multicast MoH on
intercluster calls using an intercluster trunk (ICT) or SIP trunk. This
feature adds support for endpoints in one Unified CM cluster to hear
multicast MoH streamed from another Unified CM cluster, while making more
efficient use of intercluster bandwidth. A properly designed IP Multicast
environment is required to take advantage of this feature.

Earlier releases of Unified CM allow for only unicast MoH on intercluster
calls, which requires configuration of at least one unicast MoH resource in
each Unified CM cluster if MoH on the ICT or SIP trunk is desired.

As the explanation above stated between UCM to UCM cluster,i'm not sure UCME
will face the same.I might be wrong,but just the idea.
Since your UCME is a standalone cluster, i would try out unicast on that,
and believe that should no difference between using multi or uni on ur UCME.

Thanks
Shingei.



On Mon, Mar 21, 2011 at 6:06 AM, Michael Luo  wrote:

> I was trying to understand this part - "The br2 phone need to be registered
> with cucm. Not with cme."
>
> Why was that?
>
> From my troubleshooting, it looks like the CME phone was listening to the
> correct IP address/port - 239.1.1.1 16384.  But didn't receive any MOH
> music.
>
> I was under the impression that the music was continuously being played by
> the router.  The CME phones just need to "tune" to the right channel to
> receive music.
>
> Thanks!
> Michael
>
> On Sun, Mar 20, 2011 at 3:24 PM, Jimmy  wrote:
>
>> Mate
>>
>>Coorect me if I am wrong.  Doesn't matter if it's pstn or gatekeeper
>> trunk.
>>
>> When cucm phone press hold button. The phone at cme site will will receive
>> moh from cucm.
>>
>> For moh from flash to work. The br2 phone need to be registered with cucm.
>> Not with cme.
>>
>> I hope I am making sense. I am not good at writing emails.
>>
>>
>>
>> Regards
>>
>>
>> Sent from my iPad
>>
>> On Mar 21, 2011, at 8:17 AM, Michael Luo  wrote:
>>
>> One thing I noticed was the word "stopped" in show ip mroute output.  This
>> seems to be the problem by comparing with a working one.  But I don't know
>> how to fix it.  Thanks!
>>
>> R3#sh ip mro
>> IP Multicast Routing Table
>> Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
>> Connected,
>>L - Local, P - Pruned, R - RP-bit set, F - Register flag,
>>T - SPT-bit set, J - Join SPT, M - MSDP created entry,
>>X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,
>>U - URD, I - Received Source Specific Host Report,
>>Z - Multicast Tunnel, z - MDT-data group sender,
>>Y - Joined MDT-data group, y - Sending to MDT-data group,
>>V - RD & Vector, v - Vector
>> Outgoing interface flags: H - Hardware switched, A - Assert winner
>>  Timers: Uptime/Expires
>>  Interface state: Interface, Next-Hop or VCD, State/Mode
>>
>> (*, 239.1.1.1), 00:01:51/*stopped*, RP 0.0.0.0, flags: DCL
>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>   Outgoing interface list:
>> Vlan502, Forward/Dense, 00:00:14/00:00:00
>>
>>
>>
>> On Sun, Mar 20, 2011 at 1:48 PM, Michael Luo < 
>> hout...@gmail.com> wrote:
>>
>>>
>>> Call flow as below:
>>>
>>> CME phone -> CME (H.323 GW) -> GK -> GK-controlled Trunk -> CUCM -> UCM
>>> Phone
>>>
>>> Problem: When UCM Phone press the "Hold" button, CME phone hears silence.
>>> Expected behavior: CME phone hears MOH from router flash -
>>> music-on-hold.au.
>>>
>>> Troubleshooting done:
>>> 1) Verified MOH works as expected between CME phones. (CME phone can hear
>>> MOH)
>>> 2) Verified MOH works as expected with PSTN phones (PSTN phone can hear
>>> MOH)
>>>
>>> Config highlights:
>>>
>>> =
>>> ip multicast-routing
>>> !
>>> interface vlan502
>>>  description e-phone vlan
>>>  ip address 142.102.66.254 255.255.255.0
>>>  ip pim dense
>>>
>>> interface Loopback0
>>>  ip address 142.1.66.254 255.255.255.255
>>>
>>> ccm-manager music-on-hold
>>>
>>> telephony-service
>>>  ip source-address 142.102.66.254 port 2000
>>>  moh music-on-hold.au
>>>  multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
>>> =
>>>
>>> debug highlights:
>>> =
>>> R3#sh ccm mus
>>> Current active multicast sessions : 1
>>>  Multicast   RTP port   Packets   Call   CodecIncoming
>>>  Address number in/outid  Interface
>>> 

Re: [OSL | CCIE_Voice] No ISDN debug out?

[ShinGei Yong]

Hi Wael,

For you info,the inbound call from PSTN is working fine,it can reach all the
BR2 phones,
just that when nothing info display when i run the debug isdn q931.

I can see the debug isdn q931 info on the PSTN router.

Thanks
Shingei.

On Sun, Mar 20, 2011 at 8:06 PM, Wael Agina  wrote:

> Dear Shingei,
>
>   Asi understand you see output when you call outbound - callPSTN or other
> branch.
> So make sure that PSTN config is correct and that PSTN send the call to
> your BR2 rtr correctly.
> Check the debug on PSTN router side.
>
> Regards,
> Wael Agina
>
> On Sun, Mar 20, 2011 at 1:54 PM, ShinGei Yong wrote:
>
>> Hi
>>
>> I'm facing an issue, which i've BR2 GW registered with HQ UCM as H323 GW.
>> When i enabled the "debug isdn q931"on BR2-RTR,there is no output from
>> console/terminal
>> while PSTN call to BR2 phone,but i can view the output when making call
>> from BR2 internal phone
>> to PSTN.
>>
>> In simple words, no inbound debug isdn info display, but yes for outbound
>> call to PSTN.
>>
>> What is the cause of this?
>>
>> Thanks
>> Shingei
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
>
> --
>
> Thanks and Best Regards,
> Wael Agina
>
___
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[OSL | CCIE_Voice] No ISDN debug out?

[ShinGei Yong]

Hi

I'm facing an issue, which i've BR2 GW registered with HQ UCM as H323 GW.
When i enabled the "debug isdn q931"on BR2-RTR,there is no output from
console/terminal
while PSTN call to BR2 phone,but i can view the output when making call from
BR2 internal phone
to PSTN.

In simple words, no inbound debug isdn info display, but yes for outbound
call to PSTN.

What is the cause of this?

Thanks
Shingei
___
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Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 ---- HQ call BR-2 CUE failed

[ShinGei Yong]

Hi Erwan,

I hope we are referring to the same WB. The question is asked:
Question 6.2: Integrate UCME with Unity Connection using SIP and create
mailbox for br2 phone 1 and phone 2.(4pts)

Yes, the dp pointed to 3600,but did you look at the address? is that 210.13
or 202.2

Can tell me which page of the doc you are looking at?

Shingei

On Wed, Mar 16, 2011 at 11:08 PM, Erwan Erwan  wrote:

> I refer to the Guide, it is CUE and dialpeer point to CUE 3600
>
> --- On *Wed, 3/16/11, ShinGei Yong * wrote:
>
>
> From: ShinGei Yong 
> Subject: Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6  HQ call BR-2 CUE failed
> To: "Erwan Erwan" , ccie_voice@onlinestudylist.com
> Received: Wednesday, March 16, 2011, 4:20 PM
>
>
> Hi Erwan,
>
> Did you follow the instruction on the WB completely? Or you modified the
> question?
> The WB is requested a UCME integration with CUC with SIP, not CUE, and user
> mailbox should be located on CUC not CUE.
> You need a sip dialpeer to point to CUC with some additional configuration.
>
> So, can confirm this is a UCME + CUE or UCME + CUC ?
>
> Thanks
> Shingei.
>
> On Wed, Mar 16, 2011 at 2:30 PM, Shrini 
> http://ca.mc1205.mail.yahoo.com/mc/compose?to=linuxbos...@gmail.com>
> > wrote:
>
> uncheck the farend under GK gateway.
>
>
> On 3/15/2011 9:32 PM, Erwan Erwan wrote:
>
> hi all,
>
> I am practicing WB lab 6 , that use Gatekeeper for call from HQ (UCM)   to
> BR-2 (CME)
>
> - call from HQ-- 5001  to 3001 -- Br-2 and vice versa (thru Gatekeeper)  ,
> 4 digit ext   works fine
> - call from HQ to CUE pilot point  (3600)   (thru Gatekeeper)   in Br-2
> also work fine
> - call to BR-2 (3001) phone from PSTN also hit the CUE and I can leave the
> message
>
>
> Issue
> =
> however when I call from HQ to 3001 in BR-2  and I did not pick up the
> 3001, it supposed to transfer to CUE (3600)  so that I can leave message ,
> but instead  I heard the busy tone.
>
> * I suspect the BR-2   transcode is not triggering ,however "*sh sccp* "
> it said transcode register OK
>
>
> config
> =
> telephony-service
>  privacy off
>  sdspfarm units 1
>  sdspfarm transcode sessions 3
>  sdspfarm tag 1 sc-xcode
>  voicemail 3600
>
>
> ephone-dn  1  octo-line
>  number 3001 no-reg both
>  call-forward busy 3600
>  call-forward noan 3600 timeout 20
>
>
> ---
>
>
> Is there any other possibility I need to check ?
>
>
> Thanks
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
___
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Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 ---- HQ call BR-2 CUE failed

[ShinGei Yong]

Hi Erwan,

Did you follow the instruction on the WB completely? Or you modified the
question?
The WB is requested a UCME integration with CUC with SIP, not CUE, and user
mailbox should be located on CUC not CUE.
You need a sip dialpeer to point to CUC with some additional configuration.

So, can confirm this is a UCME + CUE or UCME + CUC ?

Thanks
Shingei.

On Wed, Mar 16, 2011 at 2:30 PM, Shrini  wrote:

>  uncheck the farend under GK gateway.
>
>
> On 3/15/2011 9:32 PM, Erwan Erwan wrote:
>
>   hi all,
>
> I am practicing WB lab 6 , that use Gatekeeper for call from HQ (UCM)   to
> BR-2 (CME)
>
> - call from HQ-- 5001  to 3001 -- Br-2 and vice versa (thru Gatekeeper)  ,
> 4 digit ext   works fine
> - call from HQ to CUE pilot point  (3600)   (thru Gatekeeper)   in Br-2
> also work fine
> - call to BR-2 (3001) phone from PSTN also hit the CUE and I can leave the
> message
>
>
> Issue
> =
> however when I call from HQ to 3001 in BR-2  and I did not pick up the
> 3001, it supposed to transfer to CUE (3600)  so that I can leave message ,
> but instead  I heard the busy tone.
>
> * I suspect the BR-2   transcode is not triggering ,however "*sh sccp* "
> it said transcode register OK
>
>
> config
> =
> telephony-service
>  privacy off
>  sdspfarm units 1
>  sdspfarm transcode sessions 3
>  sdspfarm tag 1 sc-xcode
>  voicemail 3600
>
>
> ephone-dn  1  octo-line
>  number 3001 no-reg both
>  call-forward busy 3600
>  call-forward noan 3600 timeout 20
>
>
> ---
>
>
> Is there any other possibility I need to check ?
>
>
> Thanks
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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[OSL | CCIE_Voice] 8.9 magnitute Earth Quake hit North Japan - CCIE Candidates in Japan please take note.

[ShinGei Yong]

Hi All,

A gently reminder to all ccie candidates who booked or intend to Japan ccie
lab.

Japan major airports Narita and Haneda have been evacuated and stop
operation.

Source: CNN 
Asia

Tsunami warning issued for at least 50 areas after
quake
Thanks.
Shingei.
___
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Re: [OSL | CCIE_Voice] CUCM vs QOS SRNDs

[ShinGei Yong]

Hi George,

When you said "both would be an acceptable answer on a lab",you mean real
exam lab or practice lab?
Personally think there's only one acceptable value,but of course both might
accepted during exam.

If not wrong (and this topic has been bring up multiple times in OSL or Ask
the expert)Ben Ng has emphasized that
he would used QoS SRND as a reference guideline during the exam.

I would used different value suggest by various expert for practicing,but
for real exam, will only use QoS SRND
reference. If not proctor not agree with the value defined in the SRDN, i'll
just tell him then he shouldn't be left
this on candidate desktop.

Just my opinion,hope i won't kick out by Cisco after all.

Shingei.



On Fri, Mar 11, 2011 at 6:19 PM, George Goglidze  wrote:

> Hi all,
>
> There is inconsistency between the two SRNDs on how much Layer 2 overhead
> is to be used for voice traffic calculations.
>
> CUCM SRND states the following:
>
>  CODEC
> Header Type and Size
> Ethernet
> 14 Bytes
> PPP
> 6 Bytes
> ATM
> 53-Byte Cells with a 48-Byte Payload
> Frame Relay
> 4 Bytes
> MLPPP
> 10 Bytes
> MPLS
> 4 Bytes
> WLAN
> 24 Bytes
>
>
> And QOS SRND states the following:
>
> • 802.1Q Ethernet adds (up to) 32 bytes of Layer 2 overhead.
> • Point-to-point protocol (PPP) adds 12 bytes of Layer 2 overhead.
> • Multilink PPP (MLP) adds 13 bytes of Layer 2 overhead.
> • Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12
> adds 8 bytes.
> • ATM adds varying amounts of overhead, depending on the cell padding
> requirements
>
> Which one do you use? I understand both would be an acceptable answer on a
> lab, but I'm just interested what most of the people use for their
> calculations.
>
> Thanks for any answers.
>
> George
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Local Lab and Remote Lab

[ShinGei Yong]

Kiwi,

So confidence to nailed it?

Gd Luck to you!!!

Shingei

On Wed, Mar 9, 2011 at 10:07 AM, Jon 1992  wrote:

>   Tokyo is remote, however speeds are pretty impressive.
> Bangalore, India lab is a bit slow I heard.
>
>  *From:* Ki Wi 
> *Sent:* Wednesday, March 09, 2011 12:47 AM
> *To:* OSL Questions 
> *Subject:* [OSL | CCIE_Voice] Local Lab and Remote Lab
>
>  Which ccie voice lab location is having physical rack and which location
> is remote ?
>
> I can confirm HK is remote, i went for my first attempt recently. Overall
> speed is ok except for those VNC connections you need to make.
> What make it pain in the arse is you have to use RTMT on the remote PC. On
> local candidate PC, you don't have it installed.
>
>
>
>
> --
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST

[ShinGei Yong]

Hi Rogers,

Yes, and again, as stated below, i'm able to achieved CFB and CFNA during
SRST,
so in other words,the required dial-peer and setting to route call to CUE is
already done,right?
And HQ/BR1 phone is able to call BR2 phone via PSTN during SRST.

Maybe put the question in this way:

PSTN caller which originate from HQ/BR1, ring the destination during
SRST.(I've done this)
PSTN caller which originate from PSTN, to Voicemail without ringing BR2
phones during SRST

Is the second requirement possible?

Shingei.


On Tue, Mar 8, 2011 at 10:26 PM, Rogers Ochieng wrote:

> AS you've stated you are using CUE which in normal operations you've
> integrated using jtapi CUE integration, i assuem the CUE module is on the
> BR2 router. So for SRST create a voip dial-peer using sip protocol and codec
> g711ulaw, dtmf sip-notify, to route calls to CUE and set CFB and CFNA, setup
> CUE sip settings. For HQ and BR1 to access BR2 setup CFUR
>
> On 8 March 2011 16:05, ShinGei Yong  wrote:
>
>> Hi Roger,
>>
>> As stated below, i'm able to achieved that PSTN caller routed to VM when
>> CFB and CFNA.
>> Alsothere's no CUC in this lab.
>>
>> how to achieve that PSTN caller will be route to VM while allowing HQ or
>> BR1 ring the destination in SRST site?
>> TIA
>> Shingei
>>
>> 2011/3/8 Roger Källberg 
>>
>>   You need to setup CFB & CFNA in an SRST situation, so that it sends the
>>> call over PSTN to CUC VM.
>>>
>>> Sincerely
>>>
>>>  *Roger Källberg*
>>> CCIE #26199 (Voice)
>>> Consultant
>>> Cygate AB
>>> Eric Perssons väg 21, SE-217 62 MALMÖ
>>>
>>>  --
>>> *Från:* ShinGei Yong [shingei.y...@gmail.com]
>>> *Skickat:* den 8 mars 2011 11:00
>>> *Till:* ccie_voice@onlinestudylist.com
>>> *Ämne:* [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST
>>>
>>>  Hi,
>>>
>>> The question stated,caller from PSTN "CAN BE" routed to VM when there's
>>> WAN outage at BR2.
>>> Internal caller from HQ or BR1 must be able to reach BR2 phone and
>>> forward to VM if no answer.
>>>
>>> To me,there are two meaning of the sentense
>>>
>>> 1. PSTN caller routed to VM immediately when there's WAN outage at BR2,
>>>
>>> 2. PSTN caller routed to VM when CFB or CFNA.
>>>
>>> What confuse me is that,how to achieve that the PSTN caller routed to VM
>>> immediately
>>> when there's a WAN outage at BR2?I'm able to achieved that PSTN caller
>>> router to VM
>>> when CFB and CFNA.
>>>
>>> In proctor guide, Forward Unregisterd Int and External been
>>> checked(VM),but how the UCM instruct PSTN call
>>> to VM?The PSTN call will hitting the BR2 GW directly due to SRST.
>>>
>>> Am i thinking of too much?
>>>
>>> Shingei
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
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Re: [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST

[ShinGei Yong]

Hi Roger,

As stated below, i'm able to achieved that PSTN caller routed to VM when CFB
and CFNA.
Alsothere's no CUC in this lab.

how to achieve that PSTN caller will be route to VM while allowing HQ or BR1
ring the destination in SRST site?
TIA
Shingei

2011/3/8 Roger Källberg 

>  You need to setup CFB & CFNA in an SRST situation, so that it sends the
> call over PSTN to CUC VM.
>
> Sincerely
>
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
> Eric Perssons väg 21, SE-217 62 MALMÖ
>
>  ------
> *Från:* ShinGei Yong [shingei.y...@gmail.com]
> *Skickat:* den 8 mars 2011 11:00
> *Till:* ccie_voice@onlinestudylist.com
> *Ämne:* [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST
>
>  Hi,
>
> The question stated,caller from PSTN "CAN BE" routed to VM when there's WAN
> outage at BR2.
> Internal caller from HQ or BR1 must be able to reach BR2 phone and forward
> to VM if no answer.
>
> To me,there are two meaning of the sentense
>
> 1. PSTN caller routed to VM immediately when there's WAN outage at BR2,
>
> 2. PSTN caller routed to VM when CFB or CFNA.
>
> What confuse me is that,how to achieve that the PSTN caller routed to VM
> immediately
> when there's a WAN outage at BR2?I'm able to achieved that PSTN caller
> router to VM
> when CFB and CFNA.
>
> In proctor guide, Forward Unregisterd Int and External been checked(VM),but
> how the UCM instruct PSTN call
> to VM?The PSTN call will hitting the BR2 GW directly due to SRST.
>
> Am i thinking of too much?
>
> Shingei
>
___
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[OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST

[ShinGei Yong]

Hi,

The question stated,caller from PSTN "CAN BE" routed to VM when there's WAN
outage at BR2.
Internal caller from HQ or BR1 must be able to reach BR2 phone and forward
to VM if no answer.

To me,there are two meaning of the sentense

1. PSTN caller routed to VM immediately when there's WAN outage at BR2,

2. PSTN caller routed to VM when CFB or CFNA.

What confuse me is that,how to achieve that the PSTN caller routed to VM
immediately
when there's a WAN outage at BR2?I'm able to achieved that PSTN caller
router to VM
when CFB and CFNA.

In proctor guide, Forward Unregisterd Int and External been checked(VM),but
how the UCM instruct PSTN call
to VM?The PSTN call will hitting the BR2 GW directly due to SRST.

Am i thinking of too much?

Shingei
___
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Re: [OSL | CCIE_Voice] WAN QoS

[ShinGei Yong]

Hi George,
Thanks for the suggestion,but as Pablo mentioned it doesn't support the mls
qos command.

Hi Pablo,
Thanks for you suggestion too.I did an sniffing as you advice, what i seen
was the udp and rtp ports.
I'm thinking that,could that because of old legacy of hwic (with old asic
chip)that doesn't support something new like cos-dscp map table,
and also the calculation of DSCP value different, so it can only be
recognize as rtp audio?

In fact i added in additional parameters which try to match the udp ports:

access-list 101 permit udp any any range 16384 38767
!
class-map match-any Voice-SIG
 match ip dscp cs3
 match ip dscp af31
class-map match-any Voice-RTP
 match ip dscp ef
 match protocol rtp audio
 match access-group 101
!

Class-map: Voice-RTP (match-any)
  122680 packets, 7851520 bytes
  5 minute offered rate 25000 bps, drop rate 0 bps
  Match: ip dscp ef (46)
0 packets, 0 bytes
5 minute rate 0 bps
  Match: protocol rtp audio
122680 packets, 7851520 bytes
5 minute rate 25000 bps
  Match: access-group 101
0 packets, 0 bytes
5 minute rate 0 bps

I believe this should not be an abnormal case as PL Rack did used hwic as
well.
Maybe someone can provide the idea?

Shingei

On Thu, Mar 3, 2011 at 3:02 AM, Pablo Meneses  wrote:

> Shingei,
>
> Your configuration looks fine, however would would need to make sure that
> the phone is actually marking the packets with the correct DSCP value, you
> can run a packet capture from spanning the switch port of the 4ESW blade.
>
> This is done exactly the same way you do it in a 3750 by using the "monitor
> session" command.
>
> BTW, those 4ESW do not support "mls qos" commands.
>
>  -Pablo Meneses.
>
>
> On Wed, Mar 2, 2011 at 10:31 AM, George Goglidze wrote:
>
>> have you configured "mls qos trust dscp" on switchports???
>> if this command is not present, the dscp gets rewritten to default 0.
>>
>> Regards,
>>
>>  On Wed, Mar 2, 2011 at 2:44 PM, ShinGei Yong wrote:
>>
>>> Hi All,
>>> I've question regarding to the WAN QoS. I've HQ and BR2 sites setup,
>>> BR2 router equipped with HWIC-4ESW and 2 ipphone connected.
>>>
>>> 1. I configured the class-based FRTS on the BR2-RTR as below. Initially
>>> the RTP traffic didn't match
>>> the class-map (DSCP) as defined, until i re-configure it with NBAR, then
>>> it just matched!
>>>
>>> !
>>> class-map match-any Voice-SIG
>>>  match ip dscp cs3
>>>  match ip dscp af31
>>> class-map match-any Voice-RTP
>>>  match ip dscp ef
>>>  match protocol rtp audio
>>> !
>>> policy-map WAN-EDGE
>>>  class Voice-SIG
>>> bandwidth 18
>>>  class Voice-RTP
>>> priority 24
>>>compress header ip rtp
>>> policy-map MQC-FRTS-768
>>>  class class-default
>>> shape average 729600 7296 0
>>>   service-policy WAN-EDGE
>>> !
>>> !
>>> interface Serial0/1/0.102 point-to-point
>>>  bandwidth 768
>>>  ip address 10.10.112.2 255.255.255.0
>>>  ip pim sparse-dense-mode
>>>  snmp trap link-status
>>>  frame-relay interface-dlci 102
>>>   class FR-MAP-CLASS-768
>>> !
>>> !
>>> map-class frame-relay FR-MAP-CLASS-768
>>>  frame-relay fragment 960
>>>  frame-relay fair-queue
>>>  service-policy output MQC-FRTS-768
>>> !
>>> !
>>>  Class-map: Voice-RTP (match-any)
>>>   104613 packets, 6695232 bytes
>>>   5 minute offered rate 25000 bps, drop rate 0 bps
>>>   Match: ip dscp ef (46)
>>> 0 packets, 0 bytes
>>> 5 minute rate 0 bps
>>>   Match: protocol rtp
>>> 104613 packets, 6695232 bytes
>>> 5 minute rate 25000 bps
>>>   Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0
>>>
>>> Why the RTP traffic doesn't match the RTP DSCP value?
>>> How do i configure in such a way the RTP traffic match DSCP EF instead of
>>> protocol RTP?
>>>
>>> Shingei
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] WAN QoS

[ShinGei Yong]

Hi All,
I've question regarding to the WAN QoS. I've HQ and BR2 sites setup,
BR2 router equipped with HWIC-4ESW and 2 ipphone connected.

1. I configured the class-based FRTS on the BR2-RTR as below. Initially the
RTP traffic didn't match
the class-map (DSCP) as defined, until i re-configure it with NBAR, then it
just matched!

!
class-map match-any Voice-SIG
 match ip dscp cs3
 match ip dscp af31
class-map match-any Voice-RTP
 match ip dscp ef
 match protocol rtp
!
policy-map WAN-EDGE
 class Voice-SIG
bandwidth 18
 class Voice-RTP
priority 24
   compress header ip rtp
policy-map MQC-FRTS-768
 class class-default
shape average 729600 7296 0
  service-policy WAN-EDGE
!
!
interface Serial0/1/0.102 point-to-point
 bandwidth 768
 ip address 10.10.112.2 255.255.255.0
 ip pim sparse-dense-mode
 snmp trap link-status
 frame-relay interface-dlci 102
  class FR-MAP-CLASS-768
!
!
map-class frame-relay FR-MAP-CLASS-768
 frame-relay fragment 960
 frame-relay fair-queue
 service-policy output MQC-FRTS-768
!
!
 Class-map: Voice-RTP (match-any)
  104613 packets, 6695232 bytes
  5 minute offered rate 25000 bps, drop rate 0 bps
  Match: ip dscp ef (46)
0 packets, 0 bytes
5 minute rate 0 bps
  Match: protocol rtp
104613 packets, 6695232 bytes
5 minute rate 25000 bps
  Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0

Why the RTP traffic doesn't match the RTP DSCP value?
How do i configure in such a way the RTP traffic match DSCP EF instead of
protocol RTP?

Shingei
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Pressing Message button during SRST

[ShinGei Yong]

Hi Kiwi,

What is the ANI when the call arrive HQ GW?

Shingei

On Tue, Mar 1, 2011 at 6:16 AM, Friderich Claude wrote:

>  You should have first in telephony-service or call-manager fallback the
> command voicemail 5888
>
> And a dial-peer :
>
>
>
> dial-peer voice 5600 pots
>
>  translation-profile outgoing voicemail
>
>  destination-pattern 5888
>
>  no digit-strip
>
>  port 0/1/0:23
>
>  prefix 1212394
>
>
>
>
>
>
>
> *Claude Friderich*
>
> *PreSales Support*
>
> *[image: ccvp_voice_sm]***
>
> *NETCORE PSF S.A.***
>
> 49 rue du Baerendall
>
> B.P.65 L-8201 Mamer
>
> Téléphone: 31 33 80-407
>
> Fax: 31 33 80 8-407
>
> GSM: 621 303 616
>
> E-mail: cfrider...@netcore.lu
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ki Wi
> *Sent:* lundi 28 février 2011 20:13
> *To:* OSL Questions
> *Subject:* Re: [OSL | CCIE_Voice] Pressing Message button during SRST
>
>
>
> There's one more way which is to translate the calling number from original
> number into 4 digits extension. This way works as well. I suppose this is
> the best solution? Is that "vm-integration" going to do some magic? I tried
> it just now but seems like it's not working for the message button.
>
>
> !
> voice translation-rule 1
>  rule 1 /^617863\(\)/ /\1/
> !
> voice translation-rule 5
>  rule 1 // // type any national plan any isdn
> !
> voice translation-profile voicemail
>  translate calling 1
>  translate called 5
> !
> dial-peer voice 15 pots
>  translation-profile outgoing voicemail
>  destination-pattern 912123945888
>  port 0/1/0:23
>  forward-digits 0
>  prefix 12123945888
>
>
>  On Tue, Mar 1, 2011 at 3:00 AM, Ki Wi  wrote:
>
> By default, it will end up on opening greeting when SRST user press the
> message button.
>
> Is there a way to make the users enter their own voicemail account
> directly(attempt sign in page) ?
>
> I'm aware of a way currently which is to set the calling number to  in
> the hunt pilot but the method is not so graceful. There's chances that
> someone else last 4 digits number is the same or the system will recording
> the original calling number as 4 digits instead of maybe 10 or 11 digits
> long.
>
> Any interesting workaround for this?
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
<><>___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones !!!!

[ShinGei Yong]

My fren,

Friderich is doing CUC integration with UCME, not CUE.

Is there any reason that you mwi server to be 202.1 instead of CUC address?

Shingei

On Mon, Feb 28, 2011 at 11:25 AM, Romain  wrote:

> Isn't your CUE at 10.10.202.2 ? That is the IP you need for your mwi server
> instead of .1
>
>
> Sent from my iPhone
>
> On Feb 27, 2011, at 8:12 PM, "Friderich Claude" 
> wrote:
>
>  Hello,
>
>
>
> After a couple of tests and after trying to configure this feature,
> impossible to make it work L
>
>
>
> MWI on sccp phone is working fine
>
> MWI on SIP phone doesn’t work.
>
>
>
> Normaly in sip-ua
>
> *mwi-server ipv4:10.10.202.1 unsolicited* (CUCME IP Adress in
> telephony-service)
>
>
>
> In telephony-service
>
> *mwi relay***
>
>
>
> in the voice register dn  1
>
> *mwi***
>
>
>
>
>
> *sip-ua *
>
>  retry invite 3
>
>  timers trying 300
>
> * mwi-server ipv4:10.10.202.1 expires 3600 port 5060 transport udp
> unsolicited*
>
>
>
> debug ccsip gives me the results below:
>
>
>
>
>
> Sent:
>
> NOTIFY sip:3005@10.10.202.50:5060;transport=udp SIP/2.0
>
> Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9
>
> From: ;tag=137E2EC-1A2C
>
> To: 
>
> Call-ID: <942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1>
> 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1
>
> *CSeq: 101 NOTIFY***
>
> Max-Forwards: 70
>
> Date: Sun, 27 Feb 2011 23:38:56 GMT
>
> User-Agent: Cisco-SIPGateway/IOS-12.x
>
> Event: message-summary
>
> Subscription-State: active
>
> Contact: 
>
> Content-Type: application/message-summary
>
> Content-Length: 23
>
>
>
> *Messages-Waiting: yes***
>
>
>
> Feb 27 23:38:56.959: //-1//SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> *SIP/2.0 400 Bad Request* --->
> 
>
> Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9
>
> From: ;tag=137E2EC-1A2C
>
> To: 
>
> Call-ID: <942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1>
> 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1
>
> Date: Sun, 27 Feb 2011 23:38:55 GMT
>
> *Warning: 399 Bad MWI NOTIFY* ->
> ?
>
> CSeq: 101 NOTIFY
>
> Content-Length: 0
>
>
>
> Does anybody encountered this problem concerning MWI on sip phones only ??
> Does it really work ??
>
>
>
> Any suggestions are appreciated.
>
>
>
> Regards
>
>
>
> Claude.
>
>
>
>
>
>
>
> *Claude Friderich*
>
> *PreSales Support*
>
> 
>
> *NETCORE PSF S.A.***
>
> 49 rue du Baerendall
>
> B.P.65 L-8201 Mamer
>
> Téléphone: 31 33 80-407
>
> Fax: 31 33 80 8-407
>
> GSM: 621 303 616
>
> E-mail: cfrider...@netcore.lu
>
>
>
> --
> This email was Anti Virus checked.
>
> Disclaimer
> The information in this Internet e-mail is confidential and may be legally 
> privileged. It is intended solely for the addressee. Access to this Internet 
> e-mail by anyone else is unauthorized. If you are not the intended recipient, 
> any disclosure, copying, distribution or any action taken or omitted to be 
> taken in reliance on it, is prohibited and may be unlawful.
> When addressed to our clients any opinions or advice contained in this e-mail 
> are subject to the terms and conditions expressed in our governing terms of 
> business.
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not appearing

[ShinGei Yong]

Hi Roger,

What i mean is "when the incoming call to shared line answer by
phoneB,phoneA line button light up and the ANI display on phoneA."
So when try to cBarge in by pressing the line button,it should have the
option "cbarge" and "newcall" which defined under the
ephone template remote-in-use state.

I don't think i need it during the idle state.

Thanks
Shingei.

On Sat, Feb 26, 2011 at 3:58 PM, Rogers Ochieng wrote:

> I don't get what you mean by "Somehow when SRST,the softkeys
> "cbarge"doesn't appear on the phone when the
>
> shared line was answered by another ip phone"
>
> Do you mean when you press the shared line on the phone or you want it to
> appear in idle state?
>
> On 25 February 2011 17:42, ShinGei Yong  wrote:
>
>> Hi Tamer,
>>
>> Thanks for the link, but it just stated that it doesn't work without any
>> further explanation.
>> In fact there is a workaround to resolve this by Cisco, pls refer below:
>>
>> *CSCti11843 Bug Details:*
>>
>> Shared-line Cbarge remote-in-use softkeys not working in SRST
>> Symptom:
>>
>> For Shared line if you want cbarge you must disable privacy and enable
>> remote-in-use softkey under ephone-template. However if you configured SRST
>> CME with srst mode auto-provision none or dn privacy setting on the ephone
>> remains default which means it is on. Hence you can’t disable privacy.
>>
>> *Conditions:*
>>
>> When phones fallback to SRST.
>>
>> *Workaround:*
>>
>> Only workaround is to either configure:
>>
>> ephone 1
>> privacy off
>>
>> or configure srst mode auto-provision all
>>
>>
>> And yes, i just make it work few min ago. Just that the behavior is odd.
>>
>> Shingei.
>>
>>
>>
>>
>> On Fri, Feb 25, 2011 at 10:22 PM, Tamer Ismail  wrote:
>>
>>> Check following post:
>>>
>>> http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=002799
>>>
>>> it’s saying that cbarge and sharde line privacy are supported only in
>>> cme/srst 7.1 not owr version 7.0
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Tamer Ismail
>>>
>>>
>>>
>>> *From:* ShinGei Yong [mailto:shingei.y...@gmail.com]
>>> *Sent:* Friday, February 25, 2011 4:03 PM
>>> *To:* Tamer Ismail; ccie_voice@onlinestudylist.com
>>> *Subject:* Re: [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not
>>> appearing
>>>
>>>
>>>
>>> Hi Tamer,
>>>
>>> Are you sure cBarge is not possible for SRST?
>>> I believe this topic has been posted several times since-ever, just that
>>> circumstance varies.
>>>
>>> http://ccieash.wordpress.com/2010/06/21/hardware-conferencing-ios/
>>>
>>> Shingei.
>>>
>>> On Fri, Feb 25, 2011 at 9:39 PM, Tamer Ismail  wrote:
>>>
>>> As far as I know, we can’t achieve cbarge in SRST.
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Tamer Ismail
>>>
>>>
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *ShinGei Yong
>>> *Sent:* Friday, February 25, 2011 1:33 PM
>>> *To:* ccie_voice@onlinestudylist.com
>>> *Subject:* [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not
>>> appearing
>>>
>>>
>>>
>>> Hi,
>>>
>>> I'm trying the cbarge during SRST. I've privacy turn off on my two 7961
>>> ipphone,
>>> and has assigned a ephone template with softkeys remote-in-use cbarge
>>> newcall.
>>>
>>> Somehow when SRST,the softkeys "cbarge"doesn't appear on the phone when
>>> the
>>> shared line was answered by another ipphone with the same DN
>>> configured.I've check
>>> the below items:
>>>
>>> 1. Both ipphones have privacy turn off
>>>  ephone-2[1] Mac:001E.F728.2C0C TCP socket:[2] activeLine:0 REGISTERED in
>>> SCCP ver 17/9
>>> mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
>>> caps:8 privacy:0
>>> IP:10.10.202.55 43050 7961  keepalive 6 max_line 5 available_line 3
>>> button 1: dn 1  number 3001  CM Fallback CH1   IDLE CH2
>>> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
>>> IDLE CH7   IDLE CH8   IDLE
>>> button 2

Re: [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not appearing

[ShinGei Yong]

Hi Tamer,

Thanks for the link, but it just stated that it doesn't work without any
further explanation.
In fact there is a workaround to resolve this by Cisco, pls refer below:

*CSCti11843 Bug Details:*

Shared-line Cbarge remote-in-use softkeys not working in SRST
Symptom:

For Shared line if you want cbarge you must disable privacy and enable
remote-in-use softkey under ephone-template. However if you configured SRST
CME with srst mode auto-provision none or dn privacy setting on the ephone
remains default which means it is on. Hence you can’t disable privacy.

*Conditions:*

When phones fallback to SRST.

*Workaround:*

Only workaround is to either configure:

ephone 1
privacy off

or configure srst mode auto-provision all


And yes, i just make it work few min ago. Just that the behavior is odd.

Shingei.




On Fri, Feb 25, 2011 at 10:22 PM, Tamer Ismail  wrote:

> Check following post:
>
> http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=002799
>
> it’s saying that cbarge and sharde line privacy are supported only in
> cme/srst 7.1 not owr version 7.0
>
>
>
> Best regards,
>
> Tamer Ismail
>
>
>
> *From:* ShinGei Yong [mailto:shingei.y...@gmail.com]
> *Sent:* Friday, February 25, 2011 4:03 PM
> *To:* Tamer Ismail; ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not
> appearing
>
>
>
> Hi Tamer,
>
> Are you sure cBarge is not possible for SRST?
> I believe this topic has been posted several times since-ever, just that
> circumstance varies.
>
> http://ccieash.wordpress.com/2010/06/21/hardware-conferencing-ios/
>
> Shingei.
>
> On Fri, Feb 25, 2011 at 9:39 PM, Tamer Ismail  wrote:
>
> As far as I know, we can’t achieve cbarge in SRST.
>
>
>
> Best regards,
>
> Tamer Ismail
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *ShinGei Yong
> *Sent:* Friday, February 25, 2011 1:33 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not appearing
>
>
>
> Hi,
>
> I'm trying the cbarge during SRST. I've privacy turn off on my two 7961
> ipphone,
> and has assigned a ephone template with softkeys remote-in-use cbarge
> newcall.
>
> Somehow when SRST,the softkeys "cbarge"doesn't appear on the phone when the
> shared line was answered by another ipphone with the same DN
> configured.I've check
> the below items:
>
> 1. Both ipphones have privacy turn off
>  ephone-2[1] Mac:001E.F728.2C0C TCP socket:[2] activeLine:0 REGISTERED in
> SCCP ver 17/9
> mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
> caps:8 privacy:0
> IP:10.10.202.55 43050 7961  keepalive 6 max_line 5 available_line 3
> button 1: dn 1  number 3001  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE
> button 2: dn 2  number 3080  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE shared
> Preferred Codec: g711ulaw
>
>
> ephone-3[2] Mac:0019.E8AE.9479 TCP socket:[5] activeLine:0 REGISTERED in
> SCCP ver 17/9
> mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
> caps:8 privacy:0
> IP:10.10.202.53 38999 7961  keepalive 0 max_line 5 available_line 3
> button 1: dn 3  number 3002  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE
> button 2: dn 4  number 1010  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE
> button 3: dn 2  number 3080  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE shared
> Preferred Codec: g711ulaw
>
> 2. Conference resources is registered with CME SRST.
> BR2-RTR(config)#do sh sdspfar uni regi
>
> conf-1 Device:br2-cfb TCP socket:[4]  REGISTERED in SCCP ver 17/10
> actual_stream:8 max_stream 8 IP:10.10.202.1  14335  Conference Dixieland
> keepalive 3
> Supported codec:
>  G711Ulaw
>  G711Alaw
>  G729
>  G729a
>  G729b
>  G729ab
>
> 3. Correct ephone-template assigned
>
> ephone-template  1
>  softkeys remote-in-use  CBarge Newcall
>  softkeys idle  Newcall Redial Cfwdall
>
> 4. Ad-hoc confere

Re: [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not appearing

[ShinGei Yong]

Hi Tamer,

Are you sure cBarge is not possible for SRST?
I believe this topic has been posted several times since-ever, just that
circumstance varies.

http://ccieash.wordpress.com/2010/06/21/hardware-conferencing-ios/

Shingei.

On Fri, Feb 25, 2011 at 9:39 PM, Tamer Ismail  wrote:

> As far as I know, we can’t achieve cbarge in SRST.
>
>
>
> Best regards,
>
> Tamer Ismail
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *ShinGei Yong
> *Sent:* Friday, February 25, 2011 1:33 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not appearing
>
>
>
> Hi,
>
> I'm trying the cbarge during SRST. I've privacy turn off on my two 7961
> ipphone,
> and has assigned a ephone template with softkeys remote-in-use cbarge
> newcall.
>
> Somehow when SRST,the softkeys "cbarge"doesn't appear on the phone when the
> shared line was answered by another ipphone with the same DN
> configured.I've check
> the below items:
>
> 1. Both ipphones have privacy turn off
>  ephone-2[1] Mac:001E.F728.2C0C TCP socket:[2] activeLine:0 REGISTERED in
> SCCP ver 17/9
> mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
> caps:8 privacy:0
> IP:10.10.202.55 43050 7961  keepalive 6 max_line 5 available_line 3
> button 1: dn 1  number 3001  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE
> button 2: dn 2  number 3080  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE shared
> Preferred Codec: g711ulaw
>
>
> ephone-3[2] Mac:0019.E8AE.9479 TCP socket:[5] activeLine:0 REGISTERED in
> SCCP ver 17/9
> mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
> caps:8 privacy:0
> IP:10.10.202.53 38999 7961  keepalive 0 max_line 5 available_line 3
> button 1: dn 3  number 3002  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE
> button 2: dn 4  number 1010  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE
> button 3: dn 2  number 3080  CM Fallback CH1   IDLE CH2
> IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
> IDLE CH7   IDLE CH8   IDLE shared
> Preferred Codec: g711ulaw
>
> 2. Conference resources is registered with CME SRST.
> BR2-RTR(config)#do sh sdspfar uni regi
>
> conf-1 Device:br2-cfb TCP socket:[4]  REGISTERED in SCCP ver 17/10
> actual_stream:8 max_stream 8 IP:10.10.202.1  14335  Conference Dixieland
> keepalive 3
> Supported codec:
>  G711Ulaw
>  G711Alaw
>  G729
>  G729a
>  G729b
>  G729ab
>
> 3. Correct ephone-template assigned
>
> ephone-template  1
>  softkeys remote-in-use  CBarge Newcall
>  softkeys idle  Newcall Redial Cfwdall
>
> 4. Ad-hoc conference configured
> ephone-dn  10  octo-line
>  number A001
>  conference ad-hoc
>  preference 5
>
> 5. Telephony-service Setting
> telephony-service
>  sdspfarm units 2
>  sdspfarm tag 1 br2-cfb
>  sdspfarm tag 2 br2-xcoder
>  conference hardware
>  srst mode auto-provision none
>  srst ephone template 1
>  srst dn template 1
>  srst dn line-mode dual-octo
>  max-ephones 5
>  max-dn 10 preference 5 no-reg primary
>  ip source-address 10.10.202.1 port 2000
>  system message CCIE Voice Test
>  time-zone 28
>  date-format dd-mm-yy
>  voicemail 3600
>  max-conferences 4 gain -6
>  moh music-on-hold.au
>  multicast moh 239.1.1.1 port 16384 route 10.10.110.3 10.10.202.1
>  transfer-system full-consult
>  secondary-dialtone 0
>
> Both phones and router have been reset N time,but the cbarge still doesn't
> appear.
>
> Did i miss out anything?
>
> TIA
> Shingei
>
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[OSL | CCIE_Voice] cBarge in SRST:cbarge softkey not appearing

[ShinGei Yong]

Hi,

I'm trying the cbarge during SRST. I've privacy turn off on my two 7961
ipphone,
and has assigned a ephone template with softkeys remote-in-use cbarge
newcall.

Somehow when SRST,the softkeys "cbarge"doesn't appear on the phone when the
shared line was answered by another ipphone with the same DN configured.I've
check
the below items:

1. Both ipphones have privacy turn off
 ephone-2[1] Mac:001E.F728.2C0C TCP socket:[2] activeLine:0 REGISTERED in
SCCP ver 17/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
caps:8 privacy:0
IP:10.10.202.55 43050 7961  keepalive 6 max_line 5 available_line 3
button 1: dn 1  number 3001  CM Fallback CH1   IDLE CH2
IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
IDLE CH7   IDLE CH8   IDLE
button 2: dn 2  number 3080  CM Fallback CH1   IDLE CH2
IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
IDLE CH7   IDLE CH8   IDLE shared
Preferred Codec: g711ulaw


ephone-3[2] Mac:0019.E8AE.9479 TCP socket:[5] activeLine:0 REGISTERED in
SCCP ver 17/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
caps:8 privacy:0
IP:10.10.202.53 38999 7961  keepalive 0 max_line 5 available_line 3
button 1: dn 3  number 3002  CM Fallback CH1   IDLE CH2
IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
IDLE CH7   IDLE CH8   IDLE
button 2: dn 4  number 1010  CM Fallback CH1   IDLE CH2
IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
IDLE CH7   IDLE CH8   IDLE
button 3: dn 2  number 3080  CM Fallback CH1   IDLE CH2
IDLE CH3   IDLE CH4   IDLE CH5   IDLE CH6
IDLE CH7   IDLE CH8   IDLE shared
Preferred Codec: g711ulaw

2. Conference resources is registered with CME SRST.
BR2-RTR(config)#do sh sdspfar uni regi

conf-1 Device:br2-cfb TCP socket:[4]  REGISTERED in SCCP ver 17/10
actual_stream:8 max_stream 8 IP:10.10.202.1  14335  Conference Dixieland
keepalive 3
Supported codec:
 G711Ulaw
 G711Alaw
 G729
 G729a
 G729b
 G729ab

3. Correct ephone-template assigned

ephone-template  1
 softkeys remote-in-use  CBarge Newcall
 softkeys idle  Newcall Redial Cfwdall

4. Ad-hoc conference configured
ephone-dn  10  octo-line
 number A001
 conference ad-hoc
 preference 5

5. Telephony-service Setting
telephony-service
 sdspfarm units 2
 sdspfarm tag 1 br2-cfb
 sdspfarm tag 2 br2-xcoder
 conference hardware
 srst mode auto-provision none
 srst ephone template 1
 srst dn template 1
 srst dn line-mode dual-octo
 max-ephones 5
 max-dn 10 preference 5 no-reg primary
 ip source-address 10.10.202.1 port 2000
 system message CCIE Voice Test
 time-zone 28
 date-format dd-mm-yy
 voicemail 3600
 max-conferences 4 gain -6
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 10.10.110.3 10.10.202.1
 transfer-system full-consult
 secondary-dialtone 0

Both phones and router have been reset N time,but the cbarge still doesn't
appear.

Did i miss out anything?

TIA
Shingei
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Re: [OSL | CCIE_Voice] RE : BUG ????? QoS with MLP LFI

[ShinGei Yong]

Hi Caro,

So what you're trying to said is that,this MLP LFI issue could be
potentially caused by the non-genuine WIC instead
of an IOS bugs?

shingei

On Wed, Feb 23, 2011 at 3:22 AM, mihal caro  wrote:

>
> The interfaces
>
> 90% of the WIC-1T interfaces that are on ebay are  fake they don't come
> from Cisco but from china cloned.So maybe you are working on a fake ones. I
> have Cisco WIC-1T original and they all work after reboot.
>
> regards
>
>
> On 22 February 2011 17:19, Friderich Claude  wrote:
>
>>  Hi Caro,
>>
>>
>>
>> What do you mean by Cisco Original ?
>>
>>
>>
>> Regards
>>
>> Claude
>>
>>
>>
>> *Claude Friderich*
>>
>> *PreSales Support*
>>
>> *[image: ccvp_voice_sm]***
>>
>> *NETCORE PSF S.A.***
>>
>> 49 rue du Baerendall
>>
>> B.P.65 L-8201 Mamer
>>
>> Téléphone: 31 33 80-407
>>
>> Fax: 31 33 80 8-407
>>
>> GSM: 621 303 616
>>
>> E-mail: cfrider...@netcore.lu
>>
>>
>>
>> *From:* mihal caro [mailto:mihalc...@gmail.com]
>> *Sent:* mardi 22 février 2011 12:00
>> *To:* Stutz, Bernhard
>> *Cc:* Friderich Claude; Ilya Rubinchik; Miron Kobelski; Roger Källberg;
>> ccieid...@gmail.com; ccie_voice@onlinestudylist.com;
>> ccie_voice-boun...@onlinestudylist.com
>> *Subject:* Re: [OSL | CCIE_Voice] RE : BUG ? QoS with MLP LFI
>>
>>
>>
>> Hi there
>>
>>
>>
>> Yes they can reboot the router at anytime you have to explain the issue to
>> proctor, however he is not going to help you. I had similar issues here
>> and I found out that the Serial Interface was fake. I replaced the serial
>> module with Cisco orginal and it worked.
>>
>>
>>
>>
>>
>>
>>
>>  regards.
>>
>>
>>
>>
>>
>> On 22 February 2011 09:52, Stutz, Bernhard  wrote:
>>
>> Why should your proctor reboot the router?
>>
>> I would imagine that the proctors know about this bug otherwise nobody
>> will ever solve this lab where this questions comes up with (if they really
>> reboot the routers) and I think that should make them mindful.
>>
>>
>>
>> *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *Friderich Claude
>> *Gesendet:* Dienstag, 22. Februar 2011 07:07
>> *An:* Ilya Rubinchik; Miron Kobelski; Roger Källberg; ccieid...@gmail.com
>>
>>
>> *Cc:* ccie_voice@onlinestudylist.com;
>> ccie_voice-boun...@onlinestudylist.com
>>
>> *Betreff:* [OSL | CCIE_Voice] RE : BUG ? QoS with MLP LFI
>>
>>
>>
>> and imagine a similar question during the lab and Proctor reboots your
>> routers before testing your config.
>> you gonna get a zero not only for the QoS question because the branch
>> router will not be accessible anymore 
>>
>> I tried with different ios but what I know is that during the lab they
>> should use the IOS 12.4(20)T2
>> and see below 12.4(24)T2, same behaviour 
>>
>> Regards,
>>
>> Claude
>>  Message d'origine
>> De: Ilya Rubinchik 
>> [mailto:ilya.rubinc...@followmars.com
>> ]
>> Date: mar. 2/22/2011 5:32
>> À: Friderich Claude; Miron Kobelski; Roger Källberg; ccieid...@gmail.com
>> Cc: ccie_voice@onlinestudylist.com;
>> ccie_voice-boun...@onlinestudylist.com
>> Objet : RE: [OSL | CCIE_Voice] BUG ? QoS with MLP LFI
>>
>> I saw the same multiple times. Checked all configs with R&S CCIE sitting
>> near, but nothing helped.
>> Probably it's just IOS issue. ?
>>
>>
>> --
>> Best regards,
>> Ilya Rubinchik
>> Chief UC Engineer
>> Mars Solutions Ltd.
>>
>> Skype: im_citius
>> Mob: +998 (97) 712-8456
>> Office: +998 (71) 290-7364
>>
>> From: ccie_voice-boun...@onlinestudylist.com [
>> mailto:ccie_voice-boun...@onlinestudylist.com]
>> On Behalf Of Friderich Claude
>> Sent: Tuesday, February 22, 2011 1:42 AM
>> To: Miron Kobelski; Roger Källberg; ccieid...@gmail.com
>> Cc: ccie_voice@onlinestudylist.com;
>> ccie_voice-boun...@onlinestudylist.com
>> Subject: Re: [OSL | CCIE_Voice] BUG ? QoS with MLP LFI
>>
>> Hello guys,
>>
>> First, thanks a lot guys for your reply.
>>
>> For Miron,
>>
>> Respecting your different steps, it works and virtual-access becomes up
>> immediately ?
>> But after rebooting the routers, down again ?
>>
>> I have configured the MLP LFI manually instead of  using auto-qos ..  The
>> same behavior . reboot .. Protocol down
>>
>> Finally, Miron a quicker step while  conserving auto qos config on each
>> site :
>> interface Serial0/1/0.1 point-to-point
>> bandwidth 768
>> ip ospf mtu-ignore
>> snmp trap link-status
>> no  frame-relay interface-dlci 102 ppp Virtual-Template100 wing steps :
>>
>> interface Serial0/1/0
>> shutdown
>> no ip address
>> encapsulation frame-relay IETF
>> frame-relay traffic-shaping
>> frame-relay lmi-type ansi
>> ip rsvp bandwidth
>>
>> and no shut and frame-relay again of course
>>
>> For Roger and ccieid1ot, hereby my config if you have any comments, don't
>> hesitate
>>
>> Thanks in advance,
>>
>> Regards
>>
>> Claude Friderich
>> PreSales Support
>> [ccvp_voice_sm]
>> NETCORE PSF S.A.
>> 49 rue du Baerendall
>> B.P.65 L-8201 Mamer
>> Téléphone: 31 33 80-407
>> Fax: 31 33 80 8-407
>> GSM: 621

Re: [OSL | CCIE_Voice] Phone Service at remote site issue

[ShinGei Yong]

Hi Amit,

As, mentioned i did restart and reset the phone,
but the result still same,when i pressed the service button,
it just show "No service configured"???

Why do i need CME or SRST configure on BR2 router?
I'm currently testing on the phone services.

Many thank for your  response.
Shingei

On Fri, Feb 11, 2011 at 7:37 AM, amit batra  wrote:

> Try reset your BR2 Phones..
>
> Do you have CME or SRST configure on BR2 Router ???
>
>
>
> --- On *Thu, 2/10/11, ShinGei Yong * wrote:
>
>
> From: ShinGei Yong 
> Subject: [OSL | CCIE_Voice] Phone Service at remote site issue
> To: ccie_voice@onlinestudylist.com
> Date: Thursday, February 10, 2011, 9:59 PM
>
>
> Hi,
> Im encounter a weird issue.I've configured a few
> phone service such as IPPA,Voiceview,IPMA, then assigned to
> ipphone across HQ and BR2 sites.
> Problem is,when i press the services button on HQ phone,
> the service appear as it's configured,but for BR2 phone,
> it show no services configured.Not just a single BR2 phone
> but all 3 of them having this symptom.
>
> I restarted UCM, BR2 GW, BR2 ipphone but still
> no luck,the phone service just doesn't appear.
>
> Anyone seen this issue before?
>
> Shingei.
>
>
> -Inline Attachment Follows-
>
> ___
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> visit www.ipexpert.com
>
>
> --
> No need to miss a message. Get email on-the-go
> <http://us.rd.yahoo.com/evt=43910/*http://mobile.yahoo.com/mail>
> with Yahoo! Mail for Mobile. Get 
> started.<http://us.rd.yahoo.com/evt=43910/*http://mobile.yahoo.com/mail>
>
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Re: [OSL | CCIE_Voice] CUCM/Unity Express MWI

[ShinGei Yong]

Hi,

Ensure that you have the correct CSS assigned to the CTIRP and CTI port,
which the CSS can "view"all your internal phone DN.

Also,remember to associate both above to your CUE jtapi user.

Shingei

On Fri, Feb 11, 2011 at 7:48 AM, Edwin Dotson  wrote:

> I have a CUCM with Unity Express configured using JTAPI and am having
> trouble getting MWI to turn on the phones that are assigned to the Unity
> Express.
>
>
>
> As far as I can tell everything happens through the JTAPI interface and I
> don’t see anything abnormal in the debugs.
>
>
>
> Any ideas?
>
>
>
> *Edwin Dotson*
> Senior Systems Engineer
>
> CCNA, CCVP
>
> Cisco Unity Support and IP Contact Center Express Specialist
> *AMS.NET*
> 925-245-6144 – Office
>
> 925-960-6644 – Fax
> www.ams.net
>
> *New Content & Information*
>
> Events, Webinars, Videos, Case Studies, Downloads & More!
> Click Here 
>
>
>
> *Cisco Award Winner
> *Vertical Partner of the Year, Voice Partner of the Year…
> Read More 
>
>
>
> ___
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> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] Phone Service at remote site issue

[ShinGei Yong]

Hi,
Im encounter a weird issue.I've configured a few
phone service such as IPPA,Voiceview,IPMA, then assigned to
ipphone across HQ and BR2 sites.
Problem is,when i press the services button on HQ phone,
the service appear as it's configured,but for BR2 phone,
it show no services configured.Not just a single BR2 phone
but all 3 of them having this symptom.

I restarted UCM, BR2 GW, BR2 ipphone but still
no luck,the phone service just doesn't appear.

Anyone seen this issue before?

Shingei.
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[OSL | CCIE_Voice] MVA pin not recognize?

[ShinGei Yong]

Hi,
I'm having issue on the MVA while calling from PSTN phone(act as mobile)
The setup is like below:

SiteC(PSTN#)-->SiteC_GW-->FR(WAN)-->SiteA-GW-->SiteA_UCM

The VXML configured on siteC gw bring me to the MVA service on UCM
and prompt for the PIN, no RDP prompted, as the calling no# matched the RDP.

Problem is that when i entered the PIN,the system seem not recognize the PIN
entered,it just repeatedly prompt for PIN,and the PIN entered doesn't
interrupt
the MVA welcome announcement.Normal case would be once you enter the PIN
during the announcement and it will stop.

I've dtmf-relay h245-alpha configured on the dialpeer.

What could be the issue?

TIA
Shingei
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[OSL | CCIE_Voice] Cisco 7961 not register to SRST during fallback

[ShinGei Yong]

Hi,
I've couple of 7961G ipphone unable to register to SRST
during fallback.The SRST rtr is running on IOS 12.4(22)T2,

HQ-RTR(config)#do sh call-manager-fallback
CONFIG (Version=7.0(1))
=
Version 7.0(1)

I've CIPC running on a console machine managed to registered
to SRST gateway,

HQ-RTR(config-cm-fallback)#do sh ephone register

ephone-1[0] Mac:000C.2994.F87C TCP socket:[1] activeLine:0 REGISTERED in
SCCP ver 15/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
caps:8
IP:10.10.210.200 1842 CIPC  keepalive 101 max_line 8 available_line 2
button 1: dn 1  number 5001  CM Fallback CH1   IDLE CH2
IDLE
Preferred Codec: g711ulaw

Below is the config:

call-manager-fallback
 max-conferences 6 gain -6
 transfer-system full-consult
 ip source-address 10.10.200.3 port 2000
 max-ephones 4
 max-dn 10 dual-line
 system message primary SRST MODE
 voicemail 5600
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 10.10.200.3 10.10.110.1

UCM has configure to the appropriate SRST reference,which is 200.3
under device pool.

Can't think off what could be the issue?

Thanks
Shingei.
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[OSL | CCIE_Voice] Called # manupulation under SRST circumstance

[ShinGei Yong]

Hi,

I've below setup, BR2 as a mgcp gw with fall-back configured.

dial-peer voice 9001 pots
 description *** SRST TO US ***
 destination-pattern 9001[2-9]..[2-9]..
 port 0/0/0:23
 prefix 001
!
num-exp 5... 90012123945...
num-exp 1... 90016178631...
!
When in fallback mode,BR2 users dialed the 4digit number to reach HQ,
the number get expanded to matched the outbound POTS dp with no issue,
the call can established successfully.

Problem is the called# presented on ipphone,instead of 4digit,
the screen display 90012123945001,this is not the desired result.

I did configured "no supplementary-service h225-notify cid-update"
and reloaded but the result remain.
How do i achieve the 4digit display in this case?

Thanks
Shingei.
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[OSL | CCIE_Voice] Vol2 Lab 1 UCCX Scripting issue

[ShinGei Yong]

Hi,
Maybe someone has done this can provide a hints.
I'm currently doing Vol2 lab1,uccx scripting.According to PG,
there should not have any CSQ defined in the script, but when uploaded
to UCCX Apps,there is a CSQ defined. Why there is this discrepancy?

2nd question,why do we need CSQ and phone agent?
Are we testing conferencing or agent based routing?

Thanks
Shingei.
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Re: [OSL | CCIE_Voice] Phone View UC

[ShinGei Yong]

Hi Francesc,

What is the phone model you're using? 1st or 2nd gen?
CUC is supported only on 2nd gen of Cisco IP Phone,

"A supported Cisco Unified IP Phone model, with the supported Cisco Unified
Communications
Manager version firmware installed. The following IP Phone models are
supported: 7941G,
7941G-GE, 7961G, 7961G-GE, 7970G, and 7971G-GE"

System Requirement for Unity
Connection

Shingei,

On Mon, Jan 10, 2011 at 5:42 PM, Roig Borrell, Francesc Xavier <
francesc.ro...@tecnocom.es> wrote:

>  Hi all,
>
>
>
> I am going crazing with Phone View configuration with Unity Connection. I
> have configured following
>
> http://blog.ipexpert.com/2010/11/17/setting-up-phone-view/
>
> I have reviewed the autentication URL (
> http://10.10.210.10:8080/ccmcip/authenticate.jsp),  I think that UC needs
> to authenticate with the IP Phone
>
>
>
> The system just response with “there is an error for displaying your
> msg…Please contact your administrator…
>
> Have you found this problem ?Any ideas?
>
>
>
> Thanks in advance!
>
> Francesc
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Order of endpoints in "show gatekeeper" commands output

[ShinGei Yong]

Hi Prashant,

So you think the IOS version and the type of trunk under UCM will affect the

sequence display from the "show end regis"  command?

Hi Miron,
Apologize that to hijack your thread for the question

Shingei.

On Mon, Jan 10, 2011 at 3:50 AM, Miron Kobelski  wrote:

> 12.4(24)T2
>
>
>
> On Sun, Jan 9, 2011 at 20:46, Prashant Patel 
> wrote:
>
>> Yes Miron. Can you tell me what IOS version you are using on the GK
>> router? I am using 22T.
>>
>> Thanks,
>> Prashant
>>
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] IP-IP GW Unable to call India

[ShinGei Yong]

Hi Shrini,

Personally i don't think the tech-prefix 1# will caused the call fail,as the
debug
shown,the decision tree will go to the next selection which is Zone prefix
if
tech prefix match failed,and also,tech prefix used for differentiate the
type of voice or video
GW within or outside a zone.

What i'm curious is that the remote GK(PSTN-WAN) should receive the LRQ,but
it got nothing in fact(no output from gatek main 10).I'm not sure the
network routing
is configured proper for this 2 zone?

Lastly is what Claude highlighted, the remote zone prefix should be 01191 or
just 91?
Will your HQ-RTR strip off the 011 once a matched found or still forward the
entire
matched string over to PSTN_WAN?

Interesting,trying to figure out myself here too.

Shingei


On Sun, Jan 9, 2011 at 9:56 PM, Friderich Claude wrote:

>  Hi Shrini,
>
>
>
> When you dial 9.011919849098490, your gatekeeper hq-rtr receives an ARQ
> from the gateway.
>
> 1/ Jan  9 03:56:47.631:
> //80452E7A0700/80452E7A0700/GK/rassrv_get_addrinfo:
>
> (011919849098490) Tech-prefix match failed.
>
> Your tech-prefix match failed
>
> 2/ your dialed number 011 and reminder of digits matched your zone prefix
> PSTN-WAN
>
> Jan  9 03:56:47.631: //80452E7A0700/80452E7A0700/GK/rassrv_get_addrinfo:
> (011919849098490) Matched zone prefix 011 and remainder 919849098490
> PSTN-WAN is configured as a remote zone
>
> Your target zone is not local to your hq-rtr gatekeeper, so the hq
> gatekeeper sends a LRQ to the PSTN-WAN GK
>
>
>
> PSTN-WAN  GK receives the LRQ from HQ GK
>
>
>
> 1/you didn’t configure a tech-prefix, and tech-prefix match failed
>
> 2/ locally at PSTN-WAN GK, regarding the config, the zone prefix at
> PSTN-WAN GK didn’t match the dialed number.
>
> PSTN#sh run | sec gatekeep
> gatekeeper
>   zone local PSTN-WAN cisco.com 10.21.200.2
>   zone remote US cisco.com 10.10.110.1 1719
>   zone prefix PSTN-WAN 91*
>
>
>
>  3/ your target zone = local zone = pstn-wan
>
>
>
> After that, if your tech-prefix didn’t match (step 4), or if your target
> address is not registered (step 5) and a default-tech-prefix was not set
> (step 6), your call is going to fail…..
>
>
>
> Let me know
>
>
>
> Regards
>
>
>
> Claude.
>
>
>
> *Claude Friderich*
>
> *PreSales Support*
>
> *[image: ccvp_voice_sm]***
>
> *NETCORE PSF S.A.***
>
> 49 rue du Baerendall
>
> B.P.65 L-8201 Mamer
>
> Téléphone: 31 33 80-407
>
> Fax: 31 33 80 8-407
>
> GSM: 621 303 616
>
> E-mail: cfrider...@netcore.lu
>
>
>
> *From:* Shrini [mailto:linuxbos...@gmail.com]
> *Sent:* dimanche 9 janvier 2011 10:02
> *To:* Friderich Claude; ccie_voice@onlinestudylist.com
>
> *Subject:* RE: [OSL | CCIE_Voice] IP-IP GW Unable to call India
>
>
>
> Hi Friderich,
>
>
>
> I want to configure without def tech prefix
>
>
>
> Thanks
> Shrini
>
>
>  --
>
> *From:* Friderich Claude [mailto:cfrider...@netcore.lu]
> *Sent:* Sunday, January 09, 2011 12:58 AM
> *To:* Shrini; ccie_voice@onlinestudylist.com
> *Subject:* RE : [OSL | CCIE_Voice] IP-IP GW Unable to call India
>
>
>
> Hello,
>
> did you configure a default tech prefix ?
>
>
>  Message d'origine
> De: ccie_voice-boun...@onlinestudylist.com de la part de Shrini
> Date: dim. 1/9/2011 5:23
> À: ccie_voice@onlinestudylist.com
> Objet : [OSL | CCIE_Voice] IP-IP GW Unable to call India
>
> Hi All,
> I am working on IPIPGW and below is my config.
> When I dial India number 9.011919849098490, below is the message I get
> in gatekeeper debug
> Jan  9 03:56:47.631: //80452E7A0700/80452E7A0700/GK/rassrv_get_addrinfo:
> (011919849098490) Tech-prefix match failed.
> Jan  9 03:56:47.631: //80452E7A0700/80452E7A0700/GK/rassrv_get_addrinfo:
> (011919849098490) Matched zone prefix 011 and remainder 919849098490
> I configured it earlier and worked fine, now I am doing something wrong,
> unfortunately not able to figure it out.
> Here is the config from my HQ and PSTN router
> HQ-RTR#sh run | sec gatekee
> gatekeeper
>   zone local US cisco.com 10.10.110.1
>   zone local VIA cisco.com
>   zone remote PSTN-WAN cisco.com 10.21.200.2 1719 outvia VIA
>   zone prefix PSTN-WAN 011*
>   no shutdown
> HQ-RTR#sh run int fa0/0.11
> Building configuration...
> Current configuration : 293 bytes
> !
> interface FastEthernet0/0.11
>   encapsulation dot1Q 11
>   ip address 10.10.200.3 255.255.255.0
>   ip helper-address 177.1.10.10
>   h323-gateway voip interface
>   h323-gateway voip id VIA ipaddr 10.10.110.1 1719
>   h323-gateway voip h323-id IPIPGW
>   h323-gateway voip bind srcaddr 10.10.200.3
> end
> HQ-RTR#sh run | sec dial-peer
> dial-peer voice 919 voip
>   destination-pattern .T
>   session target ras
>   incoming called-number .
>   dtmf-relay h245-alphanumeric
>   no vad
> HQ-RTR#sh gateway
> H.323 ITU-T Version: 4.0   H323 Stack Version: 0.1
>   H.323 service is up
>   Gateway  IPIPGW  is registered to Gatekeeper VIA
> Alias list (CLI configured)
>   H323-ID IPIPGW
> Alias list (last RCF)
>   H323-ID IPIPGW
>   H323 resource thresholding 

Re: [OSL | CCIE_Voice] IP-IP GW Unable to call India

[ShinGei Yong]

Hi Shrini,

Can you paste the debug "gatek main 10" on PSTN router?
Just want to ensure that the called number reach PSTN GK with correct digit.

Thanks
Shingei.



On Sun, Jan 9, 2011 at 12:23 PM, Shrini  wrote:

>  Hi All,
>
> I am working on IPIPGW and below is my config.
>
> When I dial India number 9.011919849098490, below is the message I get in
> gatekeeper debug
>
> Jan  9 03:56:47.631: //80452E7A0700/80452E7A0700/GK/rassrv_get_addrinfo:
> (011919849098490) Tech-prefix match failed.
> Jan  9 03:56:47.631: //80452E7A0700/80452E7A0700/GK/rassrv_get_addrinfo:
> (011919849098490) Matched zone prefix 011 and remainder 919849098490
>
> I configured it earlier and worked fine, now I am doing something wrong,
> unfortunately not able to figure it out.
>
> Here is the config from my HQ and PSTN router
>
> HQ-RTR#sh run | sec gatekee
> gatekeeper
>  zone local US cisco.com 10.10.110.1
>  zone local VIA cisco.com
>  zone remote PSTN-WAN cisco.com 10.21.200.2 1719 outvia VIA
>  zone prefix PSTN-WAN 011*
>  no shutdown
> HQ-RTR#sh run int fa0/0.11
> Building configuration...
>
> Current configuration : 293 bytes
> !
> interface FastEthernet0/0.11
>  encapsulation dot1Q 11
>  ip address 10.10.200.3 255.255.255.0
>  ip helper-address 177.1.10.10
>  h323-gateway voip interface
>  h323-gateway voip id VIA ipaddr 10.10.110.1 1719
>  h323-gateway voip h323-id IPIPGW
>  h323-gateway voip bind srcaddr 10.10.200.3
> end
>
> HQ-RTR#sh run | sec dial-peer
> dial-peer voice 919 voip
>  destination-pattern .T
>  session target ras
>  incoming called-number .
>  dtmf-relay h245-alphanumeric
>  no vad
>
> HQ-RTR#sh gateway
> H.323 ITU-T Version: 4.0   H323 Stack Version: 0.1
>
>  H.323 service is up
>  Gateway  IPIPGW  is registered to Gatekeeper VIA
>
> Alias list (CLI configured)
>  H323-ID IPIPGW
> Alias list (last RCF)
>  H323-ID IPIPGW
>
>  H323 resource thresholding is Disabled
>
> HQ-RTR#sh gatekeeper endpoints
> GATEKEEPER ENDPOINT REGISTRATION
> 
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
>
> --- - --- - - -
>
> 177.1.10.10 1720  177.1.10.10 32829 USVOIP-GW
> H323-ID: gk-trunk_1
> Voice Capacity Max.=  Avail.=  Current.= 0
> 177.1.10.20 1720  177.1.10.20 32785 USVOIP-GW
> H323-ID: gk-trunk_2
> Voice Capacity Max.=  Avail.=  Current.= 0
> 142.1.66.2541720  10.10.110.3 63145 USH323-GW
> H323-ID: CUCME
> Voice Capacity Max.=  Avail.=  Current.= 0
> 142.102.64.254  1720  10.10.200.3 65497 VIA   VOIP-GW
> H323-ID: IPIPGW
> Voice Capacity Max.=  Avail.=  Current.= 0
> Total number of active registrations = 4
>
>
> PSTN#sh run | sec gatekeep
> gatekeeper
>  zone local PSTN-WAN cisco.com 10.21.200.2
>  zone remote US cisco.com 10.10.110.1 1719
>  zone prefix PSTN-WAN 91*
>  no shutdown
> PSTN#sh gatekee
> PSTN#sh gatekeeper en
> PSTN#sh gatekeeper endpoints
> GATEKEEPER ENDPOINT REGISTRATION
> 
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
>
> --- - --- - - -
>
> 10.21.200.2 1720  10.21.200.2 50553 PSTN-WAN  H323-GW
> E164-ID: 911
> H323-ID: pstn-gw
> Voice Capacity Max.=  Avail.=  Current.= 0
> Total number of active registrations = 1
>
> PSTN#
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation

[ShinGei Yong]

Hi Shrini,

The example you give was the RSVP calculation.In your example you calculated
all the calls
(total 4 calls)in worst case scenario,which is 40kbps per calls.

I'm not very agreed with your calculation, as per SRND pages 3-65 stated:
To provision 4 G729 streams:
(3*24) + 40 =112kbps

Only Nth call will be calculated in worst case instead of 4.

Also,my previous example is mean for LLQ, not for RSVP bandwidth.

Thanks
Shingei.

On Thu, Jan 6, 2011 at 8:02 AM,  wrote:

>  Hi Shrini,
>
>
>
> I believe you’re correct as well, but you were detailing the RSVP BW
> calculation not the LLQ which the question was asking.
>
>
>
> Jeff
>
>
>
> *From:* Shrini [mailto:linuxbos...@gmail.com]
> *Sent:* Wednesday, January 05, 2011 3:32 PM
>
> *To:* Roig Borrell, Francesc Xavier
> *Cc:* givemeccievoice2...@gmail.com; 'Miron Kobelski';
> ccie_voice@onlinestudylist.com
>
> *Subject:* Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
>
>
>
> Thanks for the details debugs Jeff.
>
> Just wanted to double check with you that my examples are also correct ?
>
> Thanks again
> Shrini
>
> On 1/5/2011 2:42 PM, Roig Borrell, Francesc Xavier wrote:
>
> Hi guys,
>
>
>
> Yes, thinking twice it doesn’t make  a lot of sense consider the call with
> the worst case payload (46.4) in order to adding RSVP signaling.
>
>
>
> 1 RSVP Request
>
> Dec 17 18:47:58.630: RSVP 10.10.110.2_16548->10.10.110.1_17938[0.0.0.0]:
> start requesting 40 kbps FF reservation for 10.10.110.2
>
>
>
> 2 RSVP update (Call established )
>
> Dec 17 18:49:10.047: RSVP-RESV: Locally created reservation. No
> admission/traffic control needed
>
> Dec 17 18:49:10.047: RSVP 10.10.110.2_16510->10.10.110.1_19416[0.0.0.0]:
> start requesting 24 kbps FF reservation for 10.10.110.2
>
>
>
> In fact in the first step, there isn’t RTP traffic, so in case of
> congestion the PQ only will have some RSVP packets.
>
>
>
> So the requirement can be achieved PQ = 27,2*2 + X, X (extra RSVP signaling
> traffic, as Miron we can consider 1kbps)
>
>
>
> Now, I believe we all agree!! J
>
>
>
> Thanks for your help! Happy studies!
>
> Francesc
>
>
>
> *De:* givemeccievoice2...@gmail.com 
> [mailto:givemeccievoice2...@gmail.com]
>
> *Enviado el:* miércoles, 05 de enero de 2011 22:42
> *Para:* 'Miron Kobelski'
> *CC:* Roig Borrell, Francesc Xavier; 'Shrini';
> ccie_voice@onlinestudylist.com
> *Asunto:* RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
>
>
>
> After I just agreed with you!  J
>
>
>
> Below is not the RSVP calculation.  That is the LLQ bandwidth
> calculations.  After I reviewed my notes and figured out the value
> necessary, I referred to the PG.  The PG calculates the PQ bandwidth by
> using 1 call at 10ms and 1 call at 20ms.  I am confused as to why they do it
> this way.  I would think that you would use the 27.2 Kbps for each call and
> arrive at a 55 Kbps BW in the LLQ.  I agree with you that the RSVP
> communications will only require minimal overhead and you can just simply
> add a couple of Kbps to accomplish this task.
>
>
>
> Remember, the question that Francesc was referring to assumes you have RSVP
> configured already, and is asking you to configure the LLQ including the
> necessary overhead for RSVP messages.
>
> Jeff
>
>
>
> *From:* Miron Kobelski [mailto:findko...@gmail.com ]
> *Sent:* Wednesday, January 05, 2011 1:13 PM
> *To:* givemeccievoice2...@gmail.com
> *Cc:* Roig Borrell, Francesc Xavier; Shrini;
> ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
>
>
>
> I disagree... I would never include L2 in RSVP bandwidth calculations.
> To see what values RSVP uses, check "show ip rsvp installed" in ringing and
> connected states. it is 40 and 24 kbps for g729.
>
> I'd say that RSVP overhead should constitute no more then 1kbps (only
> several small messages during RSVP negotations!)
>
> regards
> kobel
>
> On Wed, Jan 5, 2011 at 20:53,  wrote:
>
> FRF.12 – 8
>
> 40 + 20 + 8 = 68
>
>
>
> 68 bytes * 8 bits = 544 bits per packet
>
> 544 bpp * 50 pps = 272000 bps or 27.2 Kbps
>
>
>
> 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps
>
>
>
> A basic LLQ without RSVP overhead would need to have a priority 55
> command.  However, the question asks for you to take this extra overhead for
> RSVP into account.
>
>
>
> IP/UDP/RTP - 40
>
> Payload – 10
>
> FRF.12 – 8
>
> 40 + 10 + 8 = 58 bytes
>
>
>
> 58 * 8 = 464 bpp
>
> 464 * 100 pps = 46400 bps or 46.4 Kbps
>
>
>
> Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6
> Kbps or 74 Kbps.
>
>
>
> Hope this helps,
>
> Jeff
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation

[ShinGei Yong]

Hi,

Take your example,my calculation for LLQ would be:

(3 calls x 26.8kbps) + (1 call x  40kbps)
= 120.4kbps ==>121kbps

I provision the first 3 calls in L2 bandwidth calculation,then i'll used
L3 bandwidth calculation for the 4th call,which is the worst case.
So i'll configure the PQ with above bandwidth.

Shingei.



On Thu, Jan 6, 2011 at 1:34 AM, Shrini  wrote:

>  Just found another archive straight from Vik's mail box. In more detail
>
> http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg09933.html
>
> Thanks
> Shrini
>
> On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote:
>
>  Hi everyone!
>
>
>
> I am trying to understand the right way to calculate the priority value in
> LLQ with a RSVP configuration.
>
> I have not been able to find  documentation clarifying this.
>
>
>
> So supposing HQ-BR1 4 calls g729
>
>
>
> ip rsvp bandwitdh = 24*3 + 40 = 112
>
>
>
> No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one
> call with the worst case 10ms sample rate.
>
> So following this and considering FR12 . The priority queue should be
> calculated this way
>
>
>
> L2   7 L2   7
>
> L3   40  L3   40
>
> Payload 20 Payload 10
>
>
>
> 67*8*50= 26,8kbps57*8*100 =
> 45,6kbps
>
>
>
> LLQ
>
> priority = 28,6*3  + 45,6 = 131,4 ->132
>
>
>
>
>
> Do you agree? Is it the right way?
>
>
>
> Thanks in advance!
>
> Francesc
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Mobile Voice Access

[ShinGei Yong]

Hi Shrini,

What about your UCM configuration?
1.> is your H323 GW registered with UCM?
2.> what is your dialing behavior internally?4 or 10?if is 4,
then your in & outbound dp should be 4 digit patten as well
instead of 10.

Please provide more info

Shingei

On Sun, Jan 2, 2011 at 4:13 PM, Shrini  wrote:

>  Hi Experts,
>
> *"Wish you all a Happy and Prosperous New Year 2011"*
>
> First question this year :-)
>
> HQ Site is MGCP.
>
> When I call HQ Phone 5002 , HQ PSTN is ringing  -- > all is good.
>
> I have configured MVA number 5999 in service parameters and
> Media Resources --> MVA --> 5999 / PT-INTERNAL / English
>
> on router.
>
> application
>   service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
>   !
> !
> dial-peer voice 5999 pots
>  service cmm
>  incoming called-number 2123945999
>  no digit-strip
>
> Also on CUCM :
>
>
> But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy
> tone. It is not invoking the vxml script.
>
>
> What am I doing wrong here ?
>
> TIA
> Shrini
>
>
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Mobile Voice Access

[ShinGei Yong]

Hi Roger,

MVA on a MGCP control gateway is possible.In fact,that is a coexisting of
both
MGCP and H323 on the same gateway,but you could not used a MGCP
control PRI for MVA.

you may refer to Netpro for own interest.
https://supportforums.cisco.com/thread/2005673

Shingei.

On Sun, Jan 2, 2011 at 11:16 PM, Shrini  wrote:

>  Hi ShinGei , bkvalentine, Rogers et al
>
> I remember it was successful last time when I configured it another lab
> when HQ was h323.
>
> Now I was confused around dial-peers hence had the question.
>
> I will give a try now with MGCP + H323 on HQ and it should work.
>
> Thanks all.
> Shrini
>
>  *From:* ShinGei Yong [mailto:shingei.y...@gmail.com]
> *Sent:* Sunday, January 02, 2011 6:42 AM
> *To:* Shrini; ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] Mobile Voice Access
>
> Hi Shrini,
>
> What about your UCM configuration?
> 1.> is your H323 GW registered with UCM?
> 2.> what is your dialing behavior internally?4 or 10?if is 4,
> then your in & outbound dp should be 4 digit patten as well
> instead of 10.
>
> Please provide more info
>
> Shingei
>
> On Sun, Jan 2, 2011 at 4:13 PM, Shrini  wrote:
>
>>  Hi Experts,
>>
>> *"Wish you all a Happy and Prosperous New Year 2011"*
>>
>> First question this year :-)
>>
>> HQ Site is MGCP.
>>
>> When I call HQ Phone 5002 , HQ PSTN is ringing  -- > all is good.
>>
>> I have configured MVA number 5999 in service parameters and
>> Media Resources --> MVA --> 5999 / PT-INTERNAL / English
>>
>> on router.
>>
>> application
>>   service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
>>   !
>> !
>> dial-peer voice 5999 pots
>>  service cmm
>>  incoming called-number 2123945999
>>  no digit-strip
>>
>> Also on CUCM :
>>
>>
>> But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy
>> tone. It is not invoking the vxml script.
>>
>>
>> What am I doing wrong here ?
>>
>> TIA
>> Shrini
>>
>>
>>
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] Vol2 Lab 6 Question 4.2 Hardcode UCM Trunk Port

[ShinGei Yong]

Hi,

How can i hardcode the RAS port that used by the UCM trunk to register with
GK
to the port number that i want instead of the dynamically choose of by the
system?
I know you may put in the device name (gk-trunk) under UCM service param,but
that's for port 1719 and 1720.

Any clue?
Shingei
___
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Re: [OSL | CCIE_Voice] UCCX AA

[ShinGei Yong]

Hi Gregg,

If you're sure that the aa.aef is invoked while a call is placed to the
trigger,
are you using a customized "aa.aef" or the original aa.aef? can you check
whether the audio file (AAWelcome.wav) is on stored on the prompt directory.
I can't recall the exact path,but should be something like this
"wvffid\prompt\g711\en_US\..."
The default AAWelcome is stored here.

if you look at the aa.aef variable under uccx editor,should be like this
SP[AA\AAWelcome.wav]

Shingei.

On Thu, Dec 30, 2010 at 5:38 AM, Gregg Malcolm wrote:

> I'm following along with Vik's BLS video and one of the steps following
> integration is the set up of the AA.  I've configured the call control
> group, the application and the trigger for AA.  When I call the AA trigger
> (5710) I get connected but I hear no welcome greeting. I've checked that all
> the services are running and the codecs are all G711.  Under   time reporting>  I see script "aa.aef" is invoked when I place
> a call.  What are the steps involved in troubleshooting this? Everything
> appears to be configured correctly.
>
> Thanks, Gregg
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] IPPM on cisco7961 didn't alert

[ShinGei Yong]

Guys,
Pls ignore this mail, has managed to figured out the caused.

thanks
Shingei.

On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong wrote:

> Hi,
> I've configure the IPPM on cisco 7961 phone,
> everything works smooth other that the message receive alert.
> It doesn't "ring" when there is a mgs come in from CIPC or
> other IPPM.i've set the "audible alert" to ON but still got
> no luck.
>
> Another IPPM phone encounter the same issue, so don't think
> is the phone problem. Any idea?
>
>
> Thanks
> Shingei.
>
___
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[OSL | CCIE_Voice] IPPM on cisco7961 didn't alert

[ShinGei Yong]

Hi,
I've configure the IPPM on cisco 7961 phone,
everything works smooth other that the message receive alert.
It doesn't "ring" when there is a mgs come in from CIPC or
other IPPM.i've set the "audible alert" to ON but still got
no luck.

Another IPPM phone encounter the same issue, so don't think
is the phone problem. Any idea?


Thanks
Shingei.
___
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Re: [OSL | CCIE_Voice] Volume 2 Lab 5 Task 2.3

[ShinGei Yong]

Hi Jeff,

Thanks for the clarification on that.

Or would you mind to give any hints, is there anyway/configuration that can
affect the globalized display in missed/received entry aside from
"Globalized" in under the GW inbound calling party settings?

Thanks
Shingei


On Wed, Dec 22, 2010 at 2:45 PM, CCIE  wrote:

> I can't speak to what the PSTN should send. You should contact ipx if you
> need to. I don't have the lab to reference right now but just work with what
> you have. The concepts are the same.
>
>
> On Dec 22, 2010, at 12:34 AM, ShinGei Yong  wrote:
>
>  Hi Brian,
>
> I posted the same question on last month,which regard to the same question
> no#,
> but didn't get any response.
>
> Hi Jeff,
> If i'm understand the question correctly, it's asked to displayed 10 digit
> while
> omitting the preceding 0 from PSTN,which mean,the PSTN WILL send the
> 11 digit ANI,but actual fact is that the PSTN only sent 10 digit ANI
> instead of
> 11 digit ANI.
>
> Or maybe put the question simple, should the PSTN send 10 or 11 digits in
> this question?
>
> Please advice
> Shingei.
>
>
> <http://onlinestudylist.com/archives/ccie_voice/2010-November/071028.html>
> http://onlinestudylist.com/archives/ccie_voice/2010-November/071028.html
>
> On Tue, Dec 21, 2010 at 1:15 PM, < 
> givemeccievoice2...@gmail.com> wrote:
>
>>  Hi Brian,
>>
>>
>>
>> I would change back their pattern as they are testing you on the following
>> concepts.
>>
>>
>>
>> When the call comes into the GW on CUCM, the prefix values configured will
>> be added to the front of the number.  In order to figure out how you will
>> prefix this you need to look at the debug isdn q931 and work with what
>> you’ve got.  Meaning, if you are receiving a Subscriber number 2059432785
>> then you need to manipulate the Subscriber prefix.  This will effect what
>> you will see in the Missed Calls and solve part 2.
>>
>>
>>
>> Now, the phone’s device pool has a calling party transformation calling
>> search space applied to it.  This will be used incoming on the phone and
>> determine how you will localize the calls on the screen.  You will need to
>> have a Calling Party Xformation pattern that will manipulate it back down.
>> This effects the screen and nothing else.
>>
>>
>>
>> I’m trying not to give the answer as understanding these concepts is key
>> and you can solve this in 2 mins once you understand how these all work
>> together.  Try and fool around with these values and let us know how it
>> goes.
>>
>>
>>
>> Hope this helps,
>>
>> Jeff
>>
>>
>>
>> *From:* 
>> ccie_voice-boun...@onlinestudylist.com 
>> [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Brian Rudy
>> *Sent:* Monday, December 20, 2010 8:04 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] Volume 2 Lab 5 Task 2.3
>>
>>
>>
>> I have concluded the "preceding 0 that comes in from the PSTN" was
>> incorrect in the WB and PG?  These are the tasks where it talks
>> about having +44 020 5943 2785 in the missed/received calls directory.
>> First, I was not getting the 0 or National in my debug isdn q931 on inbound
>> calls from the 2nd Button on
>> PSTN Phone to HQ Phone 2.  I changed the translation pattern on the PSTN
>> router
>>
>> FROM
>>
>> rule 2 /^2059432785$/ /\0/ type any subscriber plan any isdn
>>
>> TO
>>
>> rule 2 /^2059432785$/ /0\0/ type any national plan any isdn
>>
>>
>> However, if i strip the 0 (National Number - 44:1) at the HQ Gateway and
>> Globalize the Number, then how is going to show in the missed/received calls
>> directory as +44 020 5943 2785?  Any insight to this is greatly appreciated!
>>
>> Brian
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit <http://www.ipexpert.com>www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Volume 2 Lab 5 Task 2.3

[ShinGei Yong]

 Hi Brian,

I posted the same question on last month,which regard to the same question
no#,
but didn't get any response.

Hi Jeff,
If i'm understand the question correctly, it's asked to displayed 10 digit
while
omitting the preceding 0 from PSTN,which mean,the PSTN WILL send the
11 digit ANI,but actual fact is that the PSTN only sent 10 digit ANI instead
of
11 digit ANI.

Or maybe put the question simple, should the PSTN send 10 or 11 digits in
this question?

Please advice
Shingei.


http://onlinestudylist.com/archives/ccie_voice/2010-November/071028.html

On Tue, Dec 21, 2010 at 1:15 PM,  wrote:

>  Hi Brian,
>
>
>
> I would change back their pattern as they are testing you on the following
> concepts.
>
>
>
> When the call comes into the GW on CUCM, the prefix values configured will
> be added to the front of the number.  In order to figure out how you will
> prefix this you need to look at the debug isdn q931 and work with what
> you’ve got.  Meaning, if you are receiving a Subscriber number 2059432785
> then you need to manipulate the Subscriber prefix.  This will effect what
> you will see in the Missed Calls and solve part 2.
>
>
>
> Now, the phone’s device pool has a calling party transformation calling
> search space applied to it.  This will be used incoming on the phone and
> determine how you will localize the calls on the screen.  You will need to
> have a Calling Party Xformation pattern that will manipulate it back down.
> This effects the screen and nothing else.
>
>
>
> I’m trying not to give the answer as understanding these concepts is key
> and you can solve this in 2 mins once you understand how these all work
> together.  Try and fool around with these values and let us know how it
> goes.
>
>
>
> Hope this helps,
>
> Jeff
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Brian Rudy
> *Sent:* Monday, December 20, 2010 8:04 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Volume 2 Lab 5 Task 2.3
>
>
>
> I have concluded the "preceding 0 that comes in from the PSTN" was
> incorrect in the WB and PG?  These are the tasks where it talks
> about having +44 020 5943 2785 in the missed/received calls directory.
> First, I was not getting the 0 or National in my debug isdn q931 on inbound
> calls from the 2nd Button on
> PSTN Phone to HQ Phone 2.  I changed the translation pattern on the PSTN
> router
>
> FROM
>
> rule 2 /^2059432785$/ /\0/ type any subscriber plan any isdn
>
> TO
>
> rule 2 /^2059432785$/ /0\0/ type any national plan any isdn
>
>
> However, if i strip the 0 (National Number - 44:1) at the HQ Gateway and
> Globalize the Number, then how is going to show in the missed/received calls
> directory as +44 020 5943 2785?  Any insight to this is greatly appreciated!
>
> Brian
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab 8 Question 7.1 HA

[ShinGei Yong]

Hi Prashant,

Not sure i'm understand ur explaination correctly.
The question asked NOT to use translation-rule, but the PG solution with
translation rule,
So not sure i'm understand the question correctly.

Please advice.

Shingei

On Tue, Dec 21, 2010 at 10:24 PM, Prashant Patel  wrote:

> Hi Shin,
>
> Use "dial-plan pattern"
>
> HTH
> Prashant
>
> On Tue, Dec 21, 2010 at 9:21 AM, ShinGei Yong wrote:
>
>> Hi,
>>
>> The question asked to configure BR1 to allow certain call during SRST
>> failover
>> without the use of voice translation rule, but the PG solution was
>> achieved it
>> with translation-rule. Can anyone explain the meaning of this discrepancy?
>>
>> TIA
>> Shingei.
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Vol2 Lab 8 Question 7.1 HA

[ShinGei Yong]

Hi,

The question asked to configure BR1 to allow certain call during SRST
failover
without the use of voice translation rule, but the PG solution was achieved
it
with translation-rule. Can anyone explain the meaning of this discrepancy?

TIA
Shingei.
___
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www.ipexpert.com

[OSL | CCIE_Voice] Router stop function after apply auto qos

[ShinGei Yong]

Hi,

Has anyone encounter the same error as mine?
Once i applied the command "auto qos voip fr-atm",
the router just stop respond.You may still enter any command,
but you won't get any result. Below is the error msg:
The error showing that is a memory problem, i've a 128MB DRAM
installed. The router IOS is 12.4(22)T,chassis is cisco 2801.

Q-RTR(config-fr-dlci)#auto qos voip fr-atm
% NBAR ERROR: due to memory problem, parsing stopped
% NBAR Error : Activation failed due to insufficient dynamic memory
HQ-RTR(config-fr-dlci)#
.Dec 17 01:20:21.371: %SYS-2-MALLOCFAIL: Memory allocation of 10260 bytes
failed from 0x62757BCC, alignment 0
Pool: Processor  Free: 86116  Cause: Memory fragmentation
Alternate Pool: None  Free: 0  Cause: No Alternate pool
 -Process= "Virtual Exec", ipl= 0, pid= 176,  -Traceback= 0x613D6724
0x60333A30 0x6034C72C 0x62D7F6E4 0x62757BD4 0x62757D38 0x62732F78 0x62733050
0x6273314C 0x62732FF0 0x6273314C 0x62733CB4 0x6276C378 0x627773BC 0x62763730
0x6275D96C
.Dec 17 01:20:21.375: %NBAR-2-NOMEMORY: No memory available for StILE
lmalloc,  -Traceback= 0x613D6724 0x62757BFC 0x62757D38 0x62732F78 0x62733050
0x6273314C 0x62732FF0 0x6273314C 0x62733CB4 0x6276C378 0x627773BC 0x62763730
0x6275D96C 0x60FA0064 0x60FA0E64 0x614271FC

Thanks
Shingei
___
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[OSL | CCIE_Voice] Vol2 Lab 8 Question:Create Trunk or Gateway?

[ShinGei Yong]

Hi,

The question request that the call from HQ to BR2 must utilize CUBE and vice
versa.
PG given the solution by using a Trunk(non-gk controlled),but what i've done
was
configured the CUBE as a H323 GW in CUCM, and pointed the RP to this GW
which
achieved the same result.

>From Cisco doc "CUBE and CUCM configuration",it stated:

With a GK,configure a H225 Trunk(GK Controlled)toward to CUBE.
Without a GK,configure the CUBE as a H323 GW.

Is there any discrepancy or difference between the option in PG and
from Cisco doc?

Thanks
Shingei
___
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Re: [OSL | CCIE_Voice] Call forward to UC failed

[ShinGei Yong]

Hi Kobel, Peter,

Something that i'm not aware of until the very useful explanation by Kobel.
Thanks for the value information. My understanding was VM Pilot
configuration
should contain the VM ports only.

Just tested, got it working finally.

Really, experience make the difference.
Thanks
Shingei



2010/12/8 "Farkas Péter" 

> VM Pilot should have a CSS that can reach the hunt pilot for VM ports.
>
> try add PT-UC-VMPILOT to CSS-UC-VMPORT
>
> Peter
> - Original Message -
> From: ShinGei Yong 
> Date: Wednesday, December 8, 2010 12:02 pm
> Subject: Re: [OSL | CCIE_Voice] Call forward to UC failed
> To: Miron Kobelski , "ccie_voice@onlinestudylist.com"
> 
>
>
> > Hi Kobel,
> >
> >  Not sure i'm understand your question correctly.
> >  But i believe the VM Pilot information already stated, it used
> CSS-UC-VMPORT
> >  (PT-VMPORT,INT-DN)
> >  as below.
> >
> >  Shingei
> >
> >
> >  On Wed, Dec 8, 2010 at 6:48 PM, Miron Kobelski 
> wrote:
> >
> >  > OK, but it is not assigned to the VM pilot?
> >  >
> >  >
> >  >
> >  >
> >  > 2010/12/8 ShinGei Yong 
> >  >
> >  >> Hi Kobel,
> >  >>
> >  >> i believed is already stated there, the CSS-INT-DN which comprised (
> >  >> INT-DN and UC-VMPILOT) as well
> >  >>
> >  >>
> >  >> Shingei
> >  >>
> >  >>
> >  >> On Wed, Dec 8, 2010 at 6:42 PM, Miron Kobelski  >wrote:
> >  >>
> >  >>> It seems you don't have the PT-UC-VMPILOT partition in the CSS used
> for
> >  >>> forwarding - this might be your issue.
> >  >>>
> >  >>> regards
> >  >>> kobel
> >  >>>
> >  >>>
> >  >>> On Wed, Dec 8, 2010 at 11:39, ShinGei Yong  >wrote:
> >  >>>
> >  >>>> Hi Kobel,
> >  >>>>
> >  >>>> Below is the setup:
> >  >>>>
> >  >>>> 5600/PT-UC-VMPILOT
> >  >>>> 5601/PT-UC-VMPORT
> >  >>>> 5602/PT-UC-VMPORT
> >  >>>>
> >  >>>> 5001/PT-INT-DN  (Phone B)
> >  >>>> 5002/PT-INT-DN  (Phone A)
> >  >>>>
> >  >>>> CSS-INT-DN (PT-INT-DN,PT-UC-VMPILOT)
> >  >>>> CSS-UC-VMPORT (PT-VMPORT,PT-INT-DN)
> >  >>>>
> >  >>>> Voicemail Pilot Information
> >  >>>> Pilot Number: 5600
> >  >>>> CSS: CSS-UC-VMPORT
> >  >>>>
> >  >>>> Phone A Line Setting:
> >  >>>> CFA (Checked)  CSS-INT-DN
> >  >>>> CFB (Checked)  CSS-INT-DN
> >  >>>> CFNA (Checked)CSS-INT-DN
> >  >>>>
> >  >>>> Both phone A & B able to dial VM directly without issue.
> >  >>>>
> >  >>>> Anything goes wrong?
> >  >>>>
> >  >>>> Shingei
> >  >>>>
> >  >>>>
> >  >>>> On Wed, Dec 8, 2010 at 5:59 PM, Miron Kobelski <
> findko...@gmail.com>wrote:
> >  >>>>
> >  >>>>> Hi,
> >  >>>>>
> >  >>>>> If you entered forward destination as a number, you need to also
> >  >>>>> configure proper forwarding CSS.
> >  >>>>> if you configured forwarding for the line using the VM checkbox,
> you
> >  >>>>> also need to have proper CSS in VM pilot.
> >  >>>>>
> >  >>>>> HTH
> >  >>>>> kobel
> >  >>>>>
> >  >>>>>
> >  >>>>> 2010/12/8 ShinGei Yong 
> >  >>>>>
> >  >>>>> Hi,
> >  >>>>>>
> >  >>>>>> Yes, both the cfna and cfb has already checked on line setting,
> and
> >  >>>>>> CSS that contain the internal DN and VM pilot did assigned also.
> >  >>>>>>
> >  >>>>>> It seem to be a CSS issue but can't figure out the cause.
> >  >>>>>>
> >  >>>>>> I tried monitor the UC port status, there is no any call
> answering. So
> >  >>>>>> the call just stuck some where in UCM
> >  >>>>>>
> >  >>>>>> Shingei
> >  >&

Re: [OSL | CCIE_Voice] Call forward to UC failed

[ShinGei Yong]

Hi Kobel,

Not sure i'm understand your question correctly.
But i believe the VM Pilot information already stated, it used CSS-UC-VMPORT
(PT-VMPORT,INT-DN)
as below.

Shingei


On Wed, Dec 8, 2010 at 6:48 PM, Miron Kobelski  wrote:

> OK, but it is not assigned to the VM pilot?
>
>
>
>
> 2010/12/8 ShinGei Yong 
>
>> Hi Kobel,
>>
>> i believed is already stated there, the CSS-INT-DN which comprised (
>> INT-DN and UC-VMPILOT) as well
>>
>>
>> Shingei
>>
>>
>> On Wed, Dec 8, 2010 at 6:42 PM, Miron Kobelski wrote:
>>
>>> It seems you don't have the PT-UC-VMPILOT partition in the CSS used for
>>> forwarding - this might be your issue.
>>>
>>> regards
>>> kobel
>>>
>>>
>>> On Wed, Dec 8, 2010 at 11:39, ShinGei Yong wrote:
>>>
>>>> Hi Kobel,
>>>>
>>>> Below is the setup:
>>>>
>>>> 5600/PT-UC-VMPILOT
>>>> 5601/PT-UC-VMPORT
>>>> 5602/PT-UC-VMPORT
>>>>
>>>> 5001/PT-INT-DN  (Phone B)
>>>> 5002/PT-INT-DN  (Phone A)
>>>>
>>>> CSS-INT-DN (PT-INT-DN,PT-UC-VMPILOT)
>>>> CSS-UC-VMPORT (PT-VMPORT,PT-INT-DN)
>>>>
>>>> Voicemail Pilot Information
>>>> Pilot Number: 5600
>>>> CSS: CSS-UC-VMPORT
>>>>
>>>> Phone A Line Setting:
>>>> CFA (Checked)  CSS-INT-DN
>>>> CFB (Checked)  CSS-INT-DN
>>>> CFNA (Checked)CSS-INT-DN
>>>>
>>>> Both phone A & B able to dial VM directly without issue.
>>>>
>>>> Anything goes wrong?
>>>>
>>>> Shingei
>>>>
>>>>
>>>> On Wed, Dec 8, 2010 at 5:59 PM, Miron Kobelski wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> If you entered forward destination as a number, you need to also
>>>>> configure proper forwarding CSS.
>>>>> if you configured forwarding for the line using the VM checkbox, you
>>>>> also need to have proper CSS in VM pilot.
>>>>>
>>>>> HTH
>>>>> kobel
>>>>>
>>>>>
>>>>> 2010/12/8 ShinGei Yong 
>>>>>
>>>>> Hi,
>>>>>>
>>>>>> Yes, both the cfna and cfb has already checked on line setting, and
>>>>>> CSS that contain the internal DN and VM pilot did assigned also.
>>>>>>
>>>>>> It seem to be a CSS issue but can't figure out the cause.
>>>>>>
>>>>>> I tried monitor the UC port status, there is no any call answering. So
>>>>>> the call just stuck some where in UCM
>>>>>>
>>>>>> Shingei
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Dec 8, 2010 at 5:19 PM, mark.f.bunch 
>>>>>> wrote:
>>>>>>
>>>>>>> Have you also configured the Calling Search Space on the line for CFA
>>>>>>> and CFNA?
>>>>>>>
>>>>>>>
>>>>>>> On 08/12/2010, at 8:12 PM, Stanislav Braichuk <
>>>>>>> stanislav.braic...@gmail.com> wrote:
>>>>>>>
>>>>>>> Are you set checkbox on cfna and cfb in line configuration?
>>>>>>>
>>>>>>> 2010 12 8 10:44 пользователь "ShinGei Yong" <
>>>>>>> shingei.y...@gmail.com> написал:
>>>>>>> > Hi,
>>>>>>> > I'm facing an issue which is either cfna or cfb failed on phone A
>>>>>>> when
>>>>>>> > caller B call to caller A.
>>>>>>> > Both the phone A and B able to access to their voicemail box in UC
>>>>>>> by
>>>>>>> > pressing the
>>>>>>> > voicemail button and enter correct pin.
>>>>>>> >
>>>>>>> > Both the phone is able to dial the Pilot number directly without
>>>>>>> issue.
>>>>>>> > UC is integrate with UCM with SCCP.
>>>>>>> >
>>>>>>> > Am i miss out any setting?
>>>>>>> >
>>>>>>> > Shingei
>>>>>>>
>>>>>>> ___
>>>>>>> For more information regarding industry leading CCIE Lab training,
>>>>>>> please visit <http://www.ipexpert.com>www.ipexpert.com
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> ___
>>>>>> For more information regarding industry leading CCIE Lab training,
>>>>>> please visit www.ipexpert.com
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Call forward to UC failed

[ShinGei Yong]

Hi Kobel,

i believed is already stated there, the CSS-INT-DN which comprised ( INT-DN
and UC-VMPILOT) as well


Shingei

On Wed, Dec 8, 2010 at 6:42 PM, Miron Kobelski  wrote:

> It seems you don't have the PT-UC-VMPILOT partition in the CSS used for
> forwarding - this might be your issue.
>
> regards
> kobel
>
>
> On Wed, Dec 8, 2010 at 11:39, ShinGei Yong  wrote:
>
>> Hi Kobel,
>>
>> Below is the setup:
>>
>> 5600/PT-UC-VMPILOT
>> 5601/PT-UC-VMPORT
>> 5602/PT-UC-VMPORT
>>
>> 5001/PT-INT-DN  (Phone B)
>> 5002/PT-INT-DN  (Phone A)
>>
>> CSS-INT-DN (PT-INT-DN,PT-UC-VMPILOT)
>> CSS-UC-VMPORT (PT-VMPORT,PT-INT-DN)
>>
>> Voicemail Pilot Information
>> Pilot Number: 5600
>> CSS: CSS-UC-VMPORT
>>
>> Phone A Line Setting:
>> CFA (Checked)  CSS-INT-DN
>> CFB (Checked)  CSS-INT-DN
>> CFNA (Checked)CSS-INT-DN
>>
>> Both phone A & B able to dial VM directly without issue.
>>
>> Anything goes wrong?
>>
>> Shingei
>>
>>
>> On Wed, Dec 8, 2010 at 5:59 PM, Miron Kobelski wrote:
>>
>>> Hi,
>>>
>>> If you entered forward destination as a number, you need to also
>>> configure proper forwarding CSS.
>>> if you configured forwarding for the line using the VM checkbox, you also
>>> need to have proper CSS in VM pilot.
>>>
>>> HTH
>>> kobel
>>>
>>>
>>> 2010/12/8 ShinGei Yong 
>>>
>>> Hi,
>>>>
>>>> Yes, both the cfna and cfb has already checked on line setting, and CSS
>>>> that contain the internal DN and VM pilot did assigned also.
>>>>
>>>> It seem to be a CSS issue but can't figure out the cause.
>>>>
>>>> I tried monitor the UC port status, there is no any call answering. So
>>>> the call just stuck some where in UCM
>>>>
>>>> Shingei
>>>>
>>>>
>>>>
>>>> On Wed, Dec 8, 2010 at 5:19 PM, mark.f.bunch wrote:
>>>>
>>>>> Have you also configured the Calling Search Space on the line for CFA
>>>>> and CFNA?
>>>>>
>>>>>
>>>>> On 08/12/2010, at 8:12 PM, Stanislav Braichuk <
>>>>> stanislav.braic...@gmail.com> wrote:
>>>>>
>>>>> Are you set checkbox on cfna and cfb in line configuration?
>>>>>
>>>>> 2010 12 8 10:44 пользователь "ShinGei Yong" < 
>>>>> shingei.y...@gmail.com> написал:
>>>>> > Hi,
>>>>> > I'm facing an issue which is either cfna or cfb failed on phone A
>>>>> when
>>>>> > caller B call to caller A.
>>>>> > Both the phone A and B able to access to their voicemail box in UC by
>>>>> > pressing the
>>>>> > voicemail button and enter correct pin.
>>>>> >
>>>>> > Both the phone is able to dial the Pilot number directly without
>>>>> issue.
>>>>> > UC is integrate with UCM with SCCP.
>>>>> >
>>>>> > Am i miss out any setting?
>>>>> >
>>>>> > Shingei
>>>>>
>>>>> ___
>>>>> For more information regarding industry leading CCIE Lab training,
>>>>> please visit <http://www.ipexpert.com>www.ipexpert.com
>>>>>
>>>>>
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
>>>> please visit www.ipexpert.com
>>>>
>>>>
>>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Call forward to UC failed

[ShinGei Yong]

Hi Kobel,

Below is the setup:

5600/PT-UC-VMPILOT
5601/PT-UC-VMPORT
5602/PT-UC-VMPORT

5001/PT-INT-DN  (Phone B)
5002/PT-INT-DN  (Phone A)

CSS-INT-DN (PT-INT-DN,PT-UC-VMPILOT)
CSS-UC-VMPORT (PT-VMPORT,PT-INT-DN)

Voicemail Pilot Information
Pilot Number: 5600
CSS: CSS-UC-VMPORT

Phone A Line Setting:
CFA (Checked)  CSS-INT-DN
CFB (Checked)  CSS-INT-DN
CFNA (Checked)CSS-INT-DN

Both phone A & B able to dial VM directly without issue.

Anything goes wrong?

Shingei

On Wed, Dec 8, 2010 at 5:59 PM, Miron Kobelski  wrote:

> Hi,
>
> If you entered forward destination as a number, you need to also configure
> proper forwarding CSS.
> if you configured forwarding for the line using the VM checkbox, you also
> need to have proper CSS in VM pilot.
>
> HTH
> kobel
>
>
> 2010/12/8 ShinGei Yong 
>
> Hi,
>>
>> Yes, both the cfna and cfb has already checked on line setting, and CSS
>> that contain the internal DN and VM pilot did assigned also.
>>
>> It seem to be a CSS issue but can't figure out the cause.
>>
>> I tried monitor the UC port status, there is no any call answering. So the
>> call just stuck some where in UCM
>>
>> Shingei
>>
>>
>>
>> On Wed, Dec 8, 2010 at 5:19 PM, mark.f.bunch wrote:
>>
>>> Have you also configured the Calling Search Space on the line for CFA and
>>> CFNA?
>>>
>>>
>>> On 08/12/2010, at 8:12 PM, Stanislav Braichuk <
>>> stanislav.braic...@gmail.com> wrote:
>>>
>>> Are you set checkbox on cfna and cfb in line configuration?
>>>
>>> 2010 12 8 10:44 пользователь "ShinGei Yong" < 
>>> shingei.y...@gmail.com> написал:
>>> > Hi,
>>> > I'm facing an issue which is either cfna or cfb failed on phone A when
>>> > caller B call to caller A.
>>> > Both the phone A and B able to access to their voicemail box in UC by
>>> > pressing the
>>> > voicemail button and enter correct pin.
>>> >
>>> > Both the phone is able to dial the Pilot number directly without issue.
>>> > UC is integrate with UCM with SCCP.
>>> >
>>> > Am i miss out any setting?
>>> >
>>> > Shingei
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit <http://www.ipexpert.com>www.ipexpert.com
>>>
>>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Call forward to UC failed

[ShinGei Yong]

Hi,

Yes, both the cfna and cfb has already checked on line setting, and CSS that
contain the internal DN and VM pilot did assigned also.

It seem to be a CSS issue but can't figure out the cause.

I tried monitor the UC port status, there is no any call answering. So the
call just stuck some where in UCM

Shingei


On Wed, Dec 8, 2010 at 5:19 PM, mark.f.bunch  wrote:

> Have you also configured the Calling Search Space on the line for CFA and
> CFNA?
>
>
> On 08/12/2010, at 8:12 PM, Stanislav Braichuk <
> stanislav.braic...@gmail.com> wrote:
>
> Are you set checkbox on cfna and cfb in line configuration?
>
> 2010 12 8 10:44 пользователь "ShinGei Yong" < 
> shingei.y...@gmail.com> написал:
> > Hi,
> > I'm facing an issue which is either cfna or cfb failed on phone A when
> > caller B call to caller A.
> > Both the phone A and B able to access to their voicemail box in UC by
> > pressing the
> > voicemail button and enter correct pin.
> >
> > Both the phone is able to dial the Pilot number directly without issue.
> > UC is integrate with UCM with SCCP.
> >
> > Am i miss out any setting?
> >
> > Shingei
>
> ___
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> visit <http://www.ipexpert.com>www.ipexpert.com
>
>
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[OSL | CCIE_Voice] Call forward to UC failed

[ShinGei Yong]

Hi,
I'm facing an issue which is either cfna or cfb failed on phone A when
caller B call to caller A.
Both the phone A and B able to access to their voicemail box in UC by
pressing the
voicemail button and enter correct pin.

Both the phone is able to dial the Pilot number directly without issue.
UC is integrate with UCM with SCCP.

Am i miss out any setting?

Shingei
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Re: [OSL | CCIE_Voice] Adding subscriber problem - need ideas

[ShinGei Yong]

Hi Wael,

What version of UCM are you installing? CUCM or CUCMBE?

Shingei.

On Fri, Nov 26, 2010 at 7:44 AM, Wael Agina  wrote:

> Dears,
>
>   I added it already on pub using its IP address.
> Also I activated most needed services.
> Do you ask for specific service to be started ? Almost most services added
> and Pub is working fine and many phones connected and working.
>
> Regards,
> Wael Agina
>
>
> On Fri, Nov 26, 2010 at 2:41 AM, CCIE Voice  wrote:
>
>> Make sure you add the subscriber to the publisher list of call manager
>> servers.
>>
>> --
>>
>>
>> On Nov 25, 2010, at 14:25, Wael Agina  wrote:
>>
>> Dear All,
>>
>>I am implemnting new UC project at one customer location.
>> I installed the first node - publisher - succefully.
>> Now whenevr i add the 2nd node - subscriber - it tried to test
>> connectivity to first node but always fails and installion terminated.
>> There is network connectivity between the two nodes and from publisher i
>> can ping the new node - subscriber.
>> Also I reset the security password on the publisher [using Set *password*user
>> *security on CLI]*
>>
>> Any ideas as the connectivity test always fails ?
>>
>> --
>>
>> Thanks and Best Regards,
>> Wael Agina
>>
>> ___
>>
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
>
> --
>
> Thanks and Best Regards,
> Wael Agina
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] Intercluster Question.

[ShinGei Yong]

Hi,

Something just in my mind regarding to the intercluster design.

Scenario A: Two different clusters, cluster A with UCM and H323 GW, and
cluster B with UCME & H323 GW.
Without using CUBE or GK, can i configure ICT on UCM (without gk-controlled)
and point to the UCME address for call to cluster B
and configure voip dial-peer on UCME that point to UCM for call to cluster
A?

Or is this ICT setup valid?

Shingei
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Re: [OSL | CCIE_Voice] Volume 2 Lab 7, Task 4.4

[ShinGei Yong]

Hi,

You guys suggestion made the trick. CUC just perform the "silence" transfer
to the ext/dn after the last digit press on calling phone and the
destination phone ring.
I'm wonder is it possible to configure in such a way that the system
response with " Wait while i transfer your call" after the last digit enter.
Currently,
if i do a transfer to ext/dn that exist on CUC, this's the behavior. When i
do a transfer to ext/dn outside the CUC, it just "silence" without any
system response
and destination phone ring.

I'm looking at System Call Handler --> Opening Greeting --> Greeting,
Standard. No option for the system response.
Looking at the Transfer rules, Standard, there is a transfer action to allow
system play "Wait while i transfer your call",but that setting is mean for
System Transfer.

How do i achieve the above requirement?

Shingei



On Thu, Nov 25, 2010 at 2:34 AM, Jones, Brett wrote:

> Hi Shingei,
>
> You need to disable the last rule in the "restriction tables"
>
>
> Message: 5
> Date: Wed, 24 Nov 2010 19:39:51 +0800
> From: ShinGei Yong 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Volume 2 Lab 7, Task 4.4
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> I'm currently facing the problem in which UC response with "You cannot be
> transfer to this number..." during the opening greeting.
> Steps done according to the PG, which check the option "Allow Transfer to
> Number not associated with user or call handler", but still no luck.
>
> Some test applied was, tried calling VM by pressing the message button,
> press # while asking for login PIN. During Opening Greeting, dial the user
> DN with no mailbox on UC.
>
> Did i miss out anything?
>
> Shingei
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> DISCLAIMER:
> This correspondence may contain information which is confidential or
> proprietary or both.  Any dissemination, distribution, copying or use of
> this communication without prior permission of the sender is strictly
> prohibited. If you are not the intended recipient you may not disclose, copy
> or use this information.  If you have received this message in error, please
> contact the sender to discuss its return or destruction.
>
> The contents, comments and views contained or expressed within this
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> subsidiaries, affiliates, associates or sister companies and are not
> intended to create legal relations with the recipient.
>
> Redstone may monitor email traffic data and also the content of email for
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>
> If you would like to know more about Redstone Converged Solutions, visit us
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[OSL | CCIE_Voice] Volume 2 Lab 7, Task 4.4

[ShinGei Yong]

Hi,

I'm currently facing the problem in which UC response with "You cannot be
transfer to this number..." during the opening greeting.
Steps done according to the PG, which check the option "Allow Transfer to
Number not associated with user or call handler", but still no luck.

Some test applied was, tried calling VM by pressing the message button,
press # while asking for login PIN. During Opening Greeting, dial the user
DN with no mailbox on UC.

Did i miss out anything?

Shingei
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Re: [OSL | CCIE_Voice] SNR - Remote in Use

[ShinGei Yong]

Hi Brett,

What about pressing the line button that turned into red manually?
I believe that will change the state to "Remote in Use".

This question been posted before, and ppls claimed this is the expected
behavior.

Shingei.


On Mon, Nov 22, 2010 at 7:41 PM, Jones, Brett wrote:

> Hi All,
>
> Can someone please explain to me what I'm doing wrong, when configuring
> Signal Number Reach I can get everything to work. However when the call is
> answered on the "mobile phone" my desk phone's line button is red and the
> icon does change but it never states "In Remote Use".
>
> Thanks in advance.
> Brett
>
> DISCLAIMER:
> This correspondence may contain information which is confidential or
> proprietary or both.  Any dissemination, distribution, copying or use of
> this communication without prior permission of the sender is strictly
> prohibited. If you are not the intended recipient you may not disclose, copy
> or use this information.  If you have received this message in error, please
> contact the sender to discuss its return or destruction.
>
> The contents, comments and views contained or expressed within this
> correspondence do not necessarily reflect those of Redstone, its
> subsidiaries, affiliates, associates or sister companies and are not
> intended to create legal relations with the recipient.
>
> Redstone may monitor email traffic data and also the content of email for
> the purposes of security and staff training.
>
> If you would like to know more about Redstone Converged Solutions, visit us
> on the web at www.redstoneconverged.co.uk or contact our Head Office on
> 0845 20 1.
>
> Redstone Converged Solutions Limited
> Registered in England & Wales with Company Number: 02027207
> Registered Office: Kirtlington Business Centre, Slade Farm, Kirtlington,
> Kidlington, Oxfordshire, OX5 3JA
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Re: [OSL | CCIE_Voice] + Dialing issue

[ShinGei Yong]

Hi Shrini,

Can you ensure that the UCM mgcp config pump into the gateway correctly? Do
a " sh ccm-manager" to verify.

Shingei

On Fri, Nov 12, 2010 at 2:01 AM, Shrini  wrote:

>  Hi Iftikhar,
>
> In the translation pattern 9.1[2-9]XX[2-9]XX -->
>
> under calling party transformation select the check box use calling part's
> external phone mask.
> Calling party transform mask - XXX
> Calling /called party number type : National
> Calling/Called party numbering plan : ISDN
>
> In Gateway:
>
> Under Incoming calling part settings:
>
> Change the prefix to + from Default for National numbers.
>
> You will get +1XX as calling party displayed on phone and missed
> calls
>
> Thx
> Shrini
>
>
>
> On 11/10/2010 11:55 PM, Mujaddid Ahmed wrote:
>
>  Hi,
>
> Please confirm what can be the reason of globalization not working with
> cisco 7965 phones.
> I globalized it on CUCM gateway configuration by adding appropriate prefix
> on type of calls.
>
> did, no mgcp / mgcp on gateway and reset the gateway from cucm as well.
>
> call lands on phone and still shows only 7 digit calling number in both
> ringing and missed call.
>
> nothing has been set so far in Device pool or on device to effect the
> globalization done on Gateway.
>
> Is there any serivce parameter or any command on router that can prevent
> the globalize display on phone?
>
> Please help, as i wasted lot of time in troubleshooting this particular
> issue in a very time critical environment.
>
> Regards,
> Iftikhar Ahmed
>
>
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> visit www.ipexpert.com
>
>
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>
>
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[OSL | CCIE_Voice] Vol2 Lab 5 Question 2.3 Call Routing

[ShinGei Yong]

Hi,

According to the WB question no# stated above, PSTN will send the calling
(ANI) number as 02059432785,
the PG on page 63 described about this as well, but in fact, the ANI only
sent in as 2059432785 while call to HQ.
I capture the debug q931 as below:

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, '2059434001'
Plan:ISDN, Type:National
*Nov  9 11:35:14.079: ISDN Se0/1/0:23 Q931: TX -> CALL_PROC pd = 8  callref
= 0x808C
Channel ID i = 0xA98381
Exclusive, Channel 1
*Nov  9 11:35:14.083: ISDN Se0/1/0:23 Q931: TX -> ALERTING pd =

I'm running on my own gear, the PSTN config was downloaded from download
section of member sites.
Something missing in the PSTN config or my stuff went hair wire?

Thanks
Shingei.
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Re: [OSL | CCIE_Voice] CUE Integration Commands

[ShinGei Yong]

Hi,

Here you go...
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml

Ensure you have correct license file for CUE.

TIA
Shingei

On Fri, Oct 22, 2010 at 11:09 PM, ccieiwillb  wrote:

> Hi Experts,
>
> I was wondering if anyone happened to have a link to the commands to
> manually integrate CUE with CUCM?  I have seen the document before off of
> the Cisco website but I am not able to find it any longer.  I am having a
> strange issue whenever I launch the initiaulization webpage so I am trying
> to see if I can configure the CTI ports and jtapi manually to integrate with
> CUCM.  Any help is greatyly appreciated.
>
> Thanks
> ccieiwillb
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2

[ShinGei Yong]

Hi Mazin,

Thanks. I'm totally agreed with your calculation, but just to emphasize that
how you treat the remaining bandwidth in % calculation.
As long the *remaining* is *equally* shared, then i believe we got the
point.

And you got the point. :)

TIA
Shingei

On Fri, Oct 22, 2010 at 4:10 AM, Mazin Ahsan  wrote:

> Thanks Shingei,
>
> Lets say that 4:4 ratio shares the 50% bandwidth among the two queues.
> According to the task we also have to have 20% bandwidth with priority-queue
> .
>
> The 3750 have two input queues. If we are to give one queue priority with
> 20% bandwidth according to the question. Then we have to share the rest
> between the two queues equally, we are giving 4:4 ratio to the queues.
>
> Lets say we have Fasthethernet link
>
> Queue 1 : 20% Prioritized.   === 20Mbps
> Queue 1 : 50% Bandwidth of 80 Mbps=== 40Mbps
> Queue 2 : 50% Bandwidth of 80 Mbps=== 40Mbps
>
> Queue 1 will be  prioritized queue and will get priority upto 20% bandwidth
> after which the remaining of the traffic will be shared with the ratio of
> 4:4 . Meaning 50% to each queue.
>
>
> HTH
> Mazin
>
> -
> -
>
>
>
> Hi Greg,
>
> I think Amr is referring to the WB question that require the remaining
> bandwidth of 80% (after deducted the 20% for PQ on queue1) to be equally
> shared between Q1 and Q2.
> In this case, neither Q1 4/(4+4) nor Q2 4/(4+4) will get 50%. Q1 and Q2
> will
> get 40% each, i believe this achieve what the question asked which equally
> shared between the remaining bandwidth.
>
> If configured Q1 5/(5+5) and Q2 5/(5+5) and the result multiply by 80%, you
> will still get the 40% each queue, so mean it doesn't a matter what value u
> configured, as long both of them are same.
>
> For me is quite straight, if the question ask to disable the PQ, then
> n/(n+y) or y/(n+y) the valuable n and y can be ANY value that add up total
> to be 100, which mean it can be (25+75), (40+60)...etc.
> But if the the question asking to turn on PQ and equally shared the
> remaining bandwidth, i think the same value of n and y is applied.
>
> Please correct me if wrong.
> IMHO.
> Shingei.
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2

[ShinGei Yong]

Hi Greg,

I think Amr is referring to the WB question that require the remaining
bandwidth of 80% (after deducted the 20% for PQ on queue1) to be equally
shared between Q1 and Q2.
In this case, neither Q1 4/(4+4) nor Q2 4/(4+4) will get 50%. Q1 and Q2 will
get 40% each, i believe this achieve what the question asked which equally
shared between the remaining bandwidth.

If configured Q1 5/(5+5) and Q2 5/(5+5) and the result multiply by 80%, you
will still get the 40% each queue, so mean it doesn't a matter what value u
configured, as long both of them are same.

For me is quite straight, if the question ask to disable the PQ, then
n/(n+y) or y/(n+y) the valuable n and y can be ANY value that add up total
to be 100, which mean it can be (25+75), (40+60)...etc.
But if the the question asking to turn on PQ and equally shared the
remaining bandwidth, i think the same value of n and y is applied.

Please correct me if wrong.
IMHO.
Shingei.

On Thu, Oct 21, 2010 at 12:21 AM, Pulos, Greg  wrote:

>  Just to be clear….
>
>
>
> Your question was….
>
>
>
> “So you mean i can specify any number other than 4 but keep sure that both
> of them configured with the same vlaue, is that correct.”
>
>
>
> The responder stated “Yep J”
>
>
>
> To which, I corrected……..”No, both values do not have to be the same.” And
> if you change them to be the same, say from 4 + 4, to 5 + 5, your bandwidth
> will not  be 50%, therefore the answer to “So you mean i can specify any
> number other than 4 but keep sure that both of them configured with the same
> vlaue, is that correct.”, is no, not necessarily; it depends on the amount
> of bandwidth you want serviced.
>
>
>
> Please see the following link for more info on the ‘mls qos srr-queue input
> bandwidth’ command.
>
>
>
>
> http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_18_se/command/reference/cli1.html#wp2128913
>
>
>
>
>
> greg
>
>
>
>
>
> *From:* Amr Sherif [mailto:miroale...@hotmail.com]
> *Sent:* Wednesday, October 20, 2010 12:14 PM
> *To:* Pulos, Greg; findko...@gmail.com
>
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* RE: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
>
>
>
>
>
> But the question was specifying that both the two queues has to be equally
> shared, right .So that's why in this situation it's used for both values
>
> Best regards,
>
>  Amr Sherif
> Senior Network Voice Engineer
> CCNA,CCNP,CCVP and CCIE Voice Written *(Certified)*
> CCIE Voice Lab *(In Progress)*
>
>
>  --
>
> From: gpu...@doc.gov
> To: findko...@gmail.com; miroale...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
> Date: Wed, 20 Oct 2010 11:55:16 -0400
> Subject: RE: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
>
> I disagree…
>
>
>
> These 2 values DO NOT have to be the same.
>
>
>
> Thank you.
>
>
>
> *greg*
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Miron Kobelski
> *Sent:* Wednesday, October 20, 2010 11:51 AM
> *To:* Amr Sherif
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
>
>
>
> yep :)
> --
> Sent from my mobile device.
>
> On 20 Oct 2010 17:48, "Amr Sherif"  wrote:
>
>
> So you mean i can specify any number other than 4 but keep sure that both
> of them configured with the same vlaue,is that correct.
>
>
>
>
> Best regards,
>
>
> Amr Sherif
> Senior Network Voice Engineer
> CCNA,CCNP,CCVP and CCIE Voice Written (C...
>
> Date: Wed, 20 Oct 2010 17:27:20 +0200
> Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
> From: findko...@gmail.com
> To: miroale...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
>
>
> those two 4s are relative to each other. so each of 2 queues will get
> 4/(4+4) 50% after servicing ...
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Volume 2 Lab 3 Task 2.4, Calling Party Transformation

[ShinGei Yong]

Hi All,

More information submitted below:

CIPC (HQPH2, x5002, EPNM:2123945002, SIP)
IPBlue (HQPH5, x5005, EPNM: 2123945005, SCCP)

Globalization Configured:

PH5 call PH2
Main LCD Display: +12123945005, Missed & Received Entry: +12123945005
Vice versa for PH2 call PH5

Globalization Configured + CngPTP Xform

PH5 call PH2
Main LCD Display: 5005, Missed & Received Entry: 5005 <-- Weird?
Vice versa for PH2 call PH5

TIA
Shingei.


On Tue, Oct 19, 2010 at 11:20 AM, ShinGei Yong wrote:

> Hi Prashant,
>
> Thanks for help
> The correct CngCSS has been assigned to the DP, that why the calling party
> E164 can be "localize" to 4 digit as mentioned earlier. The problem is, it
> does "localize" the missed and received entry which is not desired. The
> missed and received need to be in E164 form. I applied the the CngPTP to DP
> instead of individual device, but i don't think and has tested it's
> irrelevant.
>
> Any clue?
> TIA
> Shingei
>
>
> On Tue, Oct 19, 2010 at 10:50 AM, Prashant Patel <
> prashantpatel...@gmail.com> wrote:
>
>> Hi,
>>
>> Please check that you have correct Calling Party CSS on the phone. Also
>> make sure the phone calling party css only sees that partition in which you
>> have the transformations.
>>
>> HTH
>> Prashant
>>
>> On Mon, Oct 18, 2010 at 10:46 PM, ShinGei Yong wrote:
>>
>>> Hi all,
>>>
>>> I'm trying to configure the CngPTP to transform (locallize) the calling
>>> party number from E164 to 4 digit as per WB3, task2.4 stated.
>>> I managed to transform the calling display to 4 digit, but some how the
>>> missed and received call also displayed the 4 digit calling number
>>> instead of E164.
>>>
>>> Prior to Localization, the Globalization was work as expected, the
>>> calling & called number displayed in E164 format in either missed or
>>> received.
>>> Once i applied the CngPTP under DP, the calling number "localize" to 4
>>> digit, but it does "localize" the missed and received entry, which is not
>>> desired.
>>>
>>> I'm testing this question by using 1 CIPC and 1 IPBlue, and calling each
>>> other to verify.
>>>
>>> Am i miss out anything?
>>>
>>> TIA
>>> Shingei.
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>
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Re: [OSL | CCIE_Voice] Volume 2 Lab 3 Task 2.4, Calling Party Transformation

[ShinGei Yong]

Hi Prashant,

Thanks for help
The correct CngCSS has been assigned to the DP, that why the calling party
E164 can be "localize" to 4 digit as mentioned earlier. The problem is, it
does "localize" the missed and received entry which is not desired. The
missed and received need to be in E164 form. I applied the the CngPTP to DP
instead of individual device, but i don't think and has tested it's
irrelevant.

Any clue?
TIA
Shingei

On Tue, Oct 19, 2010 at 10:50 AM, Prashant Patel  wrote:

> Hi,
>
> Please check that you have correct Calling Party CSS on the phone. Also
> make sure the phone calling party css only sees that partition in which you
> have the transformations.
>
> HTH
> Prashant
>
> On Mon, Oct 18, 2010 at 10:46 PM, ShinGei Yong wrote:
>
>> Hi all,
>>
>> I'm trying to configure the CngPTP to transform (locallize) the calling
>> party number from E164 to 4 digit as per WB3, task2.4 stated.
>> I managed to transform the calling display to 4 digit, but some how the
>> missed and received call also displayed the 4 digit calling number
>> instead of E164.
>>
>> Prior to Localization, the Globalization was work as expected, the calling
>> & called number displayed in E164 format in either missed or received.
>> Once i applied the CngPTP under DP, the calling number "localize" to 4
>> digit, but it does "localize" the missed and received entry, which is not
>> desired.
>>
>> I'm testing this question by using 1 CIPC and 1 IPBlue, and calling each
>> other to verify.
>>
>> Am i miss out anything?
>>
>> TIA
>> Shingei.
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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[OSL | CCIE_Voice] Volume 2 Lab 3 Task 2.4, Calling Party Transformation

[ShinGei Yong]

Hi all,

I'm trying to configure the CngPTP to transform (locallize) the calling
party number from E164 to 4 digit as per WB3, task2.4 stated.
I managed to transform the calling display to 4 digit, but some how the
missed and received call also displayed the 4 digit calling number
instead of E164.

Prior to Localization, the Globalization was work as expected, the calling &
called number displayed in E164 format in either missed or received.
Once i applied the CngPTP under DP, the calling number "localize" to 4
digit, but it does "localize" the missed and received entry, which is not
desired.

I'm testing this question by using 1 CIPC and 1 IPBlue, and calling each
other to verify.

Am i miss out anything?

TIA
Shingei.
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[OSL | CCIE_Voice] Layer 2 Overhead brainstorm

[ShinGei Yong]

Hi all,

This is obviously an old question has been repeated N times, but varies &
varies answer anywhere.

According to the QoS SRND, page 1-15 stated:

* Multilink PPP (MLP) add 13bytes of layer 2 overhead.
* Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12 adds
8 bytes.

My first question is, is this MLP included the the FR overhead as well? So
MLP (9 bytes) + FR (4 bytes) = 13 bytes

According to Cisco web page: VoIP Per Call Bandwidth consumption, 6 bytes is
selected for MLP overhead, which one should follow?

Per Call Bandwidth
Consumption

Second question is, what is the layer 2 overhead for MLP w/ LFI?

According to Cisco End-to-End QoS Network Design book,under chapter 16, the
MLP listed here is 10 bytes, + 3 for MLP LFI, total would be 13 bytes for
MLP LFI.

Confuse?

Shingei
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Re: [OSL | CCIE_Voice] GK VIA zone

[shingei . yong]

Hi,

Finally able to figured out what caused this behavior.
Guys, just ignore this stupid question from me.

regards,
Shingei



On Aug 31, 2010 4:46pm, shingei.y...@gmail.com wrote:

Hi,


I've 3 routers which is HQ-RTR, BR1-RTR & BR2-RTR, The HQ-RTR configured  
as GK as well as CUBE.
When i do a "show gatekeeper endpoints", GK shown that CUBE, BR1-RTR and  
BR2-RTR registered correctly, but the "zone name" for the registered  
devices become VIA instead of their origin zone.


I referred to the student guide, the zone name should display what i've  
configured under GK configuration.



GK configuration:

interface Loopback0
ip address 10.10.110.1 255.255.255.255



interface Loopback1
ip address 10.10.110.10 255.255.255.255
h323-gateway voip interface
h323-gateway voip id VIA ipaddr 10.10.110.1 1719
h323-gateway voip h323-id CUBE
h323-gateway voip tech-prefix 1#




gatekeeper
zone local BR1 ipexpert.com 10.10.110.1 invia VIA outvia VIA
zone local BR2 ipexpert.com invia VIA outvia VIA
zone local VIA ipexpert.com
zone prefix BR1 1*
zone prefix BR2 3*
zone prefix VIA 5*
gw-type-prefix 1#* default-technology
no shutdown
###



Q-RTR#sh gatek end
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
--- - --- - -  -
10.10.110.2 1720 10.10.110.2 56834 VIA E164-ID: 1001
H323-ID: BR1
Voice Capacity Max.= Avail.= Current.= 0
10.10.110.3 1720 10.10.110.3 49815 VIA H323-ID: BR2-GW
E164-ID: 3001
Voice Capacity Max.= Avail.= Current.= 0
10.10.110.10 1720 10.10.110.10 53994 VIA H323-GW
E164-ID: 5001
H323-ID: CUBE
Voice Capacity Max.= Avail.= Current.= 0
Total number of active registrations = 3



Am i miss out something?
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[OSL | CCIE_Voice] GK VIA zone

[shingei . yong]

Hi,

I've 3 routers which is HQ-RTR, BR1-RTR & BR2-RTR, The HQ-RTR configured as  
GK as well as CUBE.
When i do a "show gatekeeper endpoints", GK shown that CUBE, BR1-RTR and  
BR2-RTR registered correctly, but the "zone name" for the registered  
devices become VIA instead of their origin zone.


I referred to the student guide, the zone name should display what i've  
configured under GK configuration.


GK configuration:

interface Loopback0
ip address 10.10.110.1 255.255.255.255

interface Loopback1
ip address 10.10.110.10 255.255.255.255
h323-gateway voip interface
h323-gateway voip id VIA ipaddr 10.10.110.1 1719
h323-gateway voip h323-id CUBE
h323-gateway voip tech-prefix 1#


gatekeeper
zone local BR1 ipexpert.com 10.10.110.1 invia VIA outvia VIA
zone local BR2 ipexpert.com invia VIA outvia VIA
zone local VIA ipexpert.com
zone prefix BR1 1*
zone prefix BR2 3*
zone prefix VIA 5*
gw-type-prefix 1#* default-technology
no shutdown
###

Q-RTR#sh gatek end
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
--- - --- - -  -
10.10.110.2 1720 10.10.110.2 56834 VIA 

Re: [OSL | CCIE_Voice] ip phone on layer 3 interface

[ShinGei Yong]

Hi Angel,

I'm trying to achieve the same setup which connect my 7961 to cisco 2801
fa0/1, but the phone never get register and showing "Configuring IP..."

I configure a  DHCP pool, from the debug i can see that the phone is sending
the DHCPD and the server did response and offered an IP address to it, but
the phone never receive.


PSTN-WAN(dhcp-config)#
Aug  2 21:45:59.367: DHCPD: Sending notification of DISCOVER:
Aug  2 21:45:59.367:   DHCPD: htype 1 chaddr 0019.e892.4a0c
Aug  2 21:45:59.367:   DHCPD: remote id 020a0a0a64020132
Aug  2 21:45:59.367:   DHCPD: circuit id 
Aug  2 21:45:59.367: DHCPD: DHCPDISCOVER received from client
0100.19e8.924a.0c on interface FastEthernet0/1.50.
Aug  2 21:45:59.367: DHCPD: Seeing if there is an internally specified pool
class:
Aug  2 21:45:59.367:   DHCPD: htype 1 chaddr 0019.e892.4a0c
Aug  2 21:45:59.367:   DHCPD: remote id 020a0a0a64020132
Aug  2 21:45:59.367:   DHCPD: circuit id 
Aug  2 21:47:39.383: DHCPD: Sending DHCPOFFER to client 0100.19e8.924a.0c
(10.10.100.12).
Aug  2 21:47:39.383: DHCPD: broadcasting BOOTREPLY to client 0019.e892.4a0c.

The fa0/1 configuration as below:
interface FastEthernet0/1
 no ip address
 speed 100
 full-duplex
!
interface FastEthernet0/1.50
 encapsulation dot1Q 50 native  //50 is the VVLAN//
 ip address 10.10.100.2 255.255.255.0

I did explicitly configure the vlan setting on the phone itself. Did i miss
out anything?

Thanks
Shingei

On Tue, Mar 16, 2010 at 4:36 PM, Angel Perez  wrote:

>
> Hi:
>
> Thanks for the response, working here now also, I had to set the voice vlan
> manualy on ip phone and then just create to subinterfaces on the fast 0/0
> interface with dot1q encapsulation
>
> Thanks
> --
> From: pav.c...@gmail.com
> To: gorr...@hotmail.com
> Subject: Re: [OSL | CCIE_Voice] ip phone on layer 3 interface
> Date: Mon, 15 Mar 2010 21:17:51 -0500
>
> CC: ccie_voice@onlinestudylist.com
>
> I have tried it on 3600. It works
>
> Sent from my phone
>
> On Mar 15, 2010, at 3:52 AM, Angel Perez  wrote:
>
>  Hello:
>
> Anybody has tested this?
>
> Thanks
>
> --
> From: gorr...@hotmail.com
> To: ccie_voice@onlinestudylist.com
> Date: Fri, 12 Mar 2010 17:57:35 +
> Subject: [OSL | CCIE_Voice] ip phone on layer 3 interface
>
> Hello:
>
> I want to connect an ip phone to a 2811 fast ether 0/0 interface (pstn
> router) this way I wouldn't need a switch for the pstn phone
> A xcable is needed but I'm not sure if a layer 3 interface is l2 switching
> capable
>
> Any suggestion?
>
> Thanks
>
> --
> ¿Sabes que la Videollamada de Messenger es GRATIS 
> ¡Descúbrela!
> --
> Hotmail: Powerful Free email with security by Microsoft. Get it 
> now.
>
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> visit www.ipexpert.com
>
>
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>
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>
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[OSL | CCIE_Voice] Vol1 Lab 11A UnityConnection with AA

[ShinGei Yong]

Hi,

I'm trying to create a call handler with lab 11A requirement. Everytime when
i made the call to AA no# 5000, the Greeting will broadcast that "Sorry,
<> is not available."
Is this the normal system behavior because the "Caller Hear" option has
selected to System default greeting?

I tried recorded the greeting and select the (Caller Hear --> My Personal
Recording)but result still the same.

Any idea what i've miss out.

Thanks
Shingei
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[OSL | CCIE_Voice] Real time tracing of IP phone activity

[ShinGei Yong]

Hi,

I'm trying to do a real time tracing of ip phone activity, for example; when
the phone goes off hock, the line seized, CUCM sending the signaling and
tone etc...
I'm using RTMT --> Real time tracing --> View real time data to do so but
unsuccessful.

Anyone know which services should be select when using RTMT for such action?

Thanks
shingei.
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Re: [OSL | CCIE_Voice] BACD prompts

[ShinGei Yong]

Hi,

You may refer to this old post:

http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg06974.html


hope that help

On Wed, Jun 9, 2010 at 6:41 PM, wolfsrudel  wrote:

> yes, they are valid.
> au files are plain g711ulaw PCM.
>
> hth
>
> On 6/9/10, Angel Perez  wrote:
> >
> > Hi all:
> >
> >
> >
> > Anybody knows if .wav files recorded with CUE prompt manager (aka TUI)
> are
> > valid for bacd tcl scripts?
> >
> >
> >
> > BACD prompts are .au files but it think that .wav are also valid, anybody
> > can clarify this?
> >
> >
> >
> > Thanks
> >
> > _
> > Hotmail: Trusted email with Microsoft’s powerful SPAM protection.
> > https://signup.live.com/signup.aspx?id=60969
>
> --
> Sent from my mobile device
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Re: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence

[ShinGei Yong]

Hi Matthew,

I'm encounter a very similar problem as you, which the CUPC status not shown
anything. From the "Show server Health" from CUPC, the process halt at the
"Presence" status, symptom is it keep connecting & disconnecting.

I've configured the proxy under the proxy domain, and i can login to CUPC as
well.

How do you resolve the issue?

Thanks
Shingei

On Tue, Apr 27, 2010 at 3:29 PM, Angel Perez  wrote:

>  Hi I forgot to say that:
>
> 3: At CUC go to user>password>web password> uncheck "user have to change
> password next login" then at CUPC go to File>preference and add web username
> and pass at voice mail account
>
> --
> From: gorr...@hotmail.com
> To: ciscovoiceg...@gmail.com; ccie_voice@onlinestudylist.com
> Date: Tue, 27 Apr 2010 07:01:58 +
> Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence
>
>
> Hi:
>
> Assuming that you have taken all the neccesary steps to successfully
> integrate UCM and CUPS try the following:
>
> 1: Sometimes you have to restart the cups appliance if you have changed
> name to ip address
> 2: This is the normal situation, you would need to add the IPPM service
> to the hard phone associated to CUPC then add the users from the service
> menu at hard phone, then you would see other users and it presence.
>
> 3: At CUC CoS, check allow to use imap and allow to acces messages bodies,
> sometimes MWI notification on CUPC takes 30-40 sec
>
> hth
>
> --
> Date: Mon, 26 Apr 2010 12:30:57 -0500
> From: ciscovoiceg...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence
>
> All-
>
> I've been having a difficult time tonight trying to get CUPS Presence
> configured and working correctly.  I followed the Proctor Guide, referencing
> Vik's Vol 1 Walkthroughs and HouTong Luo's "Deploying Cisco Unified
> Presence" book.
>
> Here's the end result.  I can control my phones through the CUPC client.
> When calls ring inbound, I can divert them to voicemail via the client.  All
> the CUPS services are up and running (although, I did have to reboot the
> server to get AXL to remain up, much like my CUC and MWI email from late
> last week).
>
> Several things, I cannot get to work:
> 1.  Client status - My menu bars are mostly grayed out.  When I run the
> Presence Viewer on the CUPS, it cannot see my presence, even though I am
> logged in with CUPC.
>
> 2.  CUPC directory lookup - I cannot lookup and find BR1-Phone2 or
> HQ-Phone2 in the directory.  It is empty.  HuoTong mentioned that directory
> lookup was impossible without an AD integration.  Is this true?
>
> 3.  Voicemail notification - I went into CUC and enabled "Allow Users to
> Use Unified Client to Access Voice Mail" under CoS.  My phones can interact
> with MWI but not the CUPC client.
>
> When I run the troubleshooters, everything comes back green.  I have
> checked line associate, user profiles, SIP trunk, SIP trunk security
> profile, etc.
>
> Any ideas?
>
>
> --
>
> *Matthew Berry*
>
> *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written*
>
>
>
> *Vitals:*
>
> *GVoice: *+1.612.424.5044
>
> *Gmail*: ciscovoiceg...@gmail.com
>
> *Skype*: ciscovoiceguru
>
> *Twitter*: ciscovoiceguru
>
>
>
> *Cert Stats:*
>
> Cisco Cert Journey Began: Jan 1, 2009
>
> 1st Lab Attempt: Aug 16, 2010
>
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Re: [OSL | CCIE_Voice] Fwd: Failure of Adding subscriber-VM Server 2.0 - Closed

[ShinGei Yong]

Hi Otto,

I think i should apologize to the rest of fellow's and of course you too.
This is a very big mistake that i've made and let me unforgettable.
You're right, the CUCMBE is a single server solution which cannot add any
subscriber to the cluster, referring to CUCM SRND page 1017,

"Because Unified CMBE runs on a single hardware platform, it provides a
single instance of
Unified CM (a combined publisher and single subscriber instance, and a
secondary
subscriber instance is not configurable)."

Well, i think re-understand the SRND is what i need to do now before
proceeding with the new setup and installation.

Thanks for all the support.

Shingei

On Fri, Mar 5, 2010 at 8:13 PM, Otto Sanchez  wrote:

> Hey Shingei,
>
> That's why I asked those command outputs, I wanted to confirm that this was
> valid publisher node, because the message that you are getting relates to
> the sub not finding the publisher figure on the "pub" node, that should be
> the problem, you are basically installing two different products, so the sub
> installation will not recognize the cucmbe as a valid cucm "pub" node
>
> Remember that cucmbe is a single server solution and you cannot add any new
> subscriber to the cluster, it's like an standalone cluster by itself and
> hardened to work with the intended server infrastructure,
>
> HTH,
>
>
> On Thu, Mar 4, 2010 at 11:34 PM, ShinGei Yong wrote:
>
>> Hi Otto,
>>
>> I believe the installation process were followed carefully according to
>> the installation guide. The only difference is the the guide below show the
>> CUCM installation, and my current first node is CUCMBE(which mean CUCM and
>> CUC coexist). During the 2nd node(Sub) installation, i choose CUCM.
>>
>> Does it cause any issue?
>>
>> I'll capture below request file and send to you shortly.
>>
>> thanks
>> Shingei.
>>
>>
>> On Fri, Mar 5, 2010 at 10:32 AM, Otto Sanchez  wrote:
>>
>>> Hi,
>>>
>>> Did you make sure you carefully followed these guidelines?:
>>>
>>> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/7_0_1/install/cmins701.html#wp350770
>>>
>>> If yes, please provide install and upgrade log by the time the sub
>>> installation was performed and the output of the following commands:
>>>
>>> run sql select name,nodeid from ProcessNode
>>> show network cluster
>>> show network eth0
>>>
>>> Also, system and application log,
>>>
>>> thanks,
>>>
>>>
>>>
>>>
>>> On Thu, Mar 4, 2010 at 9:40 AM, ShinGei Yong wrote:
>>>
>>>> Opps, the message bounce back due to the file size restriction.
>>>>
>>>> Gang, you may request the capture log file from me if you willing to
>>>> help.
>>>>
>>>> Thanks.
>>>> Shingei.
>>>>
>>>> -- Forwarded message --
>>>> From: ShinGei Yong 
>>>> Date: Thu, Mar 4, 2010 at 10:02 PM
>>>> Subject: Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server
>>>> 2.0
>>>>  To: Duy Nguyen , ccie_voice@onlinestudylist.com,
>>>> bill.tal...@aos5.com, pav.c...@gmail.com
>>>> Cc: shingei.y...@gmail.com
>>>>
>>>>
>>>> Hi Bill,
>>>>
>>>> Regarding the VMWare setup, for VM Server 2.0, there are only two
>>>> network mode available which are to "host" only and "bridge". For the ease
>>>> of setup, i configured the network option for both nodes connected to host,
>>>> so i can monitor the setup easily, other words, my machine is the "gateway"
>>>> for both nodes. Both of them able to reach me through icmp ping and each
>>>> other vice versa, no any "firewall" in between. Also, there is only a 
>>>> single
>>>> 10.10.210.5 Pub server in VM network. Both nodes are assigned to the same
>>>> VMnet interface. You may refer to the attached pic named "CUCM-PUB-SUB &
>>>> RTMT Server connectivity"
>>>>
>>>> Hi Duy,
>>>>
>>>> IP was used throughout the entire setup, else the CUCM-PUB (10.10.210.5)
>>>> ip address will not appear on the validation error screen. Also, from the
>>>> jpg named CUCM-PUB-SUB, it shown that the IP address is assigned to PUB and
>>>> SUB.
>>>>
>>>> Hi Pavan,
>>>>
>>>> Please find attached log files. As the VM node not connect

[OSL | CCIE_Voice] Fwd: Failure of Adding subscriber-VM Server 2.0

[ShinGei Yong]

Opps, the message bounce back due to the file size restriction.

Gang, you may request the capture log file from me if you willing to help.

Thanks.
Shingei.

-- Forwarded message --
From: ShinGei Yong 
Date: Thu, Mar 4, 2010 at 10:02 PM
Subject: Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0
To: Duy Nguyen , ccie_voice@onlinestudylist.com,
bill.tal...@aos5.com, pav.c...@gmail.com
Cc: shingei.y...@gmail.com


Hi Bill,

Regarding the VMWare setup, for VM Server 2.0, there are only two network
mode available which are to "host" only and "bridge". For the ease of setup,
i configured the network option for both nodes connected to host, so i can
monitor the setup easily, other words, my machine is the "gateway" for both
nodes. Both of them able to reach me through icmp ping and each other vice
versa, no any "firewall" in between. Also, there is only a single
10.10.210.5 Pub server in VM network. Both nodes are assigned to the same
VMnet interface. You may refer to the attached pic named "CUCM-PUB-SUB &
RTMT Server connectivity"

Hi Duy,

IP was used throughout the entire setup, else the CUCM-PUB (10.10.210.5) ip
address will not appear on the validation error screen. Also, from the jpg
named CUCM-PUB-SUB, it shown that the IP address is assigned to PUB and SUB.


Hi Pavan,

Please find attached log files. As the VM node not connecting to a physical
switch, i'm not sure can sniff the VMnet adapter interface. The environment
doesn't contain a DNS server, i tried to reduce the complexity by not using
DNS in IP network.

Guys, i attached another 2 jpg which showing that i initiated the system
validation from CUCM-Sub console while RTMT is ON and monitoring the
process. Please take note on the timestamp on the WINXP machine, showing
20:55:46 which i started the validation. The 2nd jpg named log message 2
showing  that the validation has failed, and the RTMT system log showing
that the process ended at 20:55:50. During this period of time, the process
named

Mar 4 20:55:46, CUCM-PUB, Info,  sshd(pam_unix)[18244], : session opened for
user sftpuser by (uid=0), 18778
Mar 4 20:55:46, CUCM-PUB, Info,  sshd(pam_unix)[18244], : session closed for
user sftpuser, 18779...etc" and keep repeated

The last two attachment are the System Log and Security Log copy out from
RTMT. When refer to the security log at timing 20:55:46, the process show
"Mar 4 20:55:46, CUCM-PUB, Info,  sshd[18244], : Accepted password for
sftpuser from ::10.10.210.11 port 32829 ssh2, 13371
 Mar 4 20:55:46, CUCM-PUB, Info,  sshd[18246], : subsystem request for sftp,
13372"


All help are welcome, guys.

Many thanks
Shingei





On Thu, Mar 4, 2010 at 3:16 AM, Duy Nguyen  wrote:

> Make it simplistic and add by ip.
>
> Pavan  wrote:
>
> >Do you have a dns server that is reachable from sub?
> >Can you sniff the packets and post a capture?
> >
> >Sent from my phone
> >
> >On Mar 3, 2010, at 10:24 AM, ShinGei Yong 
> >wrote:
> >
> >> Hi Bill,
> >>
> >> Thanks for spotting out the difference between the naming of the
> >> nodes. Has been re-verify again the the node's naming and the
> >> confirmed the hostname is correct throughout the setup. The naming
> >> for Sub is under_scroll instead of hypen on the VM Console was just
> >> because its a naming of the VM console.
> >>
> >> Applying a upgrade patch on CUCM under VM platform don't think is
> >> possible. In fact, i didn't apply ANY patches to the CUCM-PUB, so am
> >> sure the both the CUCM-PUB and CUCM-SUB version are same.
> >>
> >> Anymore possible solution coming?
> >>
> >> Thanks
> >> Shingei
> >>
> >>
> >> On Tue, Mar 2, 2010 at 3:18 PM, Bill Talley 
> >> wrote:
> >> Are you sure hostnames are correct?   On the previous screenshot
> >> your CUCM-PUB servername has a hyphen between CUCM-PUB, however, you
> >> vmware console window for your sub has CUCM_SUB.  Is by chance your
> >> PUB hostname CUCM_PUB, with an underscore instead of a hyphen?
> >>
> >> Any chance you had applied a patch to the PUB previously and it's no
> >> longer version 7.0.1.11000-2?  I've never tried installing a sub
> >> with a different version so I'm not sure what the error would be
> >> that is generated in that scenario.
> >>
> >> On Mar 2, 2010, at 12:59 AM, ShinGei Yong wrote:
> >>
> >>> Hi Roger, Berry,
> >>>
> >>> Am already did that a few times, adding, removing and re-adding the
> >>> CUCM-Sub into the cluster inventory, but still got no chance to get
> >>> it work.
> 

Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0

[ShinGei Yong]

Hi Bill,

Thanks for spotting out the difference between the naming of the nodes. Has
been re-verify again the the node's naming and the confirmed the hostname is
correct throughout the setup. The naming for Sub is under_scroll instead of
hypen on the VM Console was just because its a naming of the VM console.

Applying a upgrade patch on CUCM under VM platform don't think is possible.
In fact, i didn't apply ANY patches to the CUCM-PUB, so am sure the both the
CUCM-PUB and CUCM-SUB version are same.

Anymore possible solution coming?

Thanks
Shingei


On Tue, Mar 2, 2010 at 3:18 PM, Bill Talley  wrote:

> Are you sure hostnames are correct?   On the previous screenshot your
> CUCM-PUB servername has a hyphen between CUCM-PUB, however, you vmware
> console window for your sub has CUCM_SUB.  Is by chance your PUB hostname
> CUCM_PUB, with an underscore instead of a hyphen?
>
> Any chance you had applied a patch to the PUB previously and it's no longer
> version 7.0.1.11000-2?  I've never tried installing a sub with a different
> version so I'm not sure what the error would be that is generated in that
> scenario.
>
> On Mar 2, 2010, at 12:59 AM, ShinGei Yong wrote:
>
> Hi Roger, Berry,
>
> Am already did that a few times, adding, removing and re-adding the
> CUCM-Sub into the cluster inventory, but still got no chance to get it work.
>
> I think i've out of my mind, can't think of any possibilities.
>
> Anyone else can help?
>
> Thanks
> Shingei
>
> On Mon, Mar 1, 2010 at 9:35 PM, Berry, Matthew J. <
> mjbe...@krollontrack.com> wrote:
>
>>  Good point, Roger. CUCM Pub needs something to validate against.
>>
>> - Sent from my Blackberry
>>
>> --
>>  *From*: ccie_voice-boun...@onlinestudylist.com <
>> ccie_voice-boun...@onlinestudylist.com>
>> *To*: ShinGei Yong ;
>> ccie_voice@onlinestudylist.com 
>> *Sent*: Mon Mar 01 04:27:15 2010
>> *Subject*: Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server
>> 2.0
>>
>> Hi Shingei,
>>
>> This maybe a long shot, but anyway, have you added the sub as a UCM server
>> on the pub before you tried to add it as a 2nd node?
>>
>>
>> *Roger Källberg*
>> Unified Communication Consultant
>> Cygate AB
>>
>>
>>
>> *From:* ShinGei Yong [mailto:shingei.y...@gmail.com]
>> *Sent:* den 1 mars 2010 03:29
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0
>>
>>
>> Hi All,
>>
>> Im currently having the issue of adding the 2nd node to the UCM cluster.
>> The error message as per attached jpg "Configuration Validation with
>> CUCM-PUB Failed".
>>
>> I've did some research based on the error message, either from CISCO or
>> some older post. What i've tried out was:
>>
>> 1. Adding a NTP server to CUCM-PUB. CUCM-PUB was able to get the time
>> source from my WINXP modified NTP server, and can be viewed from CUCM CLI.
>> 2. Changed of Security Password. Based on CISCO explanation, it could be
>> due to the security password mismatch between CUCM-PUB and SUB. I've reset
>> the security password and reboot the CUCM-PUB, but still got no luck. I did
>> tried to input an "incorrect" Security Password while adding the 2nd node,
>> it did correctly prompt that the password was error. So confirmed that the
>> security password entered was correct.
>> 3. Some post mentioned that the MTU for the CUCM, will have some issue if
>> leave it to default, but i'm not too sure what does it mean? Should i change
>> it to MTU size to 1492?
>>
>> I'm currently using VMWare Server 2.0.
>>
>> regards,
>> Shingei
>>
>>
>>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
> 
> Bill Talley
> IP Telephony Consultant
>
> Alexander Open Systems, Inc
> 12851 Foster, Suite 200
> Overland Park, KS 66213
> (913) 307-2330, Scheduling Group
> (913) 307-2380, Fax
> bill.tal...@aos5.com
> http://www.aos5.com
>
>
>
>
> CONFIDENTIALITY NOTICE: This electronic mail transmission (including any 
> accompanying attachments) is intended solely for its authorized recipient(s), 
> and may contain confidential and/or legally privileged information. If you 
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>

Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0

[ShinGei Yong]

Hi Roger, Berry,

Am already did that a few times, adding, removing and re-adding the CUCM-Sub
into the cluster inventory, but still got no chance to get it work.

I think i've out of my mind, can't think of any possibilities.

Anyone else can help?

Thanks
Shingei

On Mon, Mar 1, 2010 at 9:35 PM, Berry, Matthew J.
wrote:

>  Good point, Roger. CUCM Pub needs something to validate against.
>
> - Sent from my Blackberry
>
> --
>  *From*: ccie_voice-boun...@onlinestudylist.com <
> ccie_voice-boun...@onlinestudylist.com>
> *To*: ShinGei Yong ;
> ccie_voice@onlinestudylist.com 
> *Sent*: Mon Mar 01 04:27:15 2010
> *Subject*: Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server
> 2.0
>
>  Hi Shingei,
>
> This maybe a long shot, but anyway, have you added the sub as a UCM server
> on the pub before you tried to add it as a 2nd node?
>
>
>
> *Roger Källberg*
> Unified Communication Consultant
> Cygate AB
>
>
>
> *From:* ShinGei Yong [mailto:shingei.y...@gmail.com]
> *Sent:* den 1 mars 2010 03:29
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0
>
>
>
> Hi All,
>
> Im currently having the issue of adding the 2nd node to the UCM cluster.
> The error message as per attached jpg "Configuration Validation with
> CUCM-PUB Failed".
>
> I've did some research based on the error message, either from CISCO or
> some older post. What i've tried out was:
>
> 1. Adding a NTP server to CUCM-PUB. CUCM-PUB was able to get the time
> source from my WINXP modified NTP server, and can be viewed from CUCM CLI.
> 2. Changed of Security Password. Based on CISCO explanation, it could be
> due to the security password mismatch between CUCM-PUB and SUB. I've reset
> the security password and reboot the CUCM-PUB, but still got no luck. I did
> tried to input an "incorrect" Security Password while adding the 2nd node,
> it did correctly prompt that the password was error. So confirmed that the
> security password entered was correct.
> 3. Some post mentioned that the MTU for the CUCM, will have some issue if
> leave it to default, but i'm not too sure what does it mean? Should i change
> it to MTU size to 1492?
>
> I'm currently using VMWare Server 2.0.
>
> regards,
> Shingei
>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] CCIE VOICE LAB 3 new locations unveils..

[ShinGei Yong]

Hi all,

Just to share this info to all voice lab candidates...Another 3 new
locations available:

Bangalore, India
Hong Kong PRC, and
Beijing PRC

It sound great for those who resident in Asia, maybe.

Official Link:
http://www.cisco.com/web/learning/le3/ccie/voice/lab_exam.html
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Fwd: LAB 1A CME phone not registering

[ShinGei Yong]

forwarded to all.

-- Forwarded message --
From: ShinGei Yong 
Date: Mon, Jan 11, 2010 at 11:01 PM
Subject: Re: [OSL | CCIE_Voice] LAB 1A CME phone not registering
To: "Berry, Matthew J." 


Hi Randall,

Can you do a "debug tftp events", to check what are the events generated
during registration.

Also, are you registering SCCP or SIP phone? ur config only shown SCCP
defined.

How come you defined your VVLAN interface as your router-id? My
understanding is that, should defined the loopback interface instead. please
correct me if wrong.

regards,

On Mon, Jan 11, 2010 at 8:30 PM, Berry, Matthew J.  wrote:

>  Randall,
>
> I don't see a "ntp server x.x.x.x" command in your configuration.
>
> CUCME needs this information before allowing phones to register.
>
> Try entering that command followed by a "telephony-service" "create
> cnf-files"
>
> Restart your phone and see if it works.
>
> Matthew
>
>  Digital Footprint:
> Skype: ciscovoiceguru
>  --
> *From:* ccie_voice-boun...@onlinestudylist.com [
> ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm [
> randall.cr...@harmonicinc.com]
> *Sent:* Sunday, January 10, 2010 9:59 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] LAB 1A CME phone not registering
>
>   HI,
>
> I am having a hard time getting the 1 phone on CME to register.
>
>
>
> I am attaching my config
>
> **
>
> *Thanks*
>
> *Randall*
>
> **
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com