from:"datucha123 datucha123"

Are you using VMWare for you Unity Server?

If so then that behavour is observed when the Unity Connection is installed
in virtual environment.

I am also running to such issue some times, but the reboot of Unity server
helps me.






On Sun, Feb 19, 2012 at 4:16 AM, chase mergenthal cm3_...@hotmail.comwrote:

  Something weird, I've run into; After doing the integration between CUCM
 and CUC, I can leave messages from users that have accounts on Unity, but
 not from the PSTN or other phones...

 Logs below:

 02/18/2012 23:17:33.300
 |7915,PhoneSystem-PG-001,FBBA1FCBF2B8440982D981A38725901D,Arbiter,-1,Incoming
 Call [callerID='5002' callerName='Ron Paul' calledID='5600'
 redirectingID='' lastRedirectingID='' origin=16=Unknown xferType=16=Unknown
 reason=1=Direct lastReason=1024=Unknown] port=PhoneSystem-PG-001
 portsInUse=1 ansPortsFree=1 callGuid=FBBA1FCBF2B8440982D981A38725901D|
 02/18/2012 23:22:40.717
 |7916,PhoneSystem-PG-001,DA7997B88186462C8AD2A77DC8265A4E,Arbiter,-1,Incoming
 Call [callerID='911' callerName='Emergency Services' calledID='5002'
 redirectingID='5002' lastRedirectingID='5002' origin=16=Unknown
 xferType=16=Unknown reason=4=FwdNoAnswer lastReason=4=FwdNoAnswer]
 port=PhoneSystem-PG-001 portsInUse=1 ansPortsFree=1
 callGuid=DA7997B88186462C8AD2A77DC8265A4E|
 02/18/2012 23:22:43.090
 |7916,PhoneSystem-PG-001,DA7997B88186462C8AD2A77DC8265A4E,MiuGeneral,25,GetLicenseStatusLimit
 tag='LicRealspeakSessionsMax'. |
 02/18/2012 23:22:53.858
 |7916,PhoneSystem-PG-001,DA7997B88186462C8AD2A77DC8265A4E,MiuGeneral,25,GetLicenseStatusLimit
 tag='LicMaxMsgRecLenIsLicensed'. |
 02/18/2012 23:24:47.859
 |7915,PhoneSystem-PG-001,20C4C8D64C5D4E3892A77F79D6B5846D,Arbiter,-1,Incoming
 Call [callerID='3002' callerName='John Adams' calledID='5002'
 redirectingID='5002' lastRedirectingID='5002' origin=16=Unknown
 xferType=16=Unknown reason=4=FwdNoAnswer lastReason=4=FwdNoAnswer]
 port=PhoneSystem-PG-001 portsInUse=1 ansPortsFree=1
 callGuid=20C4C8D64C5D4E3892A77F79D6B5846D|
 02/18/2012 23:24:48.594
 |7915,PhoneSystem-PG-001,20C4C8D64C5D4E3892A77F79D6B5846D,MiuGeneral,25,GetLicenseStatusLimit
 tag='LicRealspeakSessionsMax'. |
 02/18/2012 23:24:58.690
 |7915,PhoneSystem-PG-001,20C4C8D64C5D4E3892A77F79D6B5846D,MiuGeneral,25,GetLicenseStatusLimit
 tag='LicMaxMsgRecLenIsLicensed'. |
 02/18/2012 23:52:16.695
 |7916,PhoneSystem-PG-001,517F2E6866604E18AF7384BEC27BAC8F,Arbiter,-1,Incoming
 Call [callerID='911' callerName='Emergency Services' calledID='5002'
 redirectingID='5002' lastRedirectingID='5002' origin=16=Unknown
 xferType=16=Unknown reason=4=FwdNoAnswer lastReason=4=FwdNoAnswer]
 port=PhoneSystem-PG-001 portsInUse=1 ansPortsFree=1
 callGuid=517F2E6866604E18AF7384BEC27BAC8F|
 02/18/2012 23:52:18.722
 |7916,PhoneSystem-PG-001,517F2E6866604E18AF7384BEC27BAC8F,MiuGeneral,25,GetLicenseStatusLimit
 tag='LicRealspeakSessionsMax'. |
 02/18/2012 23:52:29.195
 |7916,PhoneSystem-PG-001,517F2E6866604E18AF7384BEC27BAC8F,MiuGeneral,25,GetLicenseStatusLimit
 tag='LicMaxMsgRecLenIsLicensed'. |

 -Chase


 --
 If winners never quit and quitters never win, then who coined the phrase,
 Quit while you’re still ahead.?


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Re: [OSL | CCIE_Voice] Embedded BACD Prompts and DropThrough

1.   Are prompts also embedded in the IOS? Or do they need to be copied
in the router’s flash?

No, the Prompts are not embedded in the IOS, you need to manually add them
into Flash.


2.   Does drop through mode work with embedded BACD?
Yes, embedded BACD works for Drop Through Mode very well.

You can find the configuration examples here:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1002001




On Sun, Feb 19, 2012 at 7:59 AM, AJ BG ciscoie2...@gmail.com wrote:

 Two questions about embedded BACD.

 1.   Are prompts also embedded in the IOS? Or do they need to be
 copied in the router’s flash?

 2.   Does drop through mode work with embedded BACD?
  Does anyone have a working copy of embedded BACD configuration?
 Thanks
 AJ

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Re: [OSL | CCIE_Voice] SRST MoH issue

I have tested, and Multicast MoH is supported in SRST Mode for SCCP IP
Phones, I can hear the Music.

On Tue, Dec 13, 2011 at 2:54 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 I have not fixed it, I have still the ToH for SRST SCCP IP Phones.
 It is not possible to get the MoH for SCCP SRST IP Phones

   On Tue, Dec 13, 2011 at 2:44 PM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:

  Good …  but how you fixed.? Is it matching below statement.

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
 *Sent:* Tuesday, December 13, 2011 11:58 AM
 *To:* Mohammed Al Baqari
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] SRST MoH issue

 ** **

 Thank you very much for your answer. 

  

 I have solved it already :)

 On Tue, Dec 13, 2011 at 1:42 AM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:

 Hi,

  

 I am not sure whether you have fixed this or not. But in between in case
 you are using fixed source feed and MMoH SRST, then the following
 restriction is valid.

  

 MOH is supplied only to PSTN and VoIP G.711 calls. Local IP phone callers
 hear a repeating tone on hold for reassurance that they are still connected.
 

  

 Regards,

 Mohammed Al Baqari

  

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Friday, December 02, 2011 10:47 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] SRST MoH issue

  

 Hi Guys,

 In SRST mode Branch 1 , I cannot hear the MOH from Flash ,but I hear beep
 sound. There is no MOH playing for internal calls from BR1 phn 1 to BR1 Phn
 2.

 But when I put the PSTN caller on hold the PSTN phn can hear the MOH. I
 tried the debug ephone MOHit says:

  

 *No MOH entry for DN 1*

  

 *Dec  2 19:27:37.678: ifs_read flash:MobileConnectOn.ulaw.wav end of file
 at 15078 read 6832 = 21910
 *Dec  2 19:27:37.682: moh tail fill from 46 at 0x66BBACF4 length 1168
 *Dec  2 19:27:38.254: ephone_hold_resume ignored for s2s set on dn=1
 chan=1 hold=1 callID=1547
 **Dec  2 19:27:39.922: No MOH entry for DN 1**
 **Dec  2 19:27:39.926: ephone_hold_resume ignored for s2s set on dn=1
 chan=1 hold=0 callID=1547
 *Dec  2 19:27:40.678: ifs_read flash:MobileConnectOn.ulaw.wav end of file
 at 17214 read 4696 = 21910
 *Dec  2 19:27:40.678: moh tail fill from 46 at 0x66BB855C length 3304
 *Dec  2 19:27:40.882: MoH route If Vlan11 ETHERNET 177.2.11.1 via ARP
 *Dec  2 19:27:40.882: MoH route If Loopback0 46 177.1.254.2 via
 177.1.254.2

  

  

 ** **



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Re: [OSL | CCIE_Voice] MVA - Phone hangs up when trying to make a call.

That is because the MVA number does not match 5999 configured in CUCM.
please make sure that 5999 is also configured as MVA access number in CUCM,
and no Called Party transformation takes place from Incoming POTS dial-peer
up to CUCM.

On Sun, Feb 19, 2012 at 11:17 PM, Emanuel Damasceno
aedamasc...@gmail.comwrote:

 Hey guys,

 I've configured everything (I suppose...) RDP, RD, MVA on SP... Everything
 seems to be in order. It welcomes me, asks to dial a password, yadda yadda,
 when I ask to make a call by pressing 1, I dial the numbers and when I
 press # the connection drops.

 Here is my config:
 HQ-RTR
 application
  service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
 !
 dial-peer voice 5999 pots
  service mva
  incoming called-number 5999
 !
 dial-peer voice 5000 voip
  destination-pattern 5...
  voice-class codec 1
  voice-class h323 1
  session target ipv4:10.10.210.11
  no vad
 dial-peer voice 5001 voip
  preference 1
  destination-pattern 5...
  voice-class codec 1
  voice-class h323 1
  session target ipv4:10.10.210.10
  no vad

 The remote Profile and Remote Destination Profile are correct, otherwise
 it wouldn't ask me to put the PIN straight up. It would ask me first for my
 number then PIN.
 Any ideas?

 *Emanuel Damasceno*
 CCNP Voice




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Re: [OSL | CCIE_Voice] RSVP over MLPPP

2012-02-18 Thread datucha123 datucha123

RSVP may not work over True MLPP (I do not know exactly).

But for MLP Over Frame Relay it will work.

On Sat, Feb 18, 2012 at 6:58 PM, Radhesh Naik radheshn...@gmail.com wrote:

  Hi,

 ** **

 Came across this statement under SRND.

 ** **

 “RSVP is currently not available on Bundle Interfaces, including MLPPP,…”*
 ***

 ** **

 Does this mean we can’t configure RSVP on an interface that is configured
 with MLP ? 

 ** **

 Because auto qos fr-atm will configure the interface as  MLP LFI.

 ** **

 I am sure I am missing something here, enlighten please.

 ** **

 Regards,

 ** **

 Radhesh

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Re: [OSL | CCIE_Voice] New Labs Lab 4 7.4

2012-02-17 Thread datucha123 datucha123

No, as I remember, Paging is not supported on SIP Phones, at least in
version 7.0

On Fri, Feb 17, 2012 at 6:33 AM, chase mergenthal cm3_...@hotmail.comwrote:

  Is paging supported on SIP phones?

 Got it to work on my SCCP phones just fine...

 -Chase


 --
 If winners never quit and quitters never win, then who coined the phrase,
 Quit while you’re still ahead.?


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Re: [OSL | CCIE_Voice] built-in bridge support for iLBC g729

Interesting question.

I think that Phones Built in Bridge does support G729 and other codecs,
which are supported by IP Phone.

On Wed, Feb 15, 2012 at 10:30 AM, Juan Lopez lopez.hernandez.j...@gmail.com
 wrote:

 can anyone confirm the phone built-in bridge does support g729? I'm
 running it in my lab, no HW CFB are configured and I put all SW CFB in
 their own MRG, to be sure they are not used:

 admin:show perf query class Cisco SW Conference Bridge Device
 ==query class :
  - Perf class (Cisco SW Conference Bridge Device) has instances and values:
 CFB_2   - AllocatedResourceCannotOpenPort = 0
 CFB_2   - OutOfResources = 0
 CFB_2   - ResourceActive = 0
 CFB_2   - ResourceAvailable  = 48
 CFB_2   - ResourceTotal  = 48
 CFB_2   - SWConferenceActive = 0
 CFB_2   - SWConferenceCompleted  = 0
 CFB_3   - AllocatedResourceCannotOpenPort = 0
 CFB_3   - OutOfResources = 0
 CFB_3   - ResourceActive = 0
 CFB_3   - ResourceAvailable  = 48
 CFB_3   - ResourceTotal  = 48
 CFB_3   - SWConferenceActive = 0
 CFB_3   - SWConferenceCompleted  = 0

 admin:show perf query class Cisco HW Conference Bridge Device
 ==query class :
  - Perf class (Cisco HW Conference Bridge Device) has instances and values:
 no values are returned

 Nevertheless, the barge function is working: barger and bargee use g722,
 the 3rd party uses g729. same goes is 3rd party would use iLBC.

 thanks!

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Re: [OSL | CCIE_Voice] Multicast MOH

George,

 Flash MMoH cannot be traversed through L3.

So that the MMoH is not supported for remote IP Phones on CUCME.

CUCME Admin guide says that. Unfortunately I do not remember the exact
page, but its there. Also I have tested that and it is true, the MMoH From
flash does not go to remote IP Phones (for CUCME or SRST even) which are on
subnet separated by L3 from the CUCME Routers interface.






On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze gogli...@gmail.comwrote:

 Hi Boris,

 You are wrong, the TTL on multicast music on hold is NOT 1...
 and depending on what you need, you can have pim enabled and deliver the
 mmoh from flash to other remote networks, and by remote I mean across any
 L3 device.

 As well, you do not have a way to specify the TTL on the mmoh from router
 flash like in CUCM where you can specify the TTL you need.
 So it will be 255 I believe. I would have to check on that.


 I hope this answers the original question as well, if you need mmoh from
 router flash to traverse any other layer3 devices, you need to enable PIM.
 otherwise you don't.

 Cheers,


 On Wed, Feb 15, 2012 at 12:29 AM, Boris boris.k...@gmail.com wrote:

  Afaik you never need multicast routing configured for the Moh from
 flash because this traffic has TTL=1, so it wont traverse Layer 3 boundary.

 This is why you need the route attribute in the moh statement.

 Sent from my mobile device, sorry for typos.
 ---
 Regards
 Boris

 On 15/02/2012, at 8:28, Emanuel Damasceno aedamasc...@gmail.com wrote:

  Hello Experts,

 I am watching IPExpert's video on demand on Multicast MOH.

 Did I understand this right?

 multicast moh 239.1.1.1 port 16384  if I have it like this, I have to
 put ip pim dense-mode on my interfaces

 multicast moh 239.1.1.1 port 16384 route 10.10.201.1 10.10.110.2  if I
 have it like this, I DON'T have to put ip pim dense-mode on my interfaces

 Could somebody please confirm?
 Thanks*
 Emanuel Damasceno*
 CCNP Voice



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 www.PlatinumPlacement.com http://www.platinumplacement.com/



 ___
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Re: [OSL | CCIE_Voice] Multicast MOH

You will not be able to hear the Music itslef.

In my testing the Multicast was also traversed to my Remote Branch Router -
I was able to see the Multicast stream and endpoints joined  there with
show ip mroute and show ip igmp groups, but the actual stream of Music
was not heard, based on CUCME Admin guide.

On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze gogli...@gmail.com wrote:

 and voila, here's the proof:
 sorry but had to remove IP addresses for security reasons.

 I'm using pim sparse-dense mode, with auto RP.

 mmohrtr#show ip pim neighbor
 PIM Neighbor Table
 Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
   P - Proxy Capable, S - State Refresh Capable, G - GenID Capable
 Neighbor  InterfaceUptime/ExpiresVer   DR
 Address
 Prio/Mode
 x.x.x.x  GigabitEthernet0/1   00:23:38/00:01:24 v21 / G
 *this address is the FW! *

 mmohrtr#show ip mroute
 IP Multicast Routing Table
 Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
 Connected,
L - Local, P - Pruned, R - RP-bit set, F - Register flag,
T - SPT-bit set, J - Join SPT, M - MSDP created entry, E - Extranet,
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), 00:06:33/00:02:56, RP x.x.x.x, flags: SP
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list: Null

 (*, 224.0.1.39), 00:30:17/stopped, RP 0.0.0.0, flags: DCL
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
 Loopback1, Forward/Sparse-Dense, 00:30:17/stopped

 (x.x.x.x, 224.0.1.39), 00:30:16/00:02:42, flags: LT
   Incoming interface: Loopback1, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped

 (*, 224.0.1.40), 00:30:16/stopped, RP 0.0.0.0, flags: DCL
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
 Loopback1, Forward/Sparse-Dense, 00:30:16/stopped

 (x.x.x.x, 224.0.1.40), 00:30:16/00:02:24, flags: LT
   Incoming interface: Loopback1, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped


 And I created a SPAN port, on the switch, and spanned all traffic from
 mmohrtr to the Firewall!!!
 The part of screenshot attached... which says TTL 255 as I said!

 Regards,


 On Wed, Feb 15, 2012 at 11:54 AM, George Goglidze gogli...@gmail.comwrote:

 Datucha,

 I have it configured right now in the lab... and it works just fine.

 VGW - FW - L3 Switch - Phones.

 Works just fine...

 I will do some wireshark traces and show the packets.


 On Wed, Feb 15, 2012 at 10:53 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 George,

  Flash MMoH cannot be traversed through L3.

 So that the MMoH is not supported for remote IP Phones on CUCME.

 CUCME Admin guide says that. Unfortunately I do not remember the exact
 page, but its there. Also I have tested that and it is true, the MMoH From
 flash does not go to remote IP Phones (for CUCME or SRST even) which are on
 subnet separated by L3 from the CUCME Routers interface.






 On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze gogli...@gmail.comwrote:

 Hi Boris,

 You are wrong, the TTL on multicast music on hold is NOT 1...
 and depending on what you need, you can have pim enabled and deliver
 the mmoh from flash to other remote networks, and by remote I mean across
 any L3 device.

 As well, you do not have a way to specify the TTL on the mmoh from
 router flash like in CUCM where you can specify the TTL you need.
 So it will be 255 I believe. I would have to check on that.


 I hope this answers the original question as well, if you need mmoh
 from router flash to traverse any other layer3 devices, you need to enable
 PIM. otherwise you don't.

 Cheers,


 On Wed, Feb 15, 2012 at 12:29 AM, Boris boris.k...@gmail.com wrote:

  Afaik you never need multicast routing configured for the Moh from
 flash because this traffic has TTL=1, so it wont traverse Layer 3 
 boundary.

 This is why you need the route attribute in the moh statement.

 Sent from my mobile device, sorry for typos.
 ---
 Regards
 Boris

 On 15/02/2012, at 8:28, Emanuel Damasceno aedamasc...@gmail.com
 wrote:

  Hello Experts,

 I am watching IPExpert's video on demand on Multicast MOH.

 Did I understand this right?

 multicast moh 239.1.1.1 port 16384  if I have it like this, I have
 to put ip pim dense-mode on my interfaces

 multicast moh 239.1.1.1

Re: [OSL | CCIE_Voice] Multicast MOH

I cannot find it any more :(((

But I remember, that I have read about that somewhere.

Ok i will make the test again, (10 minutes) and will post the results.



On Wed, Feb 15, 2012 at 4:32 PM, George Goglidze gogli...@gmail.com wrote:

 Datucha,

 I'm hearing the music perfectly :-)

 If you can't make it work, that's something wrong with your configuration,
 but don't say it won't work! :-)))

 And please tell me where in CUCME admin guide it says such thing I'll
 eat my hat if you show me the section!

 Cheers,

 On Wed, Feb 15, 2012 at 1:26 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 You will not be able to hear the Music itslef.

 In my testing the Multicast was also traversed to my Remote Branch Router
 - I was able to see the Multicast stream and endpoints joined  there with
 show ip mroute and show ip igmp groups, but the actual stream of Music
 was not heard, based on CUCME Admin guide.

  On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze gogli...@gmail.comwrote:

 and voila, here's the proof:
 sorry but had to remove IP addresses for security reasons.

 I'm using pim sparse-dense mode, with auto RP.

 mmohrtr#show ip pim neighbor
 PIM Neighbor Table
 Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
   P - Proxy Capable, S - State Refresh Capable, G - GenID Capable
 Neighbor  InterfaceUptime/ExpiresVer   DR
 Address
 Prio/Mode
 x.x.x.x  GigabitEthernet0/1   00:23:38/00:01:24 v21 / G
 *this address is the FW! *

 mmohrtr#show ip mroute
 IP Multicast Routing Table
 Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
 Connected,
L - Local, P - Pruned, R - RP-bit set, F - Register flag,
T - SPT-bit set, J - Join SPT, M - MSDP created entry, E -
 Extranet,
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), 00:06:33/00:02:56, RP x.x.x.x, flags: SP
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list: Null

 (*, 224.0.1.39), 00:30:17/stopped, RP 0.0.0.0, flags: DCL
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
 Loopback1, Forward/Sparse-Dense, 00:30:17/stopped

 (x.x.x.x, 224.0.1.39), 00:30:16/00:02:42, flags: LT
   Incoming interface: Loopback1, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped

 (*, 224.0.1.40), 00:30:16/stopped, RP 0.0.0.0, flags: DCL
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
 Loopback1, Forward/Sparse-Dense, 00:30:16/stopped

 (x.x.x.x, 224.0.1.40), 00:30:16/00:02:24, flags: LT
   Incoming interface: Loopback1, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped


 And I created a SPAN port, on the switch, and spanned all traffic from
 mmohrtr to the Firewall!!!
 The part of screenshot attached... which says TTL 255 as I said!

 Regards,


 On Wed, Feb 15, 2012 at 11:54 AM, George Goglidze gogli...@gmail.comwrote:

 Datucha,

 I have it configured right now in the lab... and it works just fine.

 VGW - FW - L3 Switch - Phones.

 Works just fine...

 I will do some wireshark traces and show the packets.


 On Wed, Feb 15, 2012 at 10:53 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 George,

  Flash MMoH cannot be traversed through L3.

 So that the MMoH is not supported for remote IP Phones on CUCME.

 CUCME Admin guide says that. Unfortunately I do not remember the exact
 page, but its there. Also I have tested that and it is true, the MMoH From
 flash does not go to remote IP Phones (for CUCME or SRST even) which are 
 on
 subnet separated by L3 from the CUCME Routers interface.






 On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze 
 gogli...@gmail.comwrote:

 Hi Boris,

 You are wrong, the TTL on multicast music on hold is NOT 1...
 and depending on what you need, you can have pim enabled and deliver
 the mmoh from flash to other remote networks, and by remote I mean across
 any L3 device.

 As well, you do not have a way to specify the TTL on the mmoh from
 router flash like in CUCM where you can specify the TTL you need.
 So it will be 255 I believe. I would have to check on that.


 I hope this answers the original question as well, if you need mmoh
 from router flash to traverse any other layer3 devices, you need to 
 enable
 PIM. otherwise you don't.

 Cheers,


 On Wed, Feb 15, 2012 at 12:29 AM

Re: [OSL | CCIE_Voice] Multicast MOH

I have tested that, and the MMoH is working.

Sorry for incorrect information.

In previous test I had the MMoH binded to incorrect interface.

Now it is working

I am burning with shame, sorry

On Wed, Feb 15, 2012 at 4:32 PM, George Goglidze gogli...@gmail.com wrote:

 Datucha,

 I'm hearing the music perfectly :-)

 If you can't make it work, that's something wrong with your configuration,
 but don't say it won't work! :-)))

 And please tell me where in CUCME admin guide it says such thing I'll
 eat my hat if you show me the section!

 Cheers,

 On Wed, Feb 15, 2012 at 1:26 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 You will not be able to hear the Music itslef.

 In my testing the Multicast was also traversed to my Remote Branch Router
 - I was able to see the Multicast stream and endpoints joined  there with
 show ip mroute and show ip igmp groups, but the actual stream of Music
 was not heard, based on CUCME Admin guide.

  On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze gogli...@gmail.comwrote:

 and voila, here's the proof:
 sorry but had to remove IP addresses for security reasons.

 I'm using pim sparse-dense mode, with auto RP.

 mmohrtr#show ip pim neighbor
 PIM Neighbor Table
 Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
   P - Proxy Capable, S - State Refresh Capable, G - GenID Capable
 Neighbor  InterfaceUptime/ExpiresVer   DR
 Address
 Prio/Mode
 x.x.x.x  GigabitEthernet0/1   00:23:38/00:01:24 v21 / G
 *this address is the FW! *

 mmohrtr#show ip mroute
 IP Multicast Routing Table
 Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
 Connected,
L - Local, P - Pruned, R - RP-bit set, F - Register flag,
T - SPT-bit set, J - Join SPT, M - MSDP created entry, E -
 Extranet,
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), 00:06:33/00:02:56, RP x.x.x.x, flags: SP
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list: Null

 (*, 224.0.1.39), 00:30:17/stopped, RP 0.0.0.0, flags: DCL
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
 Loopback1, Forward/Sparse-Dense, 00:30:17/stopped

 (x.x.x.x, 224.0.1.39), 00:30:16/00:02:42, flags: LT
   Incoming interface: Loopback1, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped

 (*, 224.0.1.40), 00:30:16/stopped, RP 0.0.0.0, flags: DCL
   Incoming interface: Null, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
 Loopback1, Forward/Sparse-Dense, 00:30:16/stopped

 (x.x.x.x, 224.0.1.40), 00:30:16/00:02:24, flags: LT
   Incoming interface: Loopback1, RPF nbr 0.0.0.0
   Outgoing interface list:
 GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped


 And I created a SPAN port, on the switch, and spanned all traffic from
 mmohrtr to the Firewall!!!
 The part of screenshot attached... which says TTL 255 as I said!

 Regards,


 On Wed, Feb 15, 2012 at 11:54 AM, George Goglidze gogli...@gmail.comwrote:

 Datucha,

 I have it configured right now in the lab... and it works just fine.

 VGW - FW - L3 Switch - Phones.

 Works just fine...

 I will do some wireshark traces and show the packets.


 On Wed, Feb 15, 2012 at 10:53 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 George,

  Flash MMoH cannot be traversed through L3.

 So that the MMoH is not supported for remote IP Phones on CUCME.

 CUCME Admin guide says that. Unfortunately I do not remember the exact
 page, but its there. Also I have tested that and it is true, the MMoH From
 flash does not go to remote IP Phones (for CUCME or SRST even) which are 
 on
 subnet separated by L3 from the CUCME Routers interface.






 On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze 
 gogli...@gmail.comwrote:

 Hi Boris,

 You are wrong, the TTL on multicast music on hold is NOT 1...
 and depending on what you need, you can have pim enabled and deliver
 the mmoh from flash to other remote networks, and by remote I mean across
 any L3 device.

 As well, you do not have a way to specify the TTL on the mmoh from
 router flash like in CUCM where you can specify the TTL you need.
 So it will be 255 I believe. I would have to check on that.


 I hope this answers the original question as well, if you need mmoh
 from router flash to traverse any other layer3 devices, you need to 
 enable
 PIM. otherwise you don't.

 Cheers,


 On Wed, Feb

Re: [OSL | CCIE_Voice] Query wrt to Cbarge on BR1 router without using the CME for SRST

You can also configure the Privacy Settings globally, at Telephony-service
configuration. with no privacy command, so that you will not need to
disable it per ephone.



On Wed, Feb 15, 2012 at 11:10 PM, Vik Malhi vma...@ipexpert.com wrote:

 I don't see the restriction in OWLE lab #2.

 The HA question 7.1 states ·   You should ensure that all learned
 ephones and ephone-dn’s appear in the running config to achieve this task.

 This indicates srst mode auto-provision all which is required for cBarge
 preservation since privacy needs to be disabled at the ephone level.

 Let me know the details of the lab and question number and I'll try and
 clear it up- but nonetheless- you are correct in what you have stated- you
 cannot preserve cBarge with call-manager-fallback

 Thanks


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

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




   On Feb 15, 2012, at 4:12 PM, Rajasekar Shanmugam wrote:

   Experts -

 I`m practicing the scenario 2 from the IP Expert OWLE series  the
 question asks us ,  not to use the CME based SRST for BR1. So we are forced
 to use the call-manager-fallback. There is a requirement later in the lab
 ,asking for Cbarge functionality on SRST. Wondering , if we have an option
 to register the hardware media resources (CFB) with the
 call-manager-fallback to get this working ? The solution guide suggests to
 configure the telephony service in order to do so. Confused here won`t
 that break the original requirement  in the HA section , that asked us not
 to use the CME SRST ? Please advise.

 --
 Raj
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Re: [OSL | CCIE_Voice] enabling Chat option in Cisco Unified personal communicator

Well, I think you have to add contacts to you CUPC client.

But here are some restrictions:

1) When you add Contacts from CUPC client directly, those are just
contacts, and they do not support Presence.
2) To support Presence (Chat also will be supported) you need to add
contacts from End User Web page of Presence Server  -
http://Presence/ccmuser http://%3cpresence%3e/ccmuser   and sign in
with End User username and password, then fill out the Contacts there.
3) The mentioned method (2), will support Presence for other Users who are
also having CUPC. So to configure the Presence for users endpoints (like IP
Phone DN's) you have to associate those DNs with corresponding End Users as
well, along with section 2, for watcher.
4) Integrate CUPS with AD, and let the users to pull out the Contacts from
AD using CUPC. In this case, CUPC will support Presence (Chat) with added
contacts.

Also do not forget to assign the License Capabilites to all End Users, that
must be part of Presence and CUPC, so that they will be visible by Presence
Server. And Chat option will be available for you added users in CUPC.

If that is not the right answer you were expecting, then please right in
more details what are you trying to achieve.

On Thu, Feb 16, 2012 at 12:38 AM, darshan ccievoice0...@hotmail.com wrote:

  Dear Experts;

 ** **

 I want to enable CHAT option in Cisco Unified Personal communicator for
 Presence question.

 ** **

 Appreciate your help for this.

 ** **

 Regards

 darsh

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Re: [OSL | CCIE_Voice] New 5 Labs: Lab 2 UXXC

Zero and One are already pre-recorded (by default) in UCCX, so you can use
them.

On Sun, Feb 12, 2012 at 12:09 AM, Randall Crumm rrcr...@yahoo.com wrote:

  HI,
 I know the steps I need to do but I am unsure of the prompts I need to
 record.

 Please let know what I need to record.

 I believe I need to record:

 1. The number of people ahead of you is
 2. zero
 3. one

 Please let me know

 Cheers,
 Randall

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[OSL | CCIE_Voice] NTP for UCCX

Hello,

Do we need to set the NTP Server for UCCX Win Server? Or the NTP settings
during the Initial Setup is enough? Well, at least for the exam.

I cannot set the NTP Server for Win2003 (UCCX Server)  -  net time
command does not work, as it is looking for Win Server NTP, and is not
conencting to IOS NTP.
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Re: [OSL | CCIE_Voice] New 5 Labs: Lab 2 UXXC

Prompt  -   Create Generated Prompt, where you have to point the
String which you want to play out (in your case it would be String with
value of 1 or 2 or anything you like to announce/play out.

You have to select the Constructor Type as well, as it rules how the Number
is played out gramaitically - 1 as First, 1 as One and etc, (there are
some Types like Number, Telephone Number and etc. You can get the
documentation for each type in UCCX Scripting Guides, or even in UCCX Scipr
Help Files).

Select the Prompt File where you want to save the Generated Prompt.

You can also choose the language, or leave it as it is.

After you have to use the Play Prompt step to play previously
generated Prompt.

On Sun, Feb 12, 2012 at 5:45 PM, George Goglidze gogli...@gmail.com wrote:

  There is a step to generate a number... M not in front of UCCX right now
 so can't tell you exact name of the step, but it is very easy to find it.

 In this step you tell it what you want it to gemerate, in your case
 number, and then provide it with the value, and then output prompt.

 Sent from my iPad

 On 11 Feb 2012, at 21:09, Randall Crumm rrcr...@yahoo.com wrote:

   HI,
 I know the steps I need to do but I am unsure of the prompts I need to
 record.

 Please let know what I need to record.

 I believe I need to record:

 1. The number of people ahead of you is
 2. zero
 3. one

 Please let me know

 Cheers,
 Randall

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[OSL | CCIE_Voice] MTP and Xcoders

Hello,

Assume that we have G711 and G729 MTP's in first MRG, and then Xcoder in
the next MRG:

MRGL oder:

1) MRG_MTP

2) MRG_Xcoder

So as we know, CUCM can allocate the required MTP's based on codec -  if
call is going to be g711 then the G711 MTP will be allocated, and if the
call is going to be G729, then G729 MTP will be allocated.

But it is also possible for G729 call to use G711 MTP but with Transcoder
support  - IP Phone G729 to Xcoder G729, Xcoder G711 to MTP G711.



So when the G729 call comes to CUCM, will the CUCM allocate G729 MTP, or
will try to allocated G711 with Transacoder?

Here is also another example, with the same question:

MRGL order:

1) MRG_MTP_G711

2)MRG_MTP_G729

3)MRG_Xcoder

In second example, as the G729 call can use G711 MTP with Transcoder
support, will the CUCM allocate G711 MTP along with Transcoder, as the G711
MTP is first in order, or it will look down and allocate G729 MTP?
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[OSL | CCIE_Voice] MMoH RSVP CAC

2012-02-11 Thread datucha123 datucha123

As we know, the Multicast MoH is not counted against the CAC bandwidth, but
the Priority Queue does. So when the Branch devices (IP Phones/Gateway) are
using the MMoH sourced from the HQ CUCM Servers, we need to take the MoH
bandwidth into account for LLQ Priority queue, but not for CAC, as
mutlicast is not counted for RSVP CAC. And at some point we may get the
Oversubscriptio for the WAN link.

Well, that's ok, but how much bandwidth must be taken into acount for LLQ
Priority queue for Multicast MoH? I mean if using G729 MMoH, then how much
MMoH sessions we have to consider when configuring the LLQ Priority Queue?
What it is based on? Becuase we do not know, how much Holds will be at the
same time on the Branch site.
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Re: [OSL | CCIE_Voice] OT - UCM 7x bugs?

That is the Browser issue.



On Fri, Feb 10, 2012 at 5:27 AM, Jeferson Guardia jefers...@gmail.comwrote:

 Hi,

 Often when doing labs over ucm 7, sometimes I notice specially when
 messing around with CSS/partitions, if you open multiple windows and for
 example:

 You have 5 firefox tabs open

 CSS-site-A
 css-B
 css-C
 css-D
 css-E

 And then you go window by window applying it and saving it, all at once..
 I usually find partitions missing later on that CSS previously assigned. I
 care to check for dbreplication and issues like that, all running fine. is
 this a bug or something? it's terrible to have to go back at something you
 sure you've done correctly it just to fix a bug!

 --
 Jeferson Guardia
 CCIE #28157

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Re: [OSL | CCIE_Voice] MVA problem

The issue nas been with RSVP CAC, and as soon as I have remove the CAC, the
MoH started to work.

Q1: Can you confirm that when you answer the cell phone (working scenario)
which gateway is being used to get to PSTN
A1:  The BR1 gateway was used.

Q2: Can you confirm that when you answer the deskphone (not working
scenario) and transfer to cell phone which gateway is being used to get to
PSTN.
A2:  HQ Gateway was used.


On Fri, Feb 10, 2012 at 5:25 AM, Vik Malhi vma...@ipexpert.com wrote:

  I found the problem, but another one has arise.

 Please inform us of the problem/solution for the benefit of others
 following this thread.


 Q1: Can you confirm that when you answer the cell phone (working scenario)
 which gateway is being used to get to PSTN.

 Q2: Can you confirm that when you answer the deskphone (not working
 scenario) and transfer to cell phone which gateway is being used to get to
 PSTN.

 I suspect there is different gateway being used due to you using SLRG
 which could explain the differences.



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

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




   On Feb 9, 2012, at 9:24 AM, datucha123 datucha123 wrote:

   I found the problem, but another one has arise.

 I am using RSVP as CAC between site, where the RSVP MTP has only G729
 codec enabled. And also the MoH is using Unicast so that it is subject to
 CAC.

 So now when the call is picked up by Mobile Phone and then dropped, the
 MoH plays good, as the BR1 site is using G729 to MoH Server. But when the
 HQ Desk Phone picks up the call first and then redirects to Mobile phone,
 after hung up the MoH does not work as the BR1 phone (somehow, why it makes
 so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP
 does not pass the G711 traffic and that is why the ToH is heard.

 Now when I have changed the BR1 to use the Local Flash MMoH, everything
 was working fine (MoH was heard always).

 But still no idea, why the BR1 is trying to negotiate G711 to MoH server
 after the Mobile has dropped the call (when the Desk had sent the call to
 Mobile).

 Also I can see that when the Mobile Phone picks up the call first, the BR1
 Phone is using G711 (call goes through SLRG BR1 local gateway).
 But when the HQ Phone picks up the call first, and then sends the call to
 Mobile, the BR1 phone shows that it is using G729, thus the call is going
 though local Gateway.




 On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:

  Hi Datucha,

 ** **

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects. 

 I suggest to set two different network moh source files. Assign one to
 RDP and one to HQ Phone. Then repeat your test scenarios above and lets
 know which MoH file is used in each case.

 ** **

 Also, please share the traces regarding the codec part.

 ** **

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Wednesday, February 08, 2012 11:28 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA problem

 ** **

 I have the following kind of probem:

 I am using SLRG for Mobile Connect calls, so that that calls to users
 mobiles are done through local gateway (this is just for test).

 Now, the HQ phone has the RDP assinged with RD of his mobile phone.

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects. 

  

 Also I have noticed that when the Mobile Phone picks up the call faster
 then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
 local gateway through which the call went out.

 But if the Desk Phone at HQ  picked up the call first, and then made Send
 calls to mobile phone, the codec stays at G729 on BR1 phone, even though
 the call is going out throuhg local BR1 gateway where it should use G711.
 


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Re: [OSL | CCIE_Voice] MVA problem

Check whats the region relation between HQ MTP RSPV and MoH. It should be
g729. Also same between BR1 MTP RSVP and HQ MTP RSVP

BR1 Phones and MTP are placed in BR1 Region, which has G711 inside and G729
outside to HQ Region and MoH Regions as well.
HQ Phones and MTP are placed in HQ Region, which has G711 inside and G711
to MoH, while G729 to BR1.

Well, I can see the HQ Region (where the HQ MTP is placed) has G711 towards
MoH Regions, but as I know that does not mean that only G711 can be
negotiated between MTP and MoH Server (everything below G711 could be). So
MTP at HQ must negotiate MoH G729.



On Fri, Feb 10, 2012 at 6:43 AM, Mohd Baqari baqari.voic...@gmail.comwrote:

  Check whats the region relation between HQ MTP RSPV and MoH. It should
 be g729. Also same between BR1 MTP RSVP and HQ MTP RSVP.

 If this is done then MoH should negotiate g729 with BR1.

 Regards,
 Mohammed Al Baqari

 Sent from my iPhone

 On Feb 9, 2012, at 9:52 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Here is what I get on HQ Router during the call hung up on the Mobile
 Phone:

 Feb 10 03:48:06.063: %ISDN-6-DISCONNECT: Interface Serial0/2/0:0
 disconnected from 206501 , call lasted 18 seconds
 Feb 10 03:48:06.187: RSVP-MSG:
 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]: Received PathTear message
 from 177.1.254.2 (on Serial0/3/0.1)
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring Serial0/3/0.1 PATH state, reason: PathTear arrival
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring  RESV state, reason: PathTear arrival
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring receiver host RESV state, reason: PathTear arrival (17:18762)
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring Serial0/3/0.1 RESV request state, reason: PathTear arrival
 Feb 10 03:48:06.195: RSVP-MSG:
 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]: Received ResvTear message
 from 177.0.101.2 (on Serial0/3/0.1)
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival (17:16464)
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Expiring sender host PATH state, reason: Local application requested tear
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 building hop object with src addr: 177.0.101.1
 Feb 10 03:48:06.215: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Sending PathTear message to 177.1.254.2


 HQ Interface:

 interface Serial0/3/0.1 point-to-point
  description == FR To BR1 (R2)
  ip address 177.0.101.1 255.255.255.0
  snmp trap link-status
  frame-relay interface-dlci 101
  ip rsvp bandwidth 500
 end

 BR1 Interface:

 interface Serial0/3/0.1 point-to-point
  description == FR To HQ (R1)
  ip address 177.0.101.2 255.255.255.0
  snmp trap link-status
  frame-relay interface-dlci 101
  ip rsvp bandwidth 500
 end

 So as I guess, reservation is not taking place after the Mobile Phone has
 hung up.

 Also here is the output before the HQ Desk Phone has sent the call to
 Mobile Phone:

 Feb 10 03:47:39.587: RSVP: session 177.1.254.1_19398[0.0.0.0]: Received
 Resv message from 127.0.0.1 (on receiver host)
 Feb 10 03:47:39.587: RSVP: 177.1.254.2_17452-177.1.254.1_19398[0.0.0.0]:
 Successfully parsed Resv message from 127.0.0.1 (on receiver host)
 Feb 10 03:47:39.587: RSVP-MSG:
 177.1.254.2_17452-177.1.254.1_19398[0.0.0.0]: no matching path state for
 Resv
 Feb 10 03:47:39.595: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Received Path message from 127.0.0.1 (on sender host)
 Feb 10 03:47:39.595: RSVP: new path message passed parsing, continue...
 Feb 10 03:47:39.595: RSVP: Triggering outgoing Path due to incoming Path
 change or new Path
 Feb 10 03:47:39.595: RSVP: Triggering outgoing Path refresh
 Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
 Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh (msec), config: 3 curr: 3 xmit: 3
 Feb 10 03:47:39.599: RSVP: Triggering outgoing Path due to incoming Path
 change or new Path
 Feb 10 03:47:39.599: RSVP: Triggering outgoing Path refresh
 Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
 Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh (msec), config: 3 curr: 3 xmit: 3
 Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Sending Path message to 177.1.254.2
 Feb 10 03:47:39.623: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 building hop object with src addr

Re: [OSL | CCIE_Voice] MVA problem

I have also tryed to reconfigure the Regions so:

HQ MTP placed into separate Region, which has G729 to everywhere (HQ
devices, included MoH, BR1 devices).

But still not success, the ToH is heard.

On Fri, Feb 10, 2012 at 6:43 AM, Mohd Baqari baqari.voic...@gmail.comwrote:

  Check whats the region relation between HQ MTP RSPV and MoH. It should
 be g729. Also same between BR1 MTP RSVP and HQ MTP RSVP.

 If this is done then MoH should negotiate g729 with BR1.

 Regards,
 Mohammed Al Baqari

 Sent from my iPhone

 On Feb 9, 2012, at 9:52 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Here is what I get on HQ Router during the call hung up on the Mobile
 Phone:

 Feb 10 03:48:06.063: %ISDN-6-DISCONNECT: Interface Serial0/2/0:0
 disconnected from 206501 , call lasted 18 seconds
 Feb 10 03:48:06.187: RSVP-MSG:
 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]: Received PathTear message
 from 177.1.254.2 (on Serial0/3/0.1)
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring Serial0/3/0.1 PATH state, reason: PathTear arrival
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring  RESV state, reason: PathTear arrival
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring receiver host RESV state, reason: PathTear arrival (17:18762)
 Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762-177.1.254.1_16464[0.0.0.0]:
 Expiring Serial0/3/0.1 RESV request state, reason: PathTear arrival
 Feb 10 03:48:06.195: RSVP-MSG:
 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]: Received ResvTear message
 from 177.0.101.2 (on Serial0/3/0.1)
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival (17:16464)
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Expiring sender host PATH state, reason: Local application requested tear
 Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 building hop object with src addr: 177.0.101.1
 Feb 10 03:48:06.215: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
 Sending PathTear message to 177.1.254.2


 HQ Interface:

 interface Serial0/3/0.1 point-to-point
  description == FR To BR1 (R2)
  ip address 177.0.101.1 255.255.255.0
  snmp trap link-status
  frame-relay interface-dlci 101
  ip rsvp bandwidth 500
 end

 BR1 Interface:

 interface Serial0/3/0.1 point-to-point
  description == FR To HQ (R1)
  ip address 177.0.101.2 255.255.255.0
  snmp trap link-status
  frame-relay interface-dlci 101
  ip rsvp bandwidth 500
 end

 So as I guess, reservation is not taking place after the Mobile Phone has
 hung up.

 Also here is the output before the HQ Desk Phone has sent the call to
 Mobile Phone:

 Feb 10 03:47:39.587: RSVP: session 177.1.254.1_19398[0.0.0.0]: Received
 Resv message from 127.0.0.1 (on receiver host)
 Feb 10 03:47:39.587: RSVP: 177.1.254.2_17452-177.1.254.1_19398[0.0.0.0]:
 Successfully parsed Resv message from 127.0.0.1 (on receiver host)
 Feb 10 03:47:39.587: RSVP-MSG:
 177.1.254.2_17452-177.1.254.1_19398[0.0.0.0]: no matching path state for
 Resv
 Feb 10 03:47:39.595: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Received Path message from 127.0.0.1 (on sender host)
 Feb 10 03:47:39.595: RSVP: new path message passed parsing, continue...
 Feb 10 03:47:39.595: RSVP: Triggering outgoing Path due to incoming Path
 change or new Path
 Feb 10 03:47:39.595: RSVP: Triggering outgoing Path refresh
 Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
 Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh (msec), config: 3 curr: 3 xmit: 3
 Feb 10 03:47:39.599: RSVP: Triggering outgoing Path due to incoming Path
 change or new Path
 Feb 10 03:47:39.599: RSVP: Triggering outgoing Path refresh
 Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
 Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Path refresh (msec), config: 3 curr: 3 xmit: 3
 Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 Sending Path message to 177.1.254.2
 Feb 10 03:47:39.623: RSVP: 177.1.254.1_19398-177.1.254.2_17452[0.0.0.0]:
 building hop object with src addr: 177.0.101.1
 Feb 10 03:47:39.663: RSVP: 177.1.254.2_17452-177.1.254.1_19398[0.0.0.0]:
 Received Path message from 177.0.101.2 (on Serial0/3/0.1)
 Feb 10 03:47:39.663: RSVP: new path message passed parsing, continue...
 Feb 10 03:47:39.663: RSVP: session 177.1.254.1_19398[0.0.0.0]: Received
 Resv message from 127.0.0.1 (on receiver host)
 Feb 10 03:47:39.663: RSVP: 177.1.254.2_17452-177.1.254.1_19398[0.0.0.0

Re: [OSL | CCIE_Voice] MVA problem

As I have mentioned, the removing CAC is not a solution.

So i will try to continue figuring it out and as soon as I will get
solution, I will definitely inform.

All my Regions setting are correct. I think there is something wrong with
Gateways, as different Gateways are selected in each case.

On Fri, Feb 10, 2012 at 5:25 AM, Vik Malhi vma...@ipexpert.com wrote:

  I found the problem, but another one has arise.

 Please inform us of the problem/solution for the benefit of others
 following this thread.


 Q1: Can you confirm that when you answer the cell phone (working scenario)
 which gateway is being used to get to PSTN.

 Q2: Can you confirm that when you answer the deskphone (not working
 scenario) and transfer to cell phone which gateway is being used to get to
 PSTN.

 I suspect there is different gateway being used due to you using SLRG
 which could explain the differences.



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

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




   On Feb 9, 2012, at 9:24 AM, datucha123 datucha123 wrote:

   I found the problem, but another one has arise.

 I am using RSVP as CAC between site, where the RSVP MTP has only G729
 codec enabled. And also the MoH is using Unicast so that it is subject to
 CAC.

 So now when the call is picked up by Mobile Phone and then dropped, the
 MoH plays good, as the BR1 site is using G729 to MoH Server. But when the
 HQ Desk Phone picks up the call first and then redirects to Mobile phone,
 after hung up the MoH does not work as the BR1 phone (somehow, why it makes
 so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP
 does not pass the G711 traffic and that is why the ToH is heard.

 Now when I have changed the BR1 to use the Local Flash MMoH, everything
 was working fine (MoH was heard always).

 But still no idea, why the BR1 is trying to negotiate G711 to MoH server
 after the Mobile has dropped the call (when the Desk had sent the call to
 Mobile).

 Also I can see that when the Mobile Phone picks up the call first, the BR1
 Phone is using G711 (call goes through SLRG BR1 local gateway).
 But when the HQ Phone picks up the call first, and then sends the call to
 Mobile, the BR1 phone shows that it is using G729, thus the call is going
 though local Gateway.




 On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:

  Hi Datucha,

 ** **

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects. 

 I suggest to set two different network moh source files. Assign one to
 RDP and one to HQ Phone. Then repeat your test scenarios above and lets
 know which MoH file is used in each case.

 ** **

 Also, please share the traces regarding the codec part.

 ** **

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Wednesday, February 08, 2012 11:28 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA problem

 ** **

 I have the following kind of probem:

 I am using SLRG for Mobile Connect calls, so that that calls to users
 mobiles are done through local gateway (this is just for test).

 Now, the HQ phone has the RDP assinged with RD of his mobile phone.

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects. 

  

 Also I have noticed that when the Mobile Phone picks up the call faster
 then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
 local gateway through which the call went out.

 But if the Desk Phone at HQ  picked up the call first, and then made Send
 calls to mobile phone, the codec stays at G729 on BR1 phone, even though
 the call is going out throuhg local BR1 gateway where it should use G711.
 


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

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



___
For more information regarding industry leading CCIE Lab training

Re: [OSL | CCIE_Voice] MVA problem

G729 Codec is enable in IP Voice Media Streaming Application Service
Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
MoH is played with G729 to BR1 phone without a problem.



On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi vma...@ipexpert.com wrote:

 My guess is MOH is not working when BR1 phones is placed on hold- and this
 has nothing to do with Mobile Connect.

 Enable G729 in the IP Voice Media Streaming Application Service Parameter.

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

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




 On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote:

 Hi Datucha,
 ** **

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects.
 I suggest to set two different network moh source files. Assign one to RDP
 and one to HQ Phone. Then repeat your test scenarios above and lets know
 which MoH file is used in each case.
 ** **
 Also, please share the traces regarding the codec part.
 ** **
 ** **
 Regards,
 Mohammed Al Baqari
 ** **
 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Wednesday, February 08, 2012 11:28 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA problem
 ** **

 I have the following kind of probem:

 I am using SLRG for Mobile Connect calls, so that that calls to users
 mobiles are done through local gateway (this is just for test).

 Now, the HQ phone has the RDP assinged with RD of his mobile phone.

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects.

  

 Also I have noticed that when the Mobile Phone picks up the call faster
 then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
 local gateway through which the call went out.

 But if the Desk Phone at HQ  picked up the call first, and then made Send
 calls to mobile phone, the codec stays at G729 on BR1 phone, even though
 the call is going out throuhg local BR1 gateway where it should use G711.*
 ***
 ___

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

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



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

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

Re: [OSL | CCIE_Voice] MVA problem

I found the problem, but another one has arise.

I am using RSVP as CAC between site, where the RSVP MTP has only G729 codec
enabled. And also the MoH is using Unicast so that it is subject to CAC.

So now when the call is picked up by Mobile Phone and then dropped, the MoH
plays good, as the BR1 site is using G729 to MoH Server. But when the HQ
Desk Phone picks up the call first and then redirects to Mobile phone,
after hung up the MoH does not work as the BR1 phone (somehow, why it makes
so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP
does not pass the G711 traffic and that is why the ToH is heard.

Now when I have changed the BR1 to use the Local Flash MMoH, everything was
working fine (MoH was heard always).

But still no idea, why the BR1 is trying to negotiate G711 to MoH server
after the Mobile has dropped the call (when the Desk had sent the call to
Mobile).

Also I can see that when the Mobile Phone picks up the call first, the BR1
Phone is using G711 (call goes through SLRG BR1 local gateway).
But when the HQ Phone picks up the call first, and then sends the call to
Mobile, the BR1 phone shows that it is using G729, thus the call is going
though local Gateway.




On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari baqari.voic...@gmail.com
 wrote:

 Hi Datucha,

 ** **

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects. 

 I suggest to set two different network moh source files. Assign one to RDP
 and one to HQ Phone. Then repeat your test scenarios above and lets know
 which MoH file is used in each case.

 ** **

 Also, please share the traces regarding the codec part.

 ** **

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Wednesday, February 08, 2012 11:28 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA problem

 ** **

 I have the following kind of probem:

 I am using SLRG for Mobile Connect calls, so that that calls to users
 mobiles are done through local gateway (this is just for test).

 Now, the HQ phone has the RDP assinged with RD of his mobile phone.

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects. 

  

 Also I have noticed that when the Mobile Phone picks up the call faster
 then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
 local gateway through which the call went out.

 But if the Desk Phone at HQ  picked up the call first, and then made Send
 calls to mobile phone, the codec stays at G729 on BR1 phone, even though
 the call is going out throuhg local BR1 gateway where it should use G711.*
 ***

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

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

Re: [OSL | CCIE_Voice] MVA problem

Also as Vik had asked, I have removed the RSVP CAC, and the MoH begin to
work and the G729 was negotiated.

Well this is very strange behavior.

While using CAC, BR1 is trying to negotiate G711 to MoH, and when not using
RSVP CAC, G729 is negotiated.

On Thu, Feb 9, 2012 at 9:16 PM, Vik Malhi vma...@ipexpert.com wrote:

 The only difference being you have an existing call in the case of the RD
 and the MOH would be the second call. Can you try removing Locations CAC
 altogether to rule this out.


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

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




 On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote:

 G729 Codec is enable in IP Voice Media Streaming Application Service
 Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
 MoH is played with G729 to BR1 phone without a problem.



 On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi vma...@ipexpert.com wrote:

 My guess is MOH is not working when BR1 phones is placed on hold- and
 this has nothing to do with Mobile Connect.

 Enable G729 in the IP Voice Media Streaming Application Service Parameter.

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

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




 On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote:

 Hi Datucha,
 ** **

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects.
 I suggest to set two different network moh source files. Assign one to
 RDP and one to HQ Phone. Then repeat your test scenarios above and lets
 know which MoH file is used in each case.
 ** **
 Also, please share the traces regarding the codec part.
 ** **
 ** **
 Regards,
 Mohammed Al Baqari
 ** **
 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Wednesday, February 08, 2012 11:28 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA problem
 ** **

 I have the following kind of probem:

 I am using SLRG for Mobile Connect calls, so that that calls to users
 mobiles are done through local gateway (this is just for test).

 Now, the HQ phone has the RDP assinged with RD of his mobile phone.

 Now when the BR1 phone calls this HQ phone, so that the mobile phone also
 ring, and when the Mobile Phone picks up the call first, then the MoH works
 when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects.

  

 Also I have noticed that when the Mobile Phone picks up the call faster
 then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
 local gateway through which the call went out.

 But if the Desk Phone at HQ  picked up the call first, and then made Send
 calls to mobile phone, the codec stays at G729 on BR1 phone, even though
 the call is going out throuhg local BR1 gateway where it should use G711.
 
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Re: [OSL | CCIE_Voice] MVA problem

]:
Resv changed: POLICY_DATA, FLOWSPEC,
Feb 10 03:47:48.703: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
process_reservation_change: Resv change requires triggering of Resv upstream
R1#
Feb 10 03:47:48.703: RSVP-RESV: accept_reservation_change: 4A374C78
Feb 10 03:47:48.703: RSVP-RESV: reservation was installed: 4A374C78
Feb 10 03:47:48.707: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
building error_spec object with err-node addr: 177.0.101.1
Feb 10 03:47:48.727: RSVP: 177.1.254.1_16464-177.1.254.2_18762[0.0.0.0]:
Sending ResvConf message to 177.0.101.2




On Thu, Feb 9, 2012 at 9:40 PM, Vik Malhi vma...@ipexpert.com wrote:

 Put the CAC back on and increase the ip rsvp bandwidth to a high value
 such as 500 (on both routers).

 Do a *debug ip rsvp signaling* on HQ and find out how much bandwidth is
 being requested when the problematic call is on hold.

 Also what Device Pool is the MOH server in? Place it inside the HQ Device
 Pool / HQ Region.

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

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




 On Feb 9, 2012, at 9:27 AM, datucha123 datucha123 wrote:

 Also as Vik had asked, I have removed the RSVP CAC, and the MoH begin to
 work and the G729 was negotiated.

 Well this is very strange behavior.

 While using CAC, BR1 is trying to negotiate G711 to MoH, and when not
 using RSVP CAC, G729 is negotiated.

 On Thu, Feb 9, 2012 at 9:16 PM, Vik Malhi vma...@ipexpert.com wrote:

 The only difference being you have an existing call in the case of the RD
 and the MOH would be the second call. Can you try removing Locations CAC
 altogether to rule this out.


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

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




 On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote:

 G729 Codec is enable in IP Voice Media Streaming Application Service
 Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
 MoH is played with G729 to BR1 phone without a problem.



 On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi vma...@ipexpert.com wrote:

 My guess is MOH is not working when BR1 phones is placed on hold- and
 this has nothing to do with Mobile Connect.

 Enable G729 in the IP Voice Media Streaming Application Service
 Parameter.

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

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




 On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote:

 Hi Datucha,
 ** **

 Now when the BR1 phone calls this HQ phone, so that the mobile phone
 also ring, and when the Mobile Phone picks up the call first, then the MoH
 works when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects.
 I suggest to set two different network moh source files. Assign one to
 RDP and one to HQ Phone. Then repeat your test scenarios above and lets
 know which MoH file is used in each case.
 ** **
 Also, please share the traces regarding the codec part.
 ** **
 ** **
 Regards,
 Mohammed Al Baqari
 ** **
 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
 datucha123
 *Sent:* Wednesday, February 08, 2012 11:28 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA problem
 ** **

 I have the following kind of probem:

 I am using SLRG for Mobile Connect calls, so that that calls to users
 mobiles are done through local gateway (this is just for test).

 Now, the HQ phone has the RDP assinged with RD of his mobile phone.

 Now when the BR1 phone calls this HQ phone, so that the mobile phone
 also ring, and when the Mobile Phone picks up the call first, then the MoH
 works when the mobile is hunged up, on the BR1 phone.

 But if the HQ phone has picked up the call first and then made the Send
 Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
 instead the ToH is heard when the mobile Phone disconnects.

  

 Also I have noticed that when the Mobile Phone picks up the call faster
 then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
 local gateway through which the call went out.

 But if the Desk Phone at HQ  picked up the call first, and then made
 Send calls to mobile phone, the codec stays at G729 on BR1 phone, even
 though the call is going out throuhg local BR1 gateway where it should use
 G711.
 ___

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 please visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] MGCP troubleshooting

2012-02-08 Thread datucha123 datucha123

debug mgcp packets

Trace in CUCM

On Wed, Feb 8, 2012 at 3:35 PM, Jeferson Guardia jefers...@gmail.comwrote:

 Hi,

 What are the techniques most used to perform MGCP troubleshooting?

 Yesterday I was doing a lab and had a router with pstn integration, it was
 set for a CSS where my phones had visibility and significant digits = 4.
 But whenever I would call out from the PSTN, I would get a second dial
 tone, I would see the call kicking in thru debug isdn q931 but my phone
 would simply not ring. Any ideas how to verify that possible behavior ?
 Articles? tech guides?

 Thanks,

 --
 Jeferson Guardia
 CCIE #28157

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[OSL | CCIE_Voice] MVA problem

2012-02-08 Thread datucha123 datucha123

I have the following kind of probem:

I am using SLRG for Mobile Connect calls, so that that calls to users
mobiles are done through local gateway (this is just for test).

Now, the HQ phone has the RDP assinged with RD of his mobile phone.

Now when the BR1 phone calls this HQ phone, so that the mobile phone also
ring, and when the Mobile Phone picks up the call first, then the MoH works
when the mobile is hunged up, on the BR1 phone.

But if the HQ phone has picked up the call first and then made the Send
Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
instead the ToH is heard when the mobile Phone disconnects.



Also I have noticed that when the Mobile Phone picks up the call faster
then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
local gateway through which the call went out.

But if the Desk Phone at HQ  picked up the call first, and then made Send
calls to mobile phone, the codec stays at G729 on BR1 phone, even though
the call is going out throuhg local BR1 gateway where it should use G711.
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Re: [OSL | CCIE_Voice] Lab Grading

2012-02-07 Thread datucha123 datucha123

You are right Emanuel.

I am a native neglish speaker, and sometimes it is hard for me to
understand some task, while rereading them several times

On Tue, Feb 7, 2012 at 8:12 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:

 Hey Vega,

 Sorry for hearing you didn't make it. I've been there and I know the
 feeling. I am preparing myself for my second attempt now.

 I don't know which strategies you used, but all areas you mentioned there
 are tricks that confuse us. I think the whole point of the exam is
 understanding and doing its requirements. I have Workbooks 1 and 2 of
 IPExpert and the whole she bang from INE. I speak fluent English, but I
 am aware that it is not my first language. Sometimes I read a question and
 I understand it some way, when I ask my wife to read it she understands it
 another way, and she explain to me what the Workbooks are actually asking
 of me. Since she is American born and raised, she is also my resource as a
 better understanding simple plain English. I am not trying to say you know
 or don't know the language, it's not that. What I am judging by your last
 name you are not an English native speaker (I might be wrong), but if
 English is not your native language, you may have an understanding that is
 different from what it is required. Thus I mentioned my example, because
 that is my case. If you are an English native speaker, nevermind what I
 just said. But I think that even re-reading it, makes better sense.

 In the exam, we don't have time to read and re-read questions until we
 understand. You are not in the exam anymore, but take a question from one
 of your Workbook studies, and pick one you didn't understand it fully. You
 can easily pick that question as you read through your workbook and you
 read a question that you stop and say to yourself Wait, what is it this
 question is saying?. You automatically re-read it for better
 understanding. As you read it twice or thrice, you realize that you are
 understanding it quite differently from the first time you read it.

 My wife is a great support, not only for leaving me alone to study but
 also to help me understand better the English on the workbooks. The exam is
 all about being literal, over configuring won't do you any good. The
 proctor won't think you're an expert because you know a command that only
 few people know it. If the question is specifically saying to do something,
 you need to do it like they ask. If they don't specify it, you can do it
 the way you know. I guess you already know all of this, but I am posting
 this mostly for the CCIE newcomers. But it might work for you too, if you
 don't already know it :)

 Best regards to all.
 *Emanuel Damasceno*
 CCNP Voice





 On Tue, Feb 7, 2012 at 4:33 AM, Vega Wong vega2...@yahoo.com.au wrote:

 Hi all

 First of all, let me clarify that I am not trying to break NDA or trying
 to sell anything here, just something I cant get my head around with.

 I just had my second attmpt yesterday, and I failed. What really
 demoralising for me is that I thought I was fully prepared. What even more
 demoralising is that when I looked at the score card, I didnt get 100% in
 the area I thought I would get a 100%.

 I dont mean to say I know it all (especially now), but there are areas on
 my score card that really shocks me. As I read through the exam, I know
 what the questions were requiring (at least I thought I did). For example,
 we all know that the topics of DHCP, NTP, VLAN will be in the network
 infrastructure area. And what could go wrong in those areas? I mean, if you
 dont setup those area correctly, all your subsequence config will has
 problem right? But yet, I dont get full marks in those. Similarly the
 gateway area, QoS area etc.

 Right now I am pretty lost as to what I can prepare or study on, or at
 least how to check my config? I know this sounds really bad (or arrogent)
 but I really have no idea as how to confirm the config I did can get the
 full mark in those area.

 I guess I am hoping if someone that share their strategy as to how to
 confirm their work is good or fulfilled the requirement?

 hope that makes sense?




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Re: [OSL | CCIE_Voice] QOS LFI and BACD files

2012-02-07 Thread datucha123 datucha123

But that won't go for HQ router, as it is a Hub Router  -- multiple PVCs
per physical interfaces via subinterfaces.

For instance, if you have a T1 frame relay link, and already shaped one PVC
to 500kbps, you cannot shape another PVC to 1536. You have to shape another
PVC to 1536 - 500 = 1036 kbps, as the full interface rate is 1536, while
one PVC is already shaped up to 500, so only 1036 is left for shaping.

On Tue, Feb 7, 2012 at 10:02 PM, Vik Malhi vma...@ipexpert.com wrote:

 I agree with the last post.

 When you have used FRTS use the command show traff to verify Interval
 time and target rate- default to 1536 if PVC speed is not given to you. The
 snippet below shows what happens when FRTS is enabled- both these PVC's
 will need fixing.


 SiteA-RTR(config)#*interface Serial0/0/1:0*
 SiteA-RTR(config-if)#*frame-relay traff*
 SiteA-RTR#*sh traff*

 Interface   Se0/0/1:0.1
Access TargetByte   Sustain   ExcessInterval  Increment
 Adapt
 VC List   Rate  Limit  bits/int  bits/int  (ms)  (bytes)
 Active
 201   *56000* 8757000  0 *125*   875
   -

 Interface   Se0/0/1:0.2
Access TargetByte   Sustain   ExcessInterval  Increment
 Adapt
 VC List   Rate  Limit  bits/int  bits/int  (ms)  (bytes)
 Active
 202   *56000* 8757000  0 *125*   875
   -

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

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




 On Feb 7, 2012, at 7:58 AM, CCIEVoiceKP wrote:

 I personally would set it to a full T1 ... Bandwidth 1536  When in
 doubt, explain what and why to the proctor to make sure it's ok.

 KP

 Sent from my iPhone and I have big thumbs ... So please excuse the typos.

 On Feb 6, 2012, at 9:02 PM, AJ BG ciscoie2...@gmail.com wrote:

  Hello,
1.   QOS question
 According to Vic, if you configure LFI for a subinterface in a hub and
 spoke environment, Your second sub interface will dopes its CIR to 56k. To
 solve this issue you should configure map-class for the second interface as
 well. I have tested this and confirmed the problem and the solution.
  But if the interface bandwidth is not given to you, then in what rate do
 you configure the second map-class? What should be your CIR and MinCIR
 bandwidth?


 2.   BACD question

 will it be possible that the lab requirement will be  to configure BACD
 without giving you direct access to the BACD files? If the above scenario
 happen then how would you copy the files into the router. I am thinking to
 use CUCM. But can you even go to Cisco’s website and download BACD tar file
 during the exam? Any suggestion?

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Re: [OSL | CCIE_Voice] OUTVIA INVIA

2012-02-07 Thread datucha123 datucha123

Outvia is more accurate.

Invia, in most cases, is used for incoming LRQs.

On Tue, Feb 7, 2012 at 11:13 PM, mercy forall mercy_for_...@hotmail.comwrote:

  Hi All

 now in outvia and invia ,,

 Are is it deference if i use it in local zone or remote zone ?

 As per Doc, outvia for any traffic leave this zone , so are this same if i
 use outvia in local or remote zone


 I  need to send the call form local zone to remote zone through CUBE as
 local zone ,

 what is the correct  [zone remote with outvia OR with invia CUBE ] ?

 thanks
 **

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Re: [OSL | CCIE_Voice] Timezone for UCM

2012-02-06 Thread datucha123 datucha123

That is for production installtion, but what about the LAB Exam?

On Mon, Feb 6, 2012 at 10:47 PM, William Bell w...@netcraftsmen.net wrote:

  As noted by Ken, you typically set the timezone during the installation
 process. If you need to adjust the timezone post-install then you can do so
 from the cli:

  admin:set timezone ?
 Syntax:
 set timezone zone
 zone mandatory   This is the new time zone. Enter the appropriate
 string
  or zone index id to uniquely identify the timezone.
  A list of valid time zones can be obtained via the
  following CLI command: show timezone list.


 Regards,
 Bill

  On Feb 6, 2012, at 11:20 AM, datucha123 datucha123 wrote:

  Hello,

 If we are told to synchronize CUCM Pub server with some NTP, do we need to
 set the correct Timezone for CUCM OS as well? Or just Date/Time Groups for
 Phones are enough?
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Re: [OSL | CCIE_Voice] Standard Local RG troubleshooting: RouteGroup :RouteGroup Name= Standard Local Route Group

2012-02-06 Thread datucha123 datucha123

stop routing on unallocated number flag  -  in this particular case, this
parameter has nothing to do with the actual problem. This parameter defines
the rerouting option as William has already mentioned.

Ricardo, try to set the Digit Analysis Complexity to Translation and
Alternate Pattern Analysis. And try to look for CUCM Traces, not the DNA.

On Tue, Feb 7, 2012 at 6:37 AM, William Bell w...@netcraftsmen.net wrote:

 Ricardo,

 IIRC, the stop routing on unallocated number flag was actually first
 introduced for ICT call flows. However, it can be applied to other call
 flows. In normal call handling, when the CUCM receives a notification that
 a call failed to complete due to unallocated number it will stop routing
 the call. When you flag this service param to false, CUCM will try the next
 trunk or gateway in the route list/route group.

 -Bill

  On Feb 6, 2012, at 8:21 PM, Ricardo Palaver wrote:

  Hi Emanuel !.

 No , it does not work .. As far as I know, this is for use AAR, or Am I
 wrong?

 Thanks  !

  --
 Date: Mon, 6 Feb 2012 23:01:14 -0200
 Subject: Re: [OSL | CCIE_Voice] Standard Local RG troubleshooting:
 RouteGroup :RouteGroup Name= Standard Local Route Group
 From: aedamasc...@gmail.com
 To: ricardo.pala...@hotmail.com
 CC: ccie_voice@onlinestudylist.com

 Hello Ricardo,

 You need to go to Service Parameters  Call Manager, and set the option *Stop
 Routing on Unallocated Number Flag* to FALSE

 I hope this helps :)
 *Emanuel Damasceno*
 CCNP Voice





 On Mon, Feb 6, 2012 at 10:05 PM, Ricardo Palaver 
 ricardo.pala...@hotmail.com wrote:

  Hi Folks,

 I am facing a problem with Standard local route group ...,  it is not
 working and I have no idea where could I troubleshoot it.  I configured as
 usual ... ,
 RL - Standard Local RG
 In each DP, I pointed to the respective gateway (using Local  Route Group
 param) and of course each phone with the respective device.

 I tried  by using DNA, but it does not go further , the last point I see
 is  RouteGroup :RouteGroup Name= Standard Local Route Group ..., As far
 as I know there is nothing in the service param to enable ... or there are
 something ?


 Thanks all !




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  *
 *

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Re: [OSL | CCIE_Voice] Reorder tone while + dialing from directories(missed, received)

You can also leave the Urgent Priority for Patterns, but use the SIP
Dial-rules for SIP phones, with \+! and interdigit timeout with 0

On Sun, Feb 5, 2012 at 12:53 AM, Ashwani ash_r...@hotmail.com wrote:

 Thanks Vik.  Yes now I am seeing inter-digit timeout dialing from missed
 and received calls.  Appreciate your help and pointing me to the right
 direction.

 Ashwani


 On 2/4/2012 3:45 PM, Vik Malhi wrote:

  From SIP phones calls from the directory are sent digit by digit. This
 is in contrast to sccp phones which send digits en bloc (as opposed to
 digit by digit).

 A route pattern such as : \+! marked as urgent priority would cause a
 call from the directory from a sip phone to fail. Since the plus would
 match and since the urgent priority has been selected the call would get
 sent to the gateway with just a + (which would be stripped in the case of
 an h323 gateway).

 Remove and plus route patterns marked as urgent priority- if you want to
 avoid inter digit timeout create route patterns without the ! and define
 the exact number of digits (x's).




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

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

And I make restart of CUC server every time.

Thus I was using SCCP integration.

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

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

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

 Thanks,

 Edgar

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

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



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

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


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

   So I am good then?

 Sent from my iPhone

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

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

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



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

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, 
 Audio
 Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
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 training
 locations throughout the United States, Europe, South Asia and Australia.
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 On Feb 4, 2012, at 11:42 AM, William Affeldt william.affe...@yahoo.com
 william.affe...@yahoo.com wrote:

   You are having one way audio issues then.

 Sent from my iPhone

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

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

 E

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

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

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

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

 Thanks

 E

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Re: [OSL | CCIE_Voice] CUE Live Reply

You have press 44 (twise digit 4) and it will dial out.

On Sun, Feb 5, 2012 at 1:50 AM, John McGaughey (jomcgaug) 
jomcg...@cisco.com wrote:

  Hello,

 ** **

 I’m in lab 4 of the new 5 labs.  Question 6.4.  It’s asking to configure
 Live Reply.  The DSG says to just click the check box “Enable Live Reply.”
 I’ve done so but when I press option 4 it says “this message cannot get a
 reply.”  I’m calling from the SiteC phone 1.

 ** **

 I added a user and mailbox for SiteC phone 1.  Now when I press option 4 I
 get an option to record a message for SiteC phone.  I cannot figure out how
 to get it to dial the caller back.

 ** **

 I’ve googled and found nothing.  Any ideas?

 ** **

 John

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Re: [OSL | CCIE_Voice] CFUR does not work

No. it is supported. The destination Phone will just ring a bit later
through PSTN.

On Sun, Feb 5, 2012 at 3:16 AM, John McGaughey (jomcgaug) 
jomcg...@cisco.com wrote:

  Hi Vik/All

 ** **

 I’m working on Lab #4 of the new 5 labs.  Quesiton 9.2.  They are asking
 you to configure CFUR on SiteB phone 2.  However this will not work because
 of the RDP assigned to the phone.

 ** **

 RDP and CFUR and not supported together.  See CSCtg43998.

 ** **

 John

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Re: [OSL | CCIE_Voice] Locations CAC

First of all check if the IP Phones are put in correct Locations.

Then for AAR to work, you need to create an AAR Group and assing it to
endpoints.

On Sun, Feb 5, 2012 at 11:03 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:

 Hello Experts,

 I am trying to set up an AAR scenario for my studies. I configured 2
 Locations, with unlimited bandwidth, but mandatory RSVP from HQ to BR2. I
 wanna use 5 concurrent calls, and I am also using g729 between sites. I
 added the MTP-HQ, and MTP-BR2 to CUCM, put them in a MRG, followed by MRGL,
 and referenced it in its respective device pool. Reset all the phones.

 So here is my config:
 HQ
 dspfarm profile 2 mtp
  codec g729r8
  rsvp
  maximum sessions software 5
  associate application SCCP

 interface Serial0/0/1:0.2 point-to-point
  description TO BR2
  bandwidth 768
  ip address 10.10.112.1 255.255.255.0
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 202
   class AutoQoS-FR-Se0/0/1:0-202
   auto qos voip trust
  frame-relay ip rtp header-compression
  ip rsvp bandwidth 136

 BR2
 dspfarm profile 1 mtp
  codec g729r8
  rsvp
  maximum sessions software 5
  associate application SCCP

 interface Serial0/1/0:0.1 point-to-point
  description to HQ
  bandwidth 768
  ip address 10.10.112.2 255.255.255.0
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 102 CISCO
   class AutoQoS-FR-Se0/1/0:0-102
   auto qos voip
  frame-relay ip rtp header-compression
  ip rsvp bandwidth 136

 The main problem is that on the FIRST call it already says Not Enough
 Bandwidth, wasn't that supposed to happen if the 6th caller tried to make
 a call? I already set to TRUE in Service Parameters for Automated Alternate
 Routing, but it's not showing the Not Enough Bandwidh, Rerouting message.
 I haven't configured my Partitions and CSSs yet, but what's up with the
 first call not going through? Am I missing something?

 *Emanuel Damasceno*
 CCNP Voice




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Re: [OSL | CCIE_Voice] SLRG with CallForwarding

2012-02-04 Thread datucha123 datucha123

What are you trying to achieve?

Write it in more details, and we will try to help you

On Sat, Feb 4, 2012 at 7:19 AM, Seifeddine Tlili
seifeddine.tl...@lvs1.comwrote:

  Hi Everyone

 ** **

 Is there a different workground then creating PT/CSS/RP for Callfowarding
 when using SLRG with Callforwarding on the line?

 ** **

 ** **

 ** **

 *Kindly*

 * *

 *Seifeddine Tlili*  

 [image: Description: Description: Long View Systems]

 M.Eng CCIE # 26440
 Systems Consultant 

 .. *
 Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900

 [image: Description: Description: 
 Linkedin]http://www.linkedin.com/company/17908
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 Twitter]http://twitter.com/LongViewSystems
  [image: Description: Description: 
 Facebook]http://www.facebook.com/longviewsystems [image:
 Description: Description: Facebook]http://www.youtube.com/longviewsystems
 www.longviewsystems.com
 This message and any attached documents are only for the use of
 the intended recipient(s), are confidential and may contain privileged
 information. Any unauthorized review, use, retransmission, or other
 disclosure is strictly prohibited. If you have received this message in
 error, notify the sender immediately, and delete the original message. ***
 *

 ** **

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Re: [OSL | CCIE_Voice] Help: Configuring static dynamic ip address assigning in same cisco IOS dhcp server

2012-02-04 Thread datucha123 datucha123

Yes, you can.

You have to use the DHCP Binding file (put it in flash), so that if the
Client identifier is found in the file, that it will give the static IP,
otherwise it will give the IP addres out of the pool.

Or you can use the DHCP Classes.

You can read about them in DHCP admin guide.

Here is the link:
http://www.cisco.com/en/US/docs/ios-xml/ios/ipaddr_dhcp/configuration/12-4t/dhcp-12-4t-book.pdf

Take a look at - *

Configuring the DHCP Server to Read a Static Mapping Text File
*

On Sat, Feb 4, 2012 at 11:00 AM, Rrcrumm rrcr...@yahoo.com wrote:

 You need to configure one for the static and one for the rest of the
 dynamic addresses

 HTH
 Randall

 Sent from my iPhone

 On Feb 3, 2012, at 10:29 PM, Prafulla Rangari rangari.prafu...@gmail.com
 wrote:

  Hi  Everyone,
 
  Can any one tell us can we configure an IOS DHCP Pool which can provide
 static as well as  dynamic address to the clients
 
  Thanks
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Re: [OSL | CCIE_Voice] load command

2012-02-03 Thread datucha123 datucha123

you want to say that we DO NOT NEED tftp on CME for our Phones on the LAB
Exam?

On Fri, Feb 3, 2012 at 10:45 PM, Vik Malhi vma...@ipexpert.com wrote:

 Correct.

 In addition you should be careful NOT to erase factory defaults on the CME
 phone too- this would mean you need the load command in order for it to
 boot up (phone would need to TFTP a .loads file.



   On Feb 3, 2012, at 9:22 AM, Ken Wyan wrote:

   For CCME , it's required to use load command as below.

 tftp-server flash:PHONES/SCCP.loads  alias  SCCP.loads

 telephony-service
 load 7965   SCCPx

 But in CCIE lab environment , this may consume lot of time for phone
 firmware upgrades.

 I think it's better not to put any of above commands in CCIE lab  unless
 any problem arises. (In Lab all phones should be already having v7
 compatible phone loads)

 Do you agree with me or not?


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Re: [OSL | CCIE_Voice] Codec's, Compression and Bandwidth

2012-02-02 Thread datucha123 datucha123

You are wong my friend.

L3 codec bandwidth means - (Payload + L3) x PPS x 8
L2 Codec bandwidth means - (Payload + L3 + L2) x PPS x 8

So here is an example for G711 and G729 for L3 BW calculations:

G711 20ms

Payload - 160 Bytes
L3 - 40 bytes
PPS - 50

(160 + 40) x 50 x 8 = 8 bps = 80kbps

G729 20ms

Payload - 20 bytes
L3 - 40 Bytes
PPS - 50

(20 + 40) x 50 x 8 = 24000 bps = 24 kbps

Now copressed RTP

G711 20ms

Payload - 160 Bytes
L3 - 2 bytes
PPS - 50

(160 + 2) x 50 x 8 = 64800 bps = 64.8kbps

G729 20ms

Payload - 20 bytes
L3 - 2 Bytes
PPS - 50

(20 + 2) x 50 x 8 = 8800 bps = 8,8 kbps

On Thu, Feb 2, 2012 at 2:51 AM, Nathan Silvers silver...@gmail.com wrote:

Whoops started my own private convo with datucha... thought i would copy
everyone else just to get more eyes on it and hopefully bring some
clarification...

From my understanding and using the Cisco doc the default g.711 packet
size is 206bytes(including 2 - 10ms 80 byte voice samples = 160bytes or
1028bits) and using crtp is 168bytes using the calculations the pps is
64kbps/1028 bits = 50pps which using the same calculations of packet size
is 1648 * 50pps = 82.4kbps total... and the same calculation using crtp
same voice load so 50 pps bw calculation is total packet 1344bits * 50pps =
67.2kbps which is well below your layer 3 base of 80kbps and includes the
same voice payload as the non-compressed packets at 82.4... the 64 and 8
kbps are the base layer 3 bandwidth for just the audio, not including any
headers...

If anyone sees it differently please let me know, i would hate to walk
into the test and have this whole concept wrong...

On Wed, Feb 1, 2012 at 2:05 PM, datucha123 datucha123
datucha...@gmail.com wrote:

G711 64kbps and G729 8kbps are not the L3 Bandwidth for those codecs.
These are the Payload Bandwidths for those codecs.

G711 and G729 on L3 are using 80 and 24 kbps respectavely.

Here is the easy way to calcualte the Codec bandwidth:

(Payload_Size + L3_Header + L2_Header) x PPS x 8

Where
*Payload_Size* - is the Codec Sample Size (Bit Rate). G711 at 20ms has
160 bytes. G729 at 20ms has 20 Bytes and etc (you can find Sampling Rate
info on Cisco.com)
*L3 Header* - is the IP/UDP/RTP, which is 40 bytes, unless compressed
RTP is used. If the UDP checksum is used then the L3 Header is compresesd
up to 4 bytes, and if not used then up to 2 bytes. (UDP checksum is enabled
on a VoIP Dial-peer: ip udp checksum command. By default is disabled)
*L2 Header* - for Ethernet it is 18 bytes, for Frame Relay is 4 bytes
(based on QoS SRND).
*PPS* - Packets per Second. This is based on the Sampling Rate as well
in the following way: PPS = 1000 / Sample_Size in ms.
*8* - this is just bits. To convert Bytes into bits.

On Thu, Feb 2, 2012 at 1:39 AM, Nathan Silvers silver...@gmail.comwrote:

Hi Everyone,

Just starting out on my CCIE path and had a revelation that might not be
huge to anyone else but if it helps one person understand I've done a good
job...

So I was struggling with Codecs how to calculate bandwidth required and
all the header and crtp if that is involved and then Cisco throws on Layer
3 Bandwidth vs Overall bandwidth and from a humble voice guy who has
focused mainly on LAN connections I am lost...

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#topic1
Using the above Cisco document which can help and confuse(as they use
median BW when it comes to the CRTP compression as it can be a 2 or 4 byte
header depending on if you are using the UDP checksum) so my exact
calculations were always off a few bytes..

So I was thinking about this logically and looking at the different
pieces of the puzzle,
Layer 3 is going to ultimately be the voice packets which is defined by
the codec.. 711 = 64kbs, 729 = 8kbs, etc...
Layer two includes my headers and such which add to the needed bandwidth
depending on if they are compressed
So if you use CRTP you only need and additional 3.6kbs for 711 which is
the 67.6kbs bandwidth, or 11.6kbs for 729
Now if uncompressed the headers can add quite a bit more requiring the
need for 18.8kbs more to put 711 at 82.8kbs and 729 at 26.8kbs

So if you see a question regarding layer 3 bandwidth it is always the
codec kbs, ie 711 is 64 and 729 is 8 regardless of any compression, the
type of connection etc.. more of a no brainer question
They Layer 2 bandwidth is where the savings are by chopping the 40byte
IP/UDP/RTP header into 2 byes (or 4 bytes if UDP Checksum is enabled.)

Gotta love white boards and running through a few situations.. the cisco
doc has the calculations to go through and how changing the payload size
can adjust the Packets Per Second which affects the required Bandwidth. If
anyone else is struggling with this I highly recommend just writing it all
out and trying hypothetical situations.

Hope this helps someone!

--
The biggest mistake

Re: [OSL | CCIE_Voice] Calling Party Transformation set at the egress GW

2012-02-01 Thread datucha123 datucha123

That is because of IOS.

IOS detects the US Dialplan, and sets the Types accordingly in H323
gateway.
It is not possible to disable that feature. So you have to use voice
translation rules to change the ANI Type.
On Wed, Feb 1, 2012 at 2:01 PM, Juan Lopez
lopez.hernandez.j...@gmail.comwrote:

 Hi all,

 I'm doing lab 5 Vol1 with some extra things into it and found the
 following:

 Has anyone noticed that when using a H323 GW in a backup fashion (example:
 RG-BR1 contains BR1-GW as primary and HQ-GW as secondary), the calling
 number type is ALWAYS set to national, unless you use a prefix in the
 calling party transformation?

 For most cases this is ok, but not for all (example: international TEHO
 for calls from the UCME)

 any feedback is welcome !

 grts,
 Juan

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Re: [OSL | CCIE_Voice] Codec's, Compression and Bandwidth

2012-02-01 Thread datucha123 datucha123

G711 64kbps and G729 8kbps are not the L3 Bandwidth for those codecs. These
are the Payload Bandwidths for those codecs.

G711 and G729 on L3 are using 80 and 24 kbps respectavely.

Here is the easy way to calcualte the Codec bandwidth:

(Payload_Size + L3_Header + L2_Header) x PPS x 8

Where
*Payload_Size* - is the Codec Sample Size (Bit Rate). G711 at 20ms has 160
bytes. G729 at 20ms has 20 Bytes and etc (you can find Sampling Rate info
on Cisco.com)
*L3 Header* - is the IP/UDP/RTP, which is 40 bytes, unless compressed
RTP is used. If the UDP checksum is used then the L3 Header is compresesd
up to 4 bytes, and if not used then up to 2 bytes. (UDP checksum is enabled
on a VoIP Dial-peer: ip udp checksum command. By default is disabled)
*L2 Header* - for Ethernet it is 18 bytes, for Frame Relay is 4 bytes
(based on QoS SRND).
*PPS* - Packets per Second. This is based on the Sampling Rate as well in
the following way: PPS = 1000 / Sample_Size in ms.
*8* - this is just bits. To convert Bytes into bits.

On Thu, Feb 2, 2012 at 1:39 AM, Nathan Silvers silver...@gmail.com wrote:

Hi Everyone,

Just starting out on my CCIE path and had a revelation that might not be
huge to anyone else but if it helps one person understand I've done a good
job...

So I was thinking about this logically and looking at the different pieces
of the puzzle,
Layer 3 is going to ultimately be the voice packets which is defined by
the codec.. 711 = 64kbs, 729 = 8kbs, etc...
Layer two includes my headers and such which add to the needed bandwidth
depending on if they are compressed
So if you use CRTP you only need and additional 3.6kbs for 711 which is
the 67.6kbs bandwidth, or 11.6kbs for 729
Now if uncompressed the headers can add quite a bit more requiring the
need for 18.8kbs more to put 711 at 82.8kbs and 729 at 26.8kbs

Hope this helps someone!

--
The biggest mistake people make in life is not trying to make a living at
doing what they most enjoy.

- Malcolm Forbes

Nathan Silvers

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Re: [OSL | CCIE_Voice] CCIE#34310

2012-01-26 Thread datucha123 datucha123

Congratulations.

On Thu, Jan 26, 2012 at 10:05 PM, Bill Lake whl...@gmail.com wrote:

 Congratulations, take a well earned break before your work picks up and
 your busier than those of us studying.

   On Thu, Jan 26, 2012 at 10:11 AM, Thomas Koch koch1...@comcast.netwrote:

Congrats!!

 ** **

 Regards,

 ** **

 Thomas Koch

 

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chris Martin
 *Sent:* Thursday, January 26, 2012 8:04 AM

 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] CCIE#34310

  ** **

 Well its official, got my results in this morning, CCIE # 34310.  Just
 want to give a big thanks to everyone here for a great community in which
 to strive toward this goal over the last year.  Thanks Vik and IPExpert for
 the great videos, boot camp, labs rack rentals, I can not imagine myself
 achieving success without your products and ongoing support.  A big shout
 out to my boot camp  buddies from 11/11/11, keep your head down and don't
 give up, I am sure you will all make it!

 ** **

 Now time for a break.. and maybe a real book, that isn't a SRND or Cisco
 Press...

 ** **

 Chris Martin

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Re: [OSL | CCIE_Voice] MGCP gateways

2012-01-25 Thread datucha123 datucha123

If you configure ccm-manager config, it will try to configure all 24/32 PRI
ports, and if you are not using full PRI, then you have to manually disable
unused PRI channels, and also configure the B channel maintenance in CUCM,
so that after every reload the Router will not try to configure all PRI
channel again.

Bu if you are using manual MGCP configuration, you do not have such problem
in case of fractional PRI.

On Wed, Jan 25, 2012 at 5:08 PM, Chevy chevy.man...@gmail.com wrote:

 I just thought I'd ping everyone to see what your thoughts are on
 configuring an MGCP gateway manually vs using the ccm-manager config
 commands to download the xml file from the call manager.


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Re: [OSL | CCIE_Voice] New Five Labs: UCCX Question

2012-01-24 Thread datucha123 datucha123

I have tested, and Telecaster user does not need any special User Groups
(except Standard CTI Enabled).

I can tell, that even the Telecaster User is not necessary at all.

On Tue, Jan 24, 2012 at 1:28 PM, Anthony Alba ascanio.al...@gmail.comwrote:

 The script populates the variable at runtime with Set Enterprise Call
 Info step; this happens in the Select Resource step before you connect
 the caller to the agent.

 Now, much to my surprise, I actually managed to get this to work and I saw
 the field get updated.

 The question I want to ask the list: does the telecaster application
 user need any specific User Group (e.g. Standard CTI Enabled)
 ? The 6.6 and 8.x CAD are quite skimpy on this: they state to create an
 application user telecaster/password telecaster and associate all Agent
 phones. They don't mention whether the telecaster user needs specific roles.
 Searches on this list turn up which state that the telecaster user is
 needed for Expanded Call Variables to work.




   On Mon, Jan 23, 2012 at 10:12 PM, John McGaughey (jomcgaug) 
 jomcg...@cisco.com wrote:

In lab 1 of the new 5 labs, question 7.3 it asks that the agent be
 able to see the ani and number of calls in queue.  The DSG says to add a
 “salesinq” field to the default layout.  The problem is that there is
 nothing telling IPPA what to populate this field with.

 ** **

 So in my lab I see the following when I press CDATA.

 ** **

 ANI: 4678124

 callsinq:

 ** **

 But the DSG is showing callsinq: 1.  There must be a step missing from
 the DSG.  How do we populate the callsinq field?

 ** **

 John

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Re: [OSL | CCIE_Voice] No VM recorded to endpoint from PSTN

2012-01-23 Thread datucha123 datucha123

I have the same issue,

That's because of VMware  -  well I got to this conclusion.

On Mon, Jan 23, 2012 at 6:24 AM, Jurassic Labs jurassicl...@gmail.comwrote:

 I've noticed this lately while loading up the newer 5-lab self study
 vRacks sessions.  Once a phone (say Ext 2002) is defined in Unity
 Connection, if the PSTN phone calls that extension, the greeting is heard
 after the appropriate no-answer timeout and you can leave a message.
 However, NO MWI light shows up on 2002...because there is no new message
 for that mailbox!  The same applies if you call the same extension - but
 from another extension that is NOT defined in Unity Connection.  No if you
 have another extension (say 3001) call 2001 and leave a message -
 everything works...MWI, new messages, etc.  because 3001 is defined in
 Unity Connection.

 So is there some setting in Unity Connection that will only record and
 retain a message from a known subscriber?  I'm scratching my head on this
 one...


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Re: [OSL | CCIE_Voice] IPX voice vol.1 lab #8 - PRI weird behaviour

2012-01-23 Thread datucha123 datucha123

Check the Inhibit restarts at PRI initialization on MGCP gateway in
CUCM configuration page.

That might help

On Mon, Jan 23, 2012 at 6:31 AM, Jeferson Guardia jefers...@gmail.comwrote:

 Getting that weird message with my PRI every 30 seconds, I googled, found
 people having the same issue but I didnt find any solution and this was at
 the end of my lab, but anyway it raised my curiosity.. you never know you
 might get the same thing on the lab, gotta be ready to catch any curve ball
 they throw at you :-)

  BR1-RTR#
 Jan 23 02:21:58.590: ISDN Se0/0/0:23 Q931: RX - RESTART pd = 8  callref =
 0x
 Restart Indicator i = 0x87

 Looking at general status looks ok

  BR1-RTR#sh isdn stat
 Global ISDN Switchtype = primary-ni

 %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may
 not apply

 ISDN Serial0/0/0:23 interface
 dsl 0, interface ISDN Switchtype = primary-ni
 L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
 Layer 1 Status:
 ACTIVE
 Layer 2 Status:
 TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
 Layer 3 Status:
 0 Active Layer 3 Call(s)
 Active dsl 0 CCBs = 0
 The Free Channel Mask:  0x8007
 Number of L2 Discards = 0, L2 Session ID = 11
 Total Allocated ISDN CCBs = 0
 BR1-RTR#

 Then my q921 debug tells me the following:

  BR1-RTR#
 Jan 23 02:24:28.589: ISDN Se0/0/0:23 Q921: User TX - RRp sapi=0 tei=0
 nr=42
 Jan 23 02:24:28.589: ISDN Se0/0/0:23 Q921: User RX - INFO sapi=0 tei=0,
 ns=42 nr=12
 Jan 23 02:24:28.589: ISDN Se0/0/0:23 Q921: User TX - RR sapi=0 tei=0 nr=43
 Jan 23 02:24:28.593: ISDN Se0/0/0:23 Q921: User RX - RRf sapi=0 tei=0
 nr=12
 BR1-RTR#


 Yes, this was at proctorlabs.. I bet some of you might have faced the same
 issue since this is on the initial config. Somehow I just hate MGCP, yes I
 tried reconfiguring it and also the famous workaround (no mgcp/mgcp), but
 it didnt work.

 Cheers!


 --
 Jeferson Guardia
 CCIE #28157

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Re: [OSL | CCIE_Voice] New Five Labs: UCCX Question

2012-01-23 Thread datucha123 datucha123

I cannot understand, how the Agent (IPPA or CAD) can show manually
configured Varibales values, until the call comes to Agents phone.

You can use the Get Reporting Statistics Step to gather info about the
Queue  -  But I do not know, how to pass that variables to Agents layout in
real time  - the call did not get to agents yet, so how that parameter is
dynamically updated on Agents screen?

On Mon, Jan 23, 2012 at 6:12 PM, John McGaughey (jomcgaug) 
jomcg...@cisco.com wrote:

  In lab 1 of the new 5 labs, question 7.3 it asks that the agent be able
 to see the ani and number of calls in queue.  The DSG says to add a
 “salesinq” field to the default layout.  The problem is that there is
 nothing telling IPPA what to populate this field with.

 ** **

 So in my lab I see the following when I press CDATA.

 ** **

 ANI: 4678124

 callsinq:

 ** **

 But the DSG is showing callsinq: 1.  There must be a step missing from the
 DSG.  How do we populate the callsinq field?

 ** **

 John

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Re: [OSL | CCIE_Voice] RSVP CAC Sampling Rate

2012-01-22 Thread datucha123 datucha123

Nicolas, if you do not like my e-mails, just do not read them, (I do not
need your personal comments at all).

There is a great feature in your E-mail client  -  *Delete This E-mail*
or even  - *Unsubscribe from OSL
*
So you can use those buttons to get rid of my e-mails.

I will always send any questions that I am interested in and I do not need
your personal comments about them.

Thank you.
On Sun, Jan 22, 2012 at 3:23 AM, Nicolas MICHEL mcl.nico...@gmail.comwrote:

 Hey there George.

 I agree with your statements, some of the posts are very very interesting
 but I think we are overwhelmed with the number of questions.

 Again I dont want to be mean with anyone but I received some private mails
 that agreed with my opinion.
 We could then start a forum that would track any voice topics...

 My 2 cts.

 Nic



 Nicolas,

 I think it would be more constructive if you actually answered the guy.
 this is a very good question, and I believe a lot of people might have this
 wrong.

 As I already answered Datucha in private. I'll just put here the correct
 formula for calculations as per Cisco SRND:

 For N amount of calls configured for any codec speed with any sampling
 period:

 bandwidth = (N-1) * (configured codec speed on configured sampling period)
 + (codec speed on worth case scenario sampling period - 10ms)
 as per this example for g729 configured with 30ms sampling period on CUCM
 for 10 calls:
 bandwidth= 9 * 19kbps + 40kbps

 where 19 is g729 speed on 30ms sampling rate... well it's actually
 18.648bps but I rounded it up.  and 40 is same codec on worth case scenario
 10ms sampling rate.

 Hope this helps,

 On Sat, Jan 21, 2012 at 4:38 PM, Nicolas MICHEL mcl.nico...@gmail.comwrote:

  Damn man,

 Do we need to correct you for everything you test ?
 Dude make your own assumption sometimes and please don't use this list as
 your please Can you verify this and tell me if I'm wrong or not mailing
 list.

 I guess this is a mailing list that is related to CCIE and I'm sure some
 of your post are very interesting but since we receive maybe 4 or 5 mails
 from you everyday I feel the need to tell you , man please don't flood us
 like this !!

 Have a good day !  ;)

 Nic

 Le 21 janv. 2012 à 13:33, datucha123 datucha123 datucha...@gmail.com a
 écrit :

   Sorry I make a little mistake here in calculation (at 30ms Sampling
 Rate G729 needs 40kbps for RSVP CAC), but it does not change the idea of
 question,

 So in case of 30ms Sampling Rate, we do not need to add any 16 bkps and:

 10 G729 30ms Calls  -  40 x 10  = 400kbps

 On Sat, Jan 21, 2012 at 4:20 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:


 Hello,

 When we are using default sampling rate for codecs, for example G729 of
 20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
 of all calls:

 10 G729 calls  -  24 x 10 + 16 = 256 kbps

 But when using non-default Sampling Rate for codecs, for instance G729
 at 30ms, then we do not have to add 16 but add 12 kbps:

 10 G729 30ms calls  -  28 x 10 +12 = 292

 Is it correct or not?


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[OSL | CCIE_Voice] H225 Preservation

2012-01-22 Thread datucha123 datucha123

Hello,

What is the difference between these two commands:


no h225 timeout keepalive

call preserve limit-media-detection


Well, both commands (separately) can be used for SRST, when the CUCM goes
down, and active calls should be left active.

Also I have read some documentation, and the final usage for those command
are the same.

So can you tell tell me what is the difference between those two commands?
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[OSL | CCIE_Voice] RSVP CAC Sampling Rate

2012-01-21 Thread datucha123 datucha123

Hello,

When we are using default sampling rate for codecs, for example G729 of
20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
of all calls:

10 G729 calls  -  24 x 10 + 16 = 256 kbps

But when using non-default Sampling Rate for codecs, for instance G729 at
30ms, then we do not have to add 16 but add 12 kbps:

10 G729 30ms calls  -  28 x 10 +12 = 292

Is it correct or not?
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Re: [OSL | CCIE_Voice] RSVP CAC Sampling Rate

2012-01-21 Thread datucha123 datucha123

Sorry I make a little mistake here in calculation (at 30ms Sampling Rate
G729 needs 40kbps for RSVP CAC), but it does not change the idea of
question,

So in case of 30ms Sampling Rate, we do not need to add any 16 bkps and:

10 G729 30ms Calls  -  40 x 10  = 400kbps

On Sat, Jan 21, 2012 at 4:20 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:


 Hello,

 When we are using default sampling rate for codecs, for example G729 of
 20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
 of all calls:

 10 G729 calls  -  24 x 10 + 16 = 256 kbps

 But when using non-default Sampling Rate for codecs, for instance G729 at
 30ms, then we do not have to add 16 but add 12 kbps:

 10 G729 30ms calls  -  28 x 10 +12 = 292

 Is it correct or not?

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[OSL | CCIE_Voice] Type and Plan on H323

2012-01-21 Thread datucha123 datucha123

Hello,

H323 Router by default (in IOS) sets the Calling Type and Plan for outgoing
calls to PRI. - I think there is something put in IOS so that it recognizes
the US dialplan, and set the Type and Plan automatically.

How can I disable it?

Because when I set the Type and Plan in Callmanager, and send the call to
h323 gateway, if the Calling Number matches the US dialplan, then the H323
gateway overwrites the Type and Plan.
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[OSL | CCIE_Voice] Cisco IP Phone Codecs

2012-01-20 Thread datucha123 datucha123

please correct if wrong

G711 and G729 variants (g722, iLBC are not part for this email)

Cisco IP phones default codecs  -  G711Ulaw, and G729ar8.

Cisco IP phone supported codecs (SCCP)  -  G711u, G711a, G729r8, G729ar8,
G729br8 and G729abr8
Cisco IP Phone supported codecs (SIP)  -  almost the same Codecs as SCCP
version, except SIP Phone does not support Annex-B variants.
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Re: [OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM

2012-01-19 Thread datucha123 datucha123

try to reload the Router.

On Wed, Jan 18, 2012 at 3:36 PM, The Masterplan winmasterp...@gmail.comwrote:

 The same thing. The only difference is that now I don't see anymore the
 phone in show run.


 On Wed, Jan 18, 2012 at 12:34 PM, Mohd Baqari baqari.voic...@gmail.comwrote:

 Ok ... Try changing the provision mode to none ... Delete ephone-dn 1 and
 ephone 3  Then do your testing.

 Regards,
 Mohammed Al Baqari

 Sent from my iPhone

 On Jan 18, 2012, at 1:12 PM, The Masterplan winmasterp...@gmail.com
 wrote:

 The phone does not appear in the output of the command:
 tftp-server flash:gui/ephone_admin.html
  max-ephones 14
 ephone-dn  11  dual-line
  number 1002 no-reg primary
  description Cisco IP Communicator
  name Cisco IP Communicator
 ephone-dn  12  dual-line
  number 1005 no-reg primary
  description IP Blue
  name IP Blue
  night-service bell
 ephone  1
  description Cisco IP Communicator
  mac-address F0DE.F173.03E2
  type CIPC
  button  1:11
 ephone  2
  description IP Blue
  mac-address 0050.56C0.0008
  type CIPC
  button  1:12

 After I switch to srst, the output of the command looks like this:
 tftp-server flash:gui/ephone_admin.html
  max-ephones 14
 ephone-dn  1  dual-line
  number 
  description 7945 hardware
  name 7945 hardware
 ephone-dn  11  dual-line
  number 1002 no-reg primary
  description Cisco IP Communicator
  name Cisco IP Communicator
 ephone-dn  12  dual-line
  number 1005 no-reg primary
  description IP Blue
  name IP Blue
  night-service bell
 ephone  1
  description Cisco IP Communicator
  mac-address F0DE.F173.03E2
  type CIPC
  button  1:11
 ephone  2
  description IP Blue
  mac-address 0050.56C0.0008
  type CIPC
  button  1:12
 ephone  3
  mac-address 0817.3514.5682
  button  1:1


 On Wed, Jan 18, 2012 at 10:25 AM, Mohd Baqari 
 baqari.voic...@gmail.comwrote:

 Post the the output of show run | sec ephone. Probably the old config
 of ephone-dn is saved in running config due to provision all

 Regards,
 Mohammed Al Baqari

 Sent from my iPhone

 On Jan 18, 2012, at 11:55 AM, The Masterplan winmasterp...@gmail.com
 wrote:

 Hi,

 I already did no create cnf-files  create cnf-files and reset the phone
 to factory defaults and nothing. I'm running srst mode auto provision all.
 See below the config:
 telephony-service
  srst mode auto-provision all
  srst dn line-mode dual
  em logout 0:0 0:0 0:0
  max-ephones 14
  max-dn 30 no-reg
  ip source-address 10.1.1.25 port 2000
  system message SRST
  max-conferences 8 gain -6
  transfer-system full-consult
  create cnf-files version-stamp 7960 Jan 18 2012 08:09:33

 On Tue, Jan 17, 2012 at 7:13 PM, Ken Wyan kew...@gmail.com wrote:

 Did you enter no create cnf-files  create cnf-files on CME Router ?

 Which srst mode are you running? srst mode auto provision all | dn |
 none ?
 Better to post full telephony-service configuration here.

 (In SRST phone may be downloading previous xml configuration file from
 the router. Delete it from flash if so)



 On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan 
 winmasterp...@gmail.com wrote:

 Hello,

 I have a problem regarding srst. The 2811 router than now is a srst
 was a acting as a cme in past in a demo lab and the 7960 phone was
 registered to it with extension . Now, the 7960 phone is registered in
 UCM with extension 5001 and the 2811 router is configured in telephony
 service srst mode. The problem is that although the old configuration of
 the router was erased, when it goes in srst fallback mode, the 7960 gets
 extension  instead of 5001 and the command show telephony-service
 ephone shows the specified phone with message:This is an srst fallback
 phone.

 Thank you for your answer

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Re: [OSL | CCIE_Voice] CUCM version to study upgrade to latest 7.0.x release?

2012-01-19 Thread datucha123 datucha123

well, I do not know the very exact version of CUCM on the LAB Exam, but I
think it is better to study with the one that proctollabs offer. After you
will know the bugs and issues for that UCM/

On Wed, Jan 18, 2012 at 9:03 PM, Juan Lopez
lopez.hernandez.j...@gmail.comwrote:

 all,
 I found the version on proctorlabs (7.0.1.11002-2) is giving me quite some
 issues with dialrules on the 7962.
 Is it a good thing to upgrade to the latest 7.0.x release to study,
 without being out of sync with the tested UCM version?
 Or should I simply upgrade the phone firmwares instead? what is the best
 way to prepare for the real exam?

 thx,
 Juan

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Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

2012-01-19 Thread datucha123 datucha123

Also as I remember, the IPIPGW tech prefix has to match the Destination
Zone prefix.

On Wed, Jan 18, 2012 at 9:39 PM, Steven 
forum.ccie.onlinestudyl...@nocer.net wrote:

 @Boris
 @Leslie
 @Amit
 I got some other issues too.
 I skipped to check the GK-only functionality (BIG mistake).
 After i fixed the normal (without outvia) GK functions i revisited the
 CUBE issue.
 It turns out i accidently put the allow-connections on the Br2 instead of
 the HQ.

 Thanks for your time and help! :D

 Regards Steven

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Re: [OSL | CCIE_Voice] redundancy for SIP dialpeers

2012-01-18 Thread datucha123 datucha123

Also do not forget to configure timer trying  - 150/200 ms is OK

On Wed, Jan 18, 2012 at 12:46 AM, Juan Lopez lopez.hernandez.j...@gmail.com
 wrote:

 When configuring 2 SIP dialpeers for redundancy, together with:
 sip-ua
 retry invite 2

 This should generate in total 3 INVITES sent to the primary UCM via the
 first dialpeer, before going over to the second sip dialpeer, right?
 Doing debug ccsip messages only shows 1 invite sent to the primary, and
 then 1 invite to the secondary.
 am I missing something?
 thanks, Juan

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Re: [OSL | CCIE_Voice] BLF Speed Dials

Thank you George

On Mon, Jan 16, 2012 at 8:09 PM, George Goglidze gogli...@gmail.com wrote:

 Hi Datucha,

 Actually for CUPS you need only line association. you do not need to
 specify owner user id.

 Misread the question a little bit initially.

 the owner user id on the phone is for SNR.

 Cheers,

 On Mon, Jan 16, 2012 at 4:01 PM, George Goglidze gogli...@gmail.comwrote:

 correctisimo  :-)



  On Mon, Jan 16, 2012 at 3:25 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

  When configuring the simple BLF Speed Dial, we need to configure the
 Subscribe CSS for watching Device. So that it could the the Watched DN.

 But, the Owner User ID,  Line Association with End User and other kinds
 of associations are not required for this BLF, right?

 Even for Call List Presence.



 The Owner User ID and Line Association with End User, along with License
 Capability Assignements are required only for CUPS Presence method.

 Is it correct?

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Re: [OSL | CCIE_Voice] Cant connect to Sub

Also check the DB Replication Status

On Tue, Jan 17, 2012 at 6:17 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 Hello Errol ,

 This issue usually mean you have incorrect Host/ processNode files in
 your PUB or the Sub and so the communication is broken between the PUB
 and the SUB and you have to find what table we have corrupted and fix
 it , or you have a big timing difference reported by your NTP ,

 can you check what service you have running on both pub and sub
 specially the A Cisco DB and let me know what you will get .

 Ash

 On Mon, Jan 16, 2012 at 5:15 PM, Errol Abrahams eabraham2...@gmail.com
 wrote:
 
 
  Hi All,
 
  I had a problem with my VMWARE Server and I had to rebuilt the system
 from
  scratch. I have reloaded PUB,SUB,CUPS,CUC and CUCCX and all virtual
  addresses are pingable. When I activate the services for the PUB from the
  Cisco Unified Serviceability screen, it worked. But, when I try to access
  the SUB from same screen then it displays'Connection to the Server
 cannot be
  established(unable to access Remote Node).
 
  Has anybody had a problem like this and how can I fix this problem. Your
  help is appreciated..thnx.
 
  Chhers
 
  EA
 
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[OSL | CCIE_Voice] Presence on CUPC

Hello,

Based on my testings, CUPC support Presence Status change in its Contacts
only when they are imported from AD.
Manually created contacts does not support Presence Status Change.

Is it correct?
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Re: [OSL | CCIE_Voice] Unity Connection Integration

That parameter actually does not effect anything.

On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl
earl.ho...@pcmallservices.comwrote:

  I guess it would determine what your requirements are.  If, for example,
 the global requirements were that your subscriber were to be the primary
 server for all call processing, you might want to take that into account
 when choosing whether to only use one server or two and which should be the
 primary and secondary.

 ** **

 Earl Hough

 CCIE #16508 (RS/Security/Voice)

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan
 *Sent:* Tuesday, January 17, 2012 11:42 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Unity Connection Integration

 ** **

 Hi,

  

 When adding a new port group to unity connection , it prompts to enter
 only one CUCM ip address. Should we essentially provide CUCM Publisher here?
 

  

 My Integration is CUCM Sub First  CUCM Pub Second as we expect to do in
 CCIE Exam.

  

 When adding a phone system we can provide Subscriber (first)  Publisher
 (second) in order (AXL Server)

  

 When adding a new portgroup , it asks for only one ip address. The ip
 address I provide is displayed under Port Group Configuration (CUCM  TFTP)
  we can later add another CUCM address also.

  

 Any guideline?

  

 _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _

 The information contained in this transmission is confidential. It is
 intended solely for the use of the individual(s) or organization(s) to
 whom it is addressed. Any disclosure, copying or further distribution is
 not permitted unless such privilege is explicitly granted in writing by
 PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for
 the proper and complete transmission of the substance of this
 communication, nor for any delay in its receipt.



 ___
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Re: [OSL | CCIE_Voice] Unity Connection Integration

You can set it to any server you like initially, but when you go to Server
configuration in Port Group page, there you have to set them as necessary
and required.

On Wed, Jan 18, 2012 at 12:18 AM, datucha123 datucha123 
datucha...@gmail.com wrote:

 That parameter actually does not effect anything.

   On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl 
 earl.ho...@pcmallservices.com wrote:

I guess it would determine what your requirements are.  If, for
 example, the global requirements were that your subscriber were to be the
 primary server for all call processing, you might want to take that into
 account when choosing whether to only use one server or two and which
 should be the primary and secondary.

 ** **

 Earl Hough

 CCIE #16508 (RS/Security/Voice)

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan
 *Sent:* Tuesday, January 17, 2012 11:42 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Unity Connection Integration

 ** **

 Hi,

  

 When adding a new port group to unity connection , it prompts to enter
 only one CUCM ip address. Should we essentially provide CUCM Publisher here?
 

  

 My Integration is CUCM Sub First  CUCM Pub Second as we expect to do in
 CCIE Exam.

  

 When adding a phone system we can provide Subscriber (first)  Publisher
 (second) in order (AXL Server)

  

 When adding a new portgroup , it asks for only one ip address. The ip
 address I provide is displayed under Port Group Configuration (CUCM  TFTP)
  we can later add another CUCM address also.

  

 Any guideline?

  

 _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _

 The information contained in this transmission is confidential. It is
 intended solely for the use of the individual(s) or organization(s) to
 whom it is addressed. Any disclosure, copying or further distribution is
 not permitted unless such privilege is explicitly granted in writing by
 PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for
 the proper and complete transmission of the substance of this
 communication, nor for any delay in its receipt.



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

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



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Re: [OSL | CCIE_Voice] CUE error

First of all ensure that the Authentication URL is set correctly.

Also you can try to reset the CUE module - just in case.

On Tue, Jan 17, 2012 at 6:30 PM, study buddy studybudd...@gmail.com wrote:

 Hi

 While accessing my voicemail via VoiceView, I get the following error when
 I click on listen

 Unknown error code {0}. Report this error to your system administrator

 Has anyone run into this error before? I tried several things but I cant
 get over this error

 TR

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Re: [OSL | CCIE_Voice] MGCP Registration

2012-01-16 Thread datucha123 datucha123

What does the show isdn status shows up?

On Mon, Jan 16, 2012 at 5:35 PM, mercy forall mercy_for_...@hotmail.comwrote:

  Hi,

 thanks for your support and good link\

 this is my GW configuration , also it is connected to other cisco GW as
 PSTN GW through E1 cross cable

  sh run :
 Current configuration : 15381 bytes
 !

 !
 version 15.0
 service timestamps debug datetime msec
 service timestamps log datetime msec
 no service password-encryption
 !
 hostname 
 !
 boot-start-marker
 boot-end-marker
 !
 logging buffered 51200 warnings

 no aaa new-model

 network-clock-participate wic 2
 !
 dot11 syslog
 ip source-route
 !
 ip cef
 !
 !
 no ipv6 cef
 multilink bundle-name authenticated
 !
 !
 !
 isdn switch-type primary-qsig
 !
 voice-card 0
 !
 !
 voice rtp send-recv
 !
 voice service voip
  allow-connections h323 to h323
  allow-connections h323 to sip
  allow-connections sip to h323
  allow-connections sip to sip
  redirect ip2ip
  h323
  sip
   header-passing
   no call service stop
 !
 voice class codec 1
  codec preference 1 g711ulaw
 !
 voice class custom-cptone
  dualtone disconnect
   frequency 425
   cadence 250 250
 !
 !
 !
 !
 http client cache memory pool 15000
 http client cache memory file 500
 http client connection timeout 60
 http client connection idle timeout 10
 http client response timeout 30

 mrcp client timeout connect 10
 mrcp client timeout message 10
 mrcp client rtpsetup enable
 vxml tree memory 500
 vxml audioerror
 vxml version 2.0
 !
 crypto pki trustpoint TP-self-signed-3307538538
  enrollment selfsigned
  subject-name cn=IOS-Self-Signed-Certificate-3307538538
  revocation-check none
  rsakeypair TP-self-signed-3307538538

 !
 controller E1 0/2/0
  pri-group timeslots 1-4,16 service mgcp
 !

 interface GigabitEthernet0/0
  no ip address
  duplex auto
  speed auto
  media-type rj45
 !


 interface GigabitEthernet0/0.1
  encapsulation dot1Q
  ip address X.X.X.X 255.255.255.0
 !

 !
 interface Serial0/2/0:15
  no ip address
  encapsulation hdlc
  isdn switch-type primary-qsig
  isdn incoming-voice voice
  isdn bind-l3 ccm-manager
  no cdp enable
 !
 ip forward-protocol nd
 !
 !
 ip http server
 ip http access-class 23
 ip http authentication local
 ip http secure-server
 ip http timeout-policy idle 60 life 86400 requests 1
 ip route 0.0.0.0 0.0.0.0 X.X.X.X
 !

 !
 !
 control-plane
 !
 call threshold global cpu-5sec low 70 high 85
 !
 voice-port 0/2/0:15
 !
 voice-port 0/3/0
 !
 voice-port 0/3/1
 !
 ccm-manager switchback immediate
 ccm-manager fallback-mgcp
 ccm-manager redundant-host X.X.x.x
 ccm-manager mgcp
 ccm-manager config server x.x.x.x
 !
 mgcp
 mgcp call-agent x.x.x.x service-type mgcp version 1.0
 mgcp bind control source-interface GigabitEthernet0/0.1
 mgcp bind media source-interface GigabitEthernet0/0.1
 !
 mgcp profile default
 !
 !
 gateway
  timer receive-rtp 1200
 !
 sip-ua
 retry invite 1
  retry bye 1
  retry cancel 1
  timers expires 6
  reason-header override
 !
 !
 telephony-service
  max-conferences 12 gain -6
  transfer-system full-consult
 !


 thanks

  --
 From: gogli...@gmail.com
 Date: Mon, 16 Jan 2012 13:29:49 +0100

 Subject: Re: [OSL | CCIE_Voice] MGCP Registration
 To: mercy_for_...@hotmail.com
 CC: ccie_voice@onlinestudylist.com


 Hi,

 The card is supported:
 http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf

 Paste your voice gateway configuration, there must be a problem somewhere.

 Cheers,

 On Sun, Jan 15, 2012 at 8:45 PM, mercy forall 
 mercy_for_...@hotmail.comwrote:


 Hi,

 I think the issue from card , i need to sure the CUCM support
 VWIC-1MFT-G703  card or not , because i did not fine it in cucm
 mgcp configuration and i use VWIC-1MFT-E1

 i made all configuration , and made same configuration on other and it is
 work fine , from where i can check if is it support tor not ?

   From: ccie_voice-requ...@onlinestudylist.com
  Subject: CCIE_Voice Digest, Vol 71, Issue 54
  To: ccie_voice@onlinestudylist.com
  Date: Sun, 15 Jan 2012 06:01:19 -0500
 
  Send CCIE_Voice mailing list submissions to
  ccie_voice@onlinestudylist.com
 
  To subscribe or unsubscribe via the World Wide Web, visit
  http://onlinestudylist.com/mailman/listinfo/ccie_voice
  or, via email, send a message with subject or body 'help' to
  ccie_voice-requ...@onlinestudylist.com
 
  You can reach the person managing the list at
  ccie_voice-ow...@onlinestudylist.com
 
  When replying, please edit your Subject line so it is more specific
  than Re: Contents of CCIE_Voice digest...
 
 
  Today's Topics:
 
  1. Re: AAR for UCCX (Mohammed Al Baqari)
  2. BAT csv template for CUC 7 (donny f)
  3. MGCP Registration (mercy forall)
  4. Re: AAR for UCCX (datucha123 datucha123)
  5. Re: MGCP Registration (datucha123 datucha123)
 
 
  --
 
  Message: 1
  Date: Sun, 15 Jan 2012 02:24:54 +0400
  From: Mohammed Al Baqari

[OSL | CCIE_Voice] BLF Speed Dials

2012-01-16 Thread datucha123 datucha123

When configuring the simple BLF Speed Dial, we need to configure the
Subscribe CSS for watching Device. So that it could the the Watched DN.

But, the Owner User ID,  Line Association with End User and other kinds of
associations are not required for this BLF, right?

Even for Call List Presence.



The Owner User ID and Line Association with End User, along with License
Capability Assignements are required only for CUPS Presence method.

Is it correct?
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Re: [OSL | CCIE_Voice] AAR for UCCX

2012-01-15 Thread datucha123 datucha123

I have not testeed AAR with EM, but for CTI RP I have made it and it works.

Maybe I made the wrong test with CTI RP, but the fact is that I could call
the UCCX IVR from Branch 1 IP phones when there was no enough bandwidth on
the WAN link, and the call to UCCX was routed through PSTN.

On Sun, Jan 15, 2012 at 2:24 AM, Mohammed Al Baqari 
baqari.voic...@gmail.com wrote:

  Hi,

 ** **

 I haven’t tested this. But my info is based on CUCM SRND. They haven’t
 given any exception on AAR and CTI inter-operability. Can you please
 highlight further.

 ** **

 “AAR does not support CTI route points as the origin or the destination
 of calls. Also, AAR is incompatible with the Extension Mobility feature
 when users roam across different sites.”

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
 *Sent:* Sunday, January 15, 2012 1:03 AM
 *To:* Mohd Baqari
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] AAR for UCCX

 ** **

 AAR is supported with CTI, when it is registered. For instance the CTI
 Route Point supports AAR, when it is registered. But as soon as it will
 unregister (or some Dummy CTI Route Point, which never gets registered)
 does not support AAR.

  


  

 On Sat, Jan 14, 2012 at 10:29 PM, Mohd Baqari baqari.voic...@gmail.com
 wrote:

 Hi Datucha,

 In your scenario whats confusing me is how aar is triggered. As far as I
 know AAR isn't supported with CTI

 Regards,
 Mohammed Al Baqari

 Sent from my iPhone


 On Jan 8, 2012, at 9:11 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Hello,
 
  I have been making a test and find out the following issue:
 
  So, I set the AAR Mask to CTI Route Points Line Settings, and the calls
 from Branch1 to UCCX when there was no WAN bandwidth available were routed
 through PSTN successfully.
 
  I have not set any AAR Group and AAR CSS for CTI ports, but the HQ
 gateway has them.
 
  So when I called the UCCX IVR (simpla AA script) through HQ Gateway and
 dialed the Branch 1 Phones extensions, the call has been routed through
 PSTN based on CAC restrictions. Well that's ok with CAC. But I cannot
 understand how the AAR got worked, when neither CTI ports nor the CTI Route
 Point had any AAR configuration (AAR Mask has been also removed from the
 CTI RP Line).
  HQ Gateway had the AAR Group and CSS set. So based on that, UCCX took
 the HQ gateway setttings of AAR and made a call?
 
 
 
 
 

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 please visit www.ipexpert.com
 
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 ** **

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Re: [OSL | CCIE_Voice] MGCP Registration

2012-01-15 Thread datucha123 datucha123

first of all make sure that the L2 is up for the PRI, otherwise the MGCP
gateway in CUCM will show unregistere status.

On Sun, Jan 15, 2012 at 12:16 PM, mercy forall mercy_for_...@hotmail.comwrote:

  Hi,


 i have 3845 with VWIC-1MFT-G703

 i configured mgcp but it is bending in registering on sh ccm-manager

 i did not find VWIC-1MFT-G703 on CUCM 7 , i use VWIC-1MFT-E1 ,

 Are this card supported in mgcp ?

 thanks

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Re: [OSL | CCIE_Voice] Privacy Calling Number

2012-01-15 Thread datucha123 datucha123

Calling Number will be visible to all other Sharing users, until you answer
the call. As soon as you will pickup the call, the other IP phone users wil
not be able to see active call on you phone any more.

On Sun, Jan 15, 2012 at 5:49 PM, Ken Wyan kew...@gmail.com wrote:

 According to cisco docs , when Privacy Button is pressed , calling number
  time should not be visible to other phones  sharing my number.

 My phone is registered to CUCM 7.

 I have a line shared with another phone.

 If I answer a call ringing on shared line , others can barge into my call.

 But if I press Privacy button after answering call , others cannot barge
 into my call. But still calling number is visible to other phone sharing
 this number.

 What can be the reason?

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Re: [OSL | CCIE_Voice] lab 4a shouldn't take too much time, right?

2012-01-14 Thread datucha123 datucha123

First of all you can look at the CUCM Traces filtered by particular Phone.

Go to Trace, then enable Device Based Tracing and choose the necessary
Phone/device

After ssh into CUCM Server and issue the command file tail activelog
/cm/trace/ccm/sdi recent,  this will give you the real time traces.



On Sat, Jan 14, 2012 at 5:41 PM, Juan Lopez
lopez.hernandez.j...@gmail.comwrote:

 someone of you guys/girls walked into following weird behaviour ? I for
 sure wasted too much time with lab 4a, 2 remote rack days :(( , where a SIP
 HQ phone (5002) is supposed to dial 3...@ipxcme.com (SIP CME phone)

 The issue was when picking up the call ath the CME phone, it disconnected.
 So a codec issue. Per lab 4a, the call at HQ phone leaves the trunk, both
 are in HQ device pool.

 Inbound dialpeer at CME has voice-class codec with g711 and g729.

 Registered SIP CME phone has g711 configured on voice register pool.

 So call should negotiate to g711: supported by the region configuration at
 UCM, on the inbound call leg and the final CME phone.

 But it fails.

 Putting g729 on voice register pool  call succeeds as g729...  So in
 g729 it works  Suspicion: is the correct inbound dialpeer matched? YES,
 the one with the voice-class codec. So that cannot be the cause why the
 call MUST be g729.

 What is? debug ccsip mess is not very descriptive: the CME router sends
 BYE to the UCM, with *some* cause code, when the CME phone is configured to
 have g711. Not familiar zith UCM traces unfortunately (hopefully this will
 improve, quickly...)

 After a *very long* journey in the dark, I reset the UCM SIP phone.
 Problem automagically gone :-(

 Question:

 - does this happen often with sip phones?

 - what trace should I have looked into on the UCM, to find more UCM info
 that the phone ONLY tried to negotiate the g729? Can someone point me in a
 good direction what trace to enable, and what text files to investigate?
 Is there some good info/examples available on this topic :S ?

 thanks
 /juan

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Re: [OSL | CCIE_Voice] incoming called-number .

For incoming calls, the 23548 is more specific match for Called Number of
235482345 then 235. And that is why the Dial-peer 2 is matched.

For outgoing calls, if you place a call to the same number (235482345) and
the destination patterns are the same (235 and 23548) then the dial-peer 2
would be matched as again, it is more specific match and if the Enblock is
used. If you will use Overlap Dialing, (UCME for instance) then dial-peer 1
would be matched

On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan kew...@gmail.com wrote:

 Is there a cisco doc for incoming called-number selection order?

 What I mean is say

 dial-peer voice 1
 incoming called-number  235
 xxx


 dial-peer voice 2
 incoming called-number 23548
 xx

 If a call arrives with DNIS  235482345 which incoming dial-peer will be
 matched?

 As per my testing it seems dial-peer 2 is selected.

 If this was an outgoing dial-peer (with default dial-peer hunt 0) ,
 dial-peer 1 should be the match  this is well documented.

 For incoming calls ??

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[OSL | CCIE_Voice] CUE MWI types

Hello,

Please correct me if I am wrong.

CUE has three types for MWI notification:

1) Outcall
2) Sub-notify (Subscribe)
3) Unsolicited

CME does support all three MWI methods:

When using Subscribe option in CME, we have to configure the mwi sip
commands on SCCP Ephone-DNs, and MWI on SIP Voice Register DNs.

When using Unsolicited option in CME, we do not have to configure the mwi
sip command for SCCP Phones any more, while the SIP Voice Register DN
still needs the MWI command.

So, the CME SCCP Phones need the MWI SIP command only when using
Subscriber option (NOT FOR UNSOLICITED). And the SIP Phones needs the MWI
command with both types - Subscribe and Unsolicited.



Now as for SRST:

SIP phones in SRST does not support MWI at all -  There is a bug in IOS
version for the exam.

SCCP phones in either CME-SRST or Traditional SRST DO NOT needs mwi sip
command, even for Subscribe (Call-manager-fallback does not support
ephone-dn command).

We have to configure the SIP-UA with mwi-server command (also in case of
CME) and it should work

please correct me if I am wrong.

Thank you very much.
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Re: [OSL | CCIE_Voice] incoming called-number .

First of all Preference command works only for Outgoing calls. It does not
make any sense for incoming dial-peer matching.

also in that particular case, the preference command will not make any
sense, because those dial-peers are having a differenct destination
patterns.

On Fri, Jan 13, 2012 at 4:31 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:

 Wouldn't the command preference X work in that situation?

 *Emanuel Damasceno*
 CCNP Voice






 On Fri, Jan 13, 2012 at 6:36 AM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 For incoming calls, the 23548 is more specific match for Called Number of
 235482345 then 235. And that is why the Dial-peer 2 is matched.

 For outgoing calls, if you place a call to the same number (235482345)
 and the destination patterns are the same (235 and 23548) then the
 dial-peer 2 would be matched as again, it is more specific match and if the
 Enblock is used. If you will use Overlap Dialing, (UCME for instance) then
 dial-peer 1 would be matched

  On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan kew...@gmail.com wrote:

  Is there a cisco doc for incoming called-number selection order?

 What I mean is say

 dial-peer voice 1
 incoming called-number  235
 xxx


 dial-peer voice 2
 incoming called-number 23548
 xx

 If a call arrives with DNIS  235482345 which incoming dial-peer will be
 matched?

 As per my testing it seems dial-peer 2 is selected.

 If this was an outgoing dial-peer (with default dial-peer hunt 0) ,
 dial-peer 1 should be the match  this is well documented.

 For incoming calls ??

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Re: [OSL | CCIE_Voice] Preference within policy-map

That is a great question.

Based on my knowledge, the Set DSCP command is executed first, because
otherwise the exceeding traffic will become EF.

You can also refer to this Linke, where the Auto QoS is duscussed:

http://ieoc.com/forums/t/17680.aspx?PageIndex=1

On Fri, Jan 13, 2012 at 5:47 PM, Ken Wyan kew...@gmail.com wrote:

 When I want to mark all voip packets to ef  mark exceeding packets to 8 I
 do following in a Catalyst Switch.

 policy-map AutoQos-Policy-Untrust
  class AutoQos-VoIP-RTP-Untrust
police 128 exceed-action policed-dscp-transmit
set dscp ef

 mls qos map policed-dscp  46 to 8


 Here will the set dscp ef command will be executed before police command?
 Otherwise exceeding packets (which are remarked to 8 by police statement )
 can again marked with dscp ef.



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Re: [OSL | CCIE_Voice] incoming called-number .

Incoming called number matching is done based on more specific pattern
(Incoming called-number ).

So it depends what Number is received as the Called Number.

2012/1/13 Farkas Péter wormh...@sch.bme.hu

 Good to know:

 When the Cisco IOS router or gateway receives a call setup request, a
 dial peer match is made for the incoming call in order to facilitate
 routing the call to different session applications. This is not a
 digit-by-digit match, rather the full digit string received in the setup
 request is used to match against configured dial peers.

 So I vote dp 2.

 Peter

 - Original Message -
 From: Ken Wyan kew...@gmail.com
 Date: Friday, January 13, 2012 3:10 pm
 Subject: Re: [OSL | CCIE_Voice] incoming called-number .
 To: Emanuel Damasceno aedamasc...@gmail.com
 Cc: ccie_voice@onlinestudylist.com


  If we had 2 incoming dial-peers with same incoming called-number , I
 think
   selection should be random.
 
   Anybody found a cisco document for this?
 
   On Fri, Jan 13, 2012 at 7:05 PM, Emanuel Damasceno 
 aedamasc...@gmail.comwrote:
 
Right. =)
   
*Emanuel Damasceno*
CCNP Voice
   
   
   
   
   
   
On Fri, Jan 13, 2012 at 10:41 AM, datucha123 datucha123 
datucha...@gmail.com wrote:
   
First of all Preference command works only for Outgoing calls. It
 does
not make any sense for incoming dial-peer matching.
   
also in that particular case, the preference command will not make
 any
sense, because those dial-peers are having a differenct destination
patterns.
   
 On Fri, Jan 13, 2012 at 4:31 PM, Emanuel Damasceno 
aedamasc...@gmail.com wrote:
   
Wouldn't the command preference X work in that situation?
   
*Emanuel Damasceno*
 CCNP Voice
   
   
   
   
   
   
On Fri, Jan 13, 2012 at 6:36 AM, datucha123 datucha123 
datucha...@gmail.com wrote:
   
For incoming calls, the 23548 is more specific match for Called
 Number
of 235482345 then 235. And that is why the Dial-peer 2 is matched.
   
For outgoing calls, if you place a call to the same number
 (235482345)
and the destination patterns are the same (235 and 23548) then the
dial-peer 2 would be matched as again, it is more specific match
 and if the
Enblock is used. If you will use Overlap Dialing, (UCME for
 instance) then
dial-peer 1 would be matched
   
 On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan kew...@gmail.com
 wrote:
   
 Is there a cisco doc for incoming called-number selection order?
   
What I mean is say
   
dial-peer voice 1
incoming called-number  235
xxx
   
   
dial-peer voice 2
incoming called-number 23548
xx
   
If a call arrives with DNIS  235482345 which incoming dial-peer
 will
be matched?
   
As per my testing it seems dial-peer 2 is selected.
   
If this was an outgoing dial-peer (with default dial-peer hunt 0)
 ,
dial-peer 1 should be the match  this is well documented.
   
For incoming calls ??
   
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[OSL | CCIE_Voice] CUE with CUCM

Hello,

Do we need to configure the CUCME-SRST (or call-manager-fallback) as the
Third Callmanager in JTAPI Subsystem configuration in CUE for SRST fallback?

I think we do not need to configure the CUCME-SRST as the Third
Callmanager, as CUCME-SRST does not use the JTAPI, but is integrated with
SIP, and we have to configure the SIP Subsystem for that. Right?
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Re: [OSL | CCIE_Voice] Local Route Groups

Different Route Lists may use the LRG

On Thu, Jan 12, 2012 at 4:26 AM, Spence, Paul paul.spe...@jacobs.comwrote:

 Guys,
 Need a little advise on LRG, I have it set up in a live environment and
 associated with a CUCM group. We recently added a new subscriber to a
 remote site and wanted to include it the LRG set up, but that meant adding
 the subscriber to the CUCM group at the cluster site and as such the
 devices would not register with the local server.  Took the remote site out
 of the LRG set up and built its own set of route patterns etc.
 My question is and not had the time to test in a lab can I use the
 Standard local Route Group in more than one RL and therefore have another
 set of common dial patterns.
 E.g RL1 uses standard local route group and CUCM group1 and RL2 uses
 standard local route group and CUCM group 2

 Thanks
 Paul


 Sent from mobile device, grammar mistakes are purely accidental.

 NOTICE - This communication may contain confidential and privileged
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 viewing, copying or distribution of, or reliance on this message by
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 Registered in England and Wales under number 2594504
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Re: [OSL | CCIE_Voice] Local Route Groups

No it is not bound to one Route List, it can be used by any Route List in
the system



On Thu, Jan 12, 2012 at 4:21 PM, Spence, Paul paul.spe...@jacobs.comwrote:

  Ok that’s good to know I though it may have been bound to one RL only

 Many Thanks

 ** **

 *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
 *Sent:* 12 January 2012 12:10
 *To:* Spence, Paul
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] Local Route Groups

 ** **

 Different Route Lists may use the LRG

 On Thu, Jan 12, 2012 at 4:26 AM, Spence, Paul paul.spe...@jacobs.com
 wrote:

 Guys,
 Need a little advise on LRG, I have it set up in a live environment and
 associated with a CUCM group. We recently added a new subscriber to a
 remote site and wanted to include it the LRG set up, but that meant adding
 the subscriber to the CUCM group at the cluster site and as such the
 devices would not register with the local server.  Took the remote site out
 of the LRG set up and built its own set of route patterns etc.
 My question is and not had the time to test in a lab can I use the
 Standard local Route Group in more than one RL and therefore have another
 set of common dial patterns.
 E.g RL1 uses standard local route group and CUCM group1 and RL2 uses
 standard local route group and CUCM group 2

 Thanks
 Paul


 Sent from mobile device, grammar mistakes are purely accidental.

 NOTICE - This communication may contain confidential and privileged
 information that is for the sole use of the intended recipient. Any
 viewing, copying or distribution of, or reliance on this message by
 unintended recipients is strictly prohibited. If you have received this
 message in error, please notify us immediately by replying to the message
 and deleting it from your computer.

 Jacobs Engineering U.K. Limited
 1180 Eskdale Road, Winnersh, Wokingham RG41 5TU
 Registered in England and Wales under number 2594504
 ___
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 visit www.ipexpert.com

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 ** **

 --
 NOTICE - This communication may contain confidential and privileged
 information that is for the sole use of the intended recipient. Any
 viewing, copying or distribution of, or reliance on this message by
 unintended recipients is strictly prohibited. If you have received this
 message in error, please notify us immediately by replying to the message
 and deleting it from your computer.

 --
 Jacobs Engineering U.K. Limited
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Re: [OSL | CCIE_Voice] RSVP and Multicast MoH

I have tested today, and Boris is right. The multicast MoH can travers
through RSVP MTPs fine.

While Mutlicast does not work for H323 and SIP gateway, when they are using
MTP

On Thu, Jan 12, 2012 at 12:14 AM, datucha123 datucha123 
datucha...@gmail.com wrote:

 I will test that tomorrow. Because the MMoH does not work for Gateways
 (H323 and SIP) when MTP is checked for those Gateways/Trunks (Multicast
 cannot traverse the MTP device, it cannot be teared down and re-originated)


 On Wed, Jan 11, 2012 at 11:43 PM, Boris boris.k...@gmail.com wrote:

 AFAIK MMoH will work together with RSVP because it is not taken into
 consideration for CAC. The multicast streams flow around MTP/RSVP
 reservations, but you need to cater for this traffic in priority queue
 towards the stream destination.

 Sent from my mobile device, sorry for typos.
 ---
 Regards
 Boris

 On 12/01/2012, at 5:31, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Hello,
 
  When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH
 should not work for Branch 1 IP Phones from UCM Server, as the Muticast
 cannot traverse the MTP, right?
 
  But when sourcing Mutlicast MoH from the Branch Router, it will work.
 
 
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[OSL | CCIE_Voice] Phone Label

Hello,

I have noticed, that the Phone Line Label is not passed with SCCP Phone,
when it goes to SRST.

But when the SIP Phones goes to SRST, then the Line Label is passed on over
the SRST.

Is it normal?
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[OSL | CCIE_Voice] SRST Auto Provision

Hello,

Can you please confirm the following?

When Using Auto Provision DN (for CME SRST), then we can manually create
the DNs, so that the CME will try to match the manually created DN with
learned DN, and if it matches, then assing to that Ephone.
When using Auto Provision ALL, then we can manually create DN's and Ephone
also, and in that case, the CME will try to match the manually created
Ephones and Ephone-dn with learned ones.
And when using NONE option, then we cannot create DNs or Ephone manually,
and the CME will not make any matching, and will create learned DNs and
Ephones.
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[OSL | CCIE_Voice] RSVP and Multicast MoH

Hello,

When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH should
not work for Branch 1 IP Phones from UCM Server, as the Muticast cannot
traverse the MTP, right?

But when sourcing Mutlicast MoH from the Branch Router, it will work.
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Re: [OSL | CCIE_Voice] RSVP and Multicast MoH

I will test that tomorrow. Because the MMoH does not work for Gateways
(H323 and SIP) when MTP is checked for those Gateways/Trunks (Multicast
cannot traverse the MTP device, it cannot be teared down and re-originated)

On Wed, Jan 11, 2012 at 11:43 PM, Boris boris.k...@gmail.com wrote:

 AFAIK MMoH will work together with RSVP because it is not taken into
 consideration for CAC. The multicast streams flow around MTP/RSVP
 reservations, but you need to cater for this traffic in priority queue
 towards the stream destination.

 Sent from my mobile device, sorry for typos.
 ---
 Regards
 Boris

 On 12/01/2012, at 5:31, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Hello,
 
  When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH
 should not work for Branch 1 IP Phones from UCM Server, as the Muticast
 cannot traverse the MTP, right?
 
  But when sourcing Mutlicast MoH from the Branch Router, it will work.
 
 
  ___
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 please visit www.ipexpert.com
 
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[OSL | CCIE_Voice] UCM Locations vs RSVP

Hello,

I have a question about the CAC:

Imagine that we are asked to configure the UCM standard Locations CAC
between HQ, BR1 and BR2 Sites in the following way (this is just an example)

BR1  to BR2  -  allow up to 10 calls

BR2  to HQ  -  allow up to 5 calls.

Well if using UCM Standard Locations that quite easy, and it will work
fine.

But when the requirements is to use the RSVP CAC to acomplish that, I think
there is a little problem:

Well, first of all the HQ Route is terminating both Sites PVC's (maybe I am
wrong, and traffic from BR2 to BR1 does not go through HQ Router, but
through FR Switch directly). So based on that I think the configuration has
to be the following:

On BR2 Serial Interface  -  ip rsvp bandwidth equivalent to 10 calls
(actual value does not mean in this case)
On BR1 Serail Interface  - ip rsvp bandwidth equivalent to 10 calls
On HQ Router to BR1 Serial interface -  ip rsvp bandwidth equivalent to 10
calls
On HQ Router to BR2 Serial Interface  -  and here comes the problem: if I
configure 10 calls equivalent value for ip rsvp bandwidth then up to 10
calls will be permitted from BR2 to HQ (whereas only 5 has to be
permitted), and if I configure 5 calls equivalent value, then the BR2 to
BR1 will also have only 5 calls.

Can you please help my to get the idea for such situation?
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[OSL | CCIE_Voice] Use External Phone Number Mask