[OSL | CCIE_Voice] WB 2 lab 6 -- Qos

2011-04-09 Thread Erwan Erwan
hi all,
 
When should we use  NBAR or DSCP  for class-map ?
 
Like :  class-map match-all RTP
   match dscp ef
 
Other :  class-map match-any RTP
   match protocol rtp audia
   match protocol rtcp
 
 
tks
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Re: [OSL | CCIE_Voice] DSCP QoS

2011-04-09 Thread Miron Kobelski
Hi,

AF31 is DSCP 26. DSCP 24 is CS3.

HTH
kobel

On Sat, Apr 9, 2011 at 06:01, lee huntl huntc...@gmail.com wrote:

 Dear all,
 With DSCP, are both of the following config for Voice QoS identical?? And
 is one preferred over the other one??


 class-map match-all VOICE
match dscp ef
 class-map match-all CALL-SIGNALING
match dscp af31


 VS


  class-map match-all VOICE
 match ip dscp 46
 class-map match-all CALL-SIGNALING
 match ip dscp 24


 Thanks,
 Hunt

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Re: [OSL | CCIE_Voice] DSCP QoS

2011-04-09 Thread Brian
hi - af31= dscp 26
  cs3 = dscp 24

hth 
Brian


Sent from my iPad

On 9 Apr 2011, at 05:01, lee huntl huntc...@gmail.com wrote:

 Dear all,
 
 With DSCP, are both of the following config for Voice QoS identical?? And is 
 one preferred over the other one??
  
  
 class-map match-all VOICE
match dscp ef
 class-map match-all CALL-SIGNALING
match dscp af31
  
  
 VS
  
  
 class-map match-all VOICE
 match ip dscp 46
 class-map match-all CALL-SIGNALING
 match ip dscp 24
  
  
 Thanks,
 Hunt
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[OSL | CCIE_Voice] System prompts in UCCX

2011-04-09 Thread Miron Kobelski
Hello,

I was wondering what are the best practices regarding default UCCX system
prompts usage.
UCCX installation comes with a set of default prompts located in C:\Program
Files\wfavvid\Prompts\system
There are prompts like Goodbye, Welcome, Thank you, Try again, etc.
They might be useful on the actual exam.
In script editor I can select those prompts for various steps by browsing
local disk directly and referencing them with complete path.
But is this the best way to use them? The other way is to upload them via
web interface to prompt repository.
I got confused today, because the first method didn't work for me, while the
second did the trick.


regards
kobel
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[OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)

2011-04-09 Thread Alex Goh
Hi Guys,

I'm trying to get the solutions for question 4.2 to work, but
apparently I'm missing something and hope someone can help.
I've search thru the list but doesn't really found a solution work for my case.

The issue I've encounter are when HQ phone 5001 calling BR2 phone
3003, 3003 ring, but when i tried to answered, the call drop.
I know it might be related to codec issue, but I've my HQ-RTR
configured with Xcoder which it is up and active but the call
still failing. I also did have the trunk in cucm Wait for Far End
H.245 Terminal Capability Set unchecked.

once things I notice is that, my call doesn't seems get re-originated
on the cube router to BR2 router, what I see during ringing state
my show gatekeeper endpoint show the call is directly from the CUCM
to BR2 It is only 2 call legs instead of 4 (see below).

hm, what have I missed?

Some Info:
HQ Router (R1)

interface Loopback0
 ip address 172.1.254.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id VIA ipaddr 172.1.254.1 1719
 h323-gateway voip h323-id R1
 h323-gateway voip bind srcaddr 172.1.254.1

gatekeeper
 zone local UCM 172.1.254.1
 zone local UCME outvia VIA
 zone local VIA
 zone prefix UCME 3...
 gw-type-prefix 1#* default-technology
 no shutdown

dial-peer voice 30 voip
 destination-pattern 3...
 session target ras
 codec g711ulaw
!
dial-peer voice 31 voip
 incoming called-number 3...

Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
511-32797  6   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: gk_trunk_25001
   CallSignalAddr  Port  RASSignalAddr   Port
   172.1.10.20 38233 172.1.10.20 32795
 Endpt(s): Alias E.164Addr
   dst EP: R33003
   CallSignalAddr  Port  RASSignalAddr   Port
   172.3.254.1 1720  172.3.254.1 49395

GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
172.1.10.10 47142 172.1.10.10 32838 UCM   VOIP-GW
H323-ID: gk_trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
172.1.10.20 38233 172.1.10.20 32795 UCM   VOIP-GW
H323-ID: gk_trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
172.1.254.1 1720  172.1.254.2 56974 VIA   H323-GW
H323-ID: R1
Voice Capacity Max.=  Avail.=  Current.= 0
172.3.254.1 1720  172.3.254.1 49395 VIA   H323-GW
H323-ID: R3
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 4

R1(config-if)#do sh gatek gw
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*(Default gateway-technology)
  Zone UCM master gateway list:
172.1.10.20:38233 gk_trunk_2
172.1.10.10:47142 gk_trunk_1
  Zone VIA master gateway list:
172.3.254.1:1720 R3
172.1.254.2:1720 R1

BR2 Router (R2)

interface Loopback0
 ip address 172.3.254.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id VIA ipaddr 172.1.254.1 1719
 h323-gateway voip h323-id R3
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 172.3.254.1

dial-peer voice 10 voip
 incoming called-number 3...
 dtmf-relay rtp-nte
 codec g711ulaw
!

CUCM Trunk
the trunk was assign a separate DP with a region that using G729 when
calling HQ and BR2.



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

2011-04-09 Thread ShinGei Yong
Hi Alex@ncs,

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

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

Thanks
Shingei.

On Sun, Apr 10, 2011 at 1:42 AM, Alex Goh ncsalex@gmail.com wrote:

 Hi Guys,

 I'm trying to get the solutions for question 4.2 to work, but
 apparently I'm missing something and hope someone can help.
 I've search thru the list but doesn't really found a solution work for my
 case.

 The issue I've encounter are when HQ phone 5001 calling BR2 phone
 3003, 3003 ring, but when i tried to answered, the call drop.
 I know it might be related to codec issue, but I've my HQ-RTR
 configured with Xcoder which it is up and active but the call
 still failing. I also did have the trunk in cucm Wait for Far End
 H.245 Terminal Capability Set unchecked.

 once things I notice is that, my call doesn't seems get re-originated
 on the cube router to BR2 router, what I see during ringing state
 my show gatekeeper endpoint show the call is directly from the CUCM
 to BR2 It is only 2 call legs instead of 4 (see below).

 hm, what have I missed?

 Some Info:
 HQ Router (R1)

 interface Loopback0
  ip address 172.1.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R1
  h323-gateway voip bind srcaddr 172.1.254.1

 gatekeeper
  zone local UCM 172.1.254.1
  zone local UCME outvia VIA
  zone local VIA
  zone prefix UCME 3...
  gw-type-prefix 1#* default-technology
  no shutdown

 dial-peer voice 30 voip
  destination-pattern 3...
  session target ras
  codec g711ulaw
 !
 dial-peer voice 31 voip
  incoming called-number 3...

 Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
 LocalCallIDAge(secs)   BW
 511-32797  6   16(Kbps)
  Endpt(s): Alias E.164Addr
   src EP: gk_trunk_25001
   CallSignalAddr  Port  RASSignalAddr   Port
   172.1.10.20 38233 172.1.10.20 32795
  Endpt(s): Alias E.164Addr
   dst EP: R33003
   CallSignalAddr  Port  RASSignalAddr   Port
   172.3.254.1 1720  172.3.254.1 49395

GATEKEEPER ENDPOINT REGISTRATION

 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
 --- - --- - - -
 172.1.10.10 47142 172.1.10.10 32838 UCM   VOIP-GW
H323-ID: gk_trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
 172.1.10.20 38233 172.1.10.20 32795 UCM   VOIP-GW
H323-ID: gk_trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
 172.1.254.1 1720  172.1.254.2 56974 VIA   H323-GW
H323-ID: R1
Voice Capacity Max.=  Avail.=  Current.= 0
 172.3.254.1 1720  172.3.254.1 49395 VIA   H323-GW
H323-ID: R3
Voice Capacity Max.=  Avail.=  Current.= 0
 Total number of active registrations = 4

 R1(config-if)#do sh gatek gw
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 1#*(Default gateway-technology)
  Zone UCM master gateway list:
172.1.10.20:38233 gk_trunk_2
172.1.10.10:47142 gk_trunk_1
  Zone VIA master gateway list:
172.3.254.1:1720 R3
172.1.254.2:1720 R1

 BR2 Router (R2)

 interface Loopback0
  ip address 172.3.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R3
  h323-gateway voip tech-prefix 1#
  h323-gateway voip bind srcaddr 172.3.254.1

 dial-peer voice 10 voip
  incoming called-number 3...
  dtmf-relay rtp-nte
  codec g711ulaw
 !

 CUCM Trunk
 the trunk was assign a separate DP with a region that using G729 when
 calling HQ and BR2.



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

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

2011-04-09 Thread Alex Goh
Hi ShinGei,

Thanks for much for the reply, I guess I must be overlooked, the BR2
was registered to the wrong zone!
I'm going to try it on my lab tomorrow.

Again thanks!

Regards,
Alex

On Sun, Apr 10, 2011 at 2:18 AM, ShinGei Yong shingei.y...@gmail.com wrote:
 Hi Alex@ncs,

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

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

 Thanks
 Shingei.

 On Sun, Apr 10, 2011 at 1:42 AM, Alex Goh ncsalex@gmail.com wrote:

 Hi Guys,

 I'm trying to get the solutions for question 4.2 to work, but
 apparently I'm missing something and hope someone can help.
 I've search thru the list but doesn't really found a solution work for my
 case.

 The issue I've encounter are when HQ phone 5001 calling BR2 phone
 3003, 3003 ring, but when i tried to answered, the call drop.
 I know it might be related to codec issue, but I've my HQ-RTR
 configured with Xcoder which it is up and active but the call
 still failing. I also did have the trunk in cucm Wait for Far End
 H.245 Terminal Capability Set unchecked.

 once things I notice is that, my call doesn't seems get re-originated
 on the cube router to BR2 router, what I see during ringing state
 my show gatekeeper endpoint show the call is directly from the CUCM
 to BR2 It is only 2 call legs instead of 4 (see below).

 hm, what have I missed?

 Some Info:
 HQ Router (R1)

 interface Loopback0
  ip address 172.1.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R1
  h323-gateway voip bind srcaddr 172.1.254.1

 gatekeeper
  zone local UCM 172.1.254.1
  zone local UCME outvia VIA
  zone local VIA
  zone prefix UCME 3...
  gw-type-prefix 1#* default-technology
  no shutdown

 dial-peer voice 30 voip
  destination-pattern 3...
  session target ras
  codec g711ulaw
 !
 dial-peer voice 31 voip
  incoming called-number 3...

 Total number of active calls = 1.
                         GATEKEEPER CALL INFO
                         
 LocalCallID                        Age(secs)   BW
 511-32797                          6           16(Kbps)
  Endpt(s): Alias                 E.164Addr
   src EP: gk_trunk_2            5001
           CallSignalAddr  Port  RASSignalAddr   Port
           172.1.10.20     38233 172.1.10.20     32795
  Endpt(s): Alias                 E.164Addr
   dst EP: R3                    3003
           CallSignalAddr  Port  RASSignalAddr   Port
           172.3.254.1     1720  172.3.254.1     49395

                    GATEKEEPER ENDPOINT REGISTRATION
                    
 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type
  Flags
 --- - --- - -         
  -
 172.1.10.10     47142 172.1.10.10     32838 UCM               VOIP-GW
    H323-ID: gk_trunk_1
    Voice Capacity Max.=  Avail.=  Current.= 0
 172.1.10.20     38233 172.1.10.20     32795 UCM               VOIP-GW
    H323-ID: gk_trunk_2
    Voice Capacity Max.=  Avail.=  Current.= 0
 172.1.254.1     1720  172.1.254.2     56974 VIA               H323-GW
    H323-ID: R1
    Voice Capacity Max.=  Avail.=  Current.= 0
 172.3.254.1     1720  172.3.254.1     49395 VIA               H323-GW
    H323-ID: R3
    Voice Capacity Max.=  Avail.=  Current.= 0
 Total number of active registrations = 4

 R1(config-if)#do sh gatek gw
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 1#*    (Default gateway-technology)
  Zone UCM master gateway list:
    172.1.10.20:38233 gk_trunk_2
    172.1.10.10:47142 gk_trunk_1
  Zone VIA master gateway list:
    172.3.254.1:1720 R3
    172.1.254.2:1720 R1

 BR2 Router (R2)

 interface Loopback0
  ip address 172.3.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R3
  h323-gateway voip tech-prefix 1#
  h323-gateway voip bind srcaddr 172.3.254.1

 dial-peer voice 10 voip
  incoming called-number 3...
  dtmf-relay rtp-nte
  codec g711ulaw
 !

 CUCM Trunk
 the trunk was assign a separate DP with a region that using G729 when
 calling HQ and BR2.



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


___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)

2011-04-09 Thread Shrini

Hi Alex,

Uncheck wait for far end ... on Gatekeeper trunk configuration on CUCM.

Thanks
Shrini

On 4/9/2011 10:42 AM, Alex Goh wrote:

Hi Guys,

I'm trying to get the solutions for question 4.2 to work, but
apparently I'm missing something and hope someone can help.
I've search thru the list but doesn't really found a solution work for my case.

The issue I've encounter are when HQ phone 5001 calling BR2 phone
3003, 3003 ring, but when i tried to answered, the call drop.
I know it might be related to codec issue, but I've my HQ-RTR
configured with Xcoder which it is up and active but the call
still failing. I also did have the trunk in cucm Wait for Far End
H.245 Terminal Capability Set unchecked.

once things I notice is that, my call doesn't seems get re-originated
on the cube router to BR2 router, what I see during ringing state
my show gatekeeper endpoint show the call is directly from the CUCM
to BR2 It is only 2 call legs instead of 4 (see below).

hm, what have I missed?

Some Info:
HQ Router (R1)

interface Loopback0
  ip address 172.1.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R1
  h323-gateway voip bind srcaddr 172.1.254.1

gatekeeper
  zone local UCM 172.1.254.1
  zone local UCME outvia VIA
  zone local VIA
  zone prefix UCME 3...
  gw-type-prefix 1#* default-technology
  no shutdown

dial-peer voice 30 voip
  destination-pattern 3...
  session target ras
  codec g711ulaw
!
dial-peer voice 31 voip
  incoming called-number 3...

Total number of active calls = 1.
  GATEKEEPER CALL INFO
  
LocalCallIDAge(secs)   BW
511-32797  6   16(Kbps)
  Endpt(s): Alias E.164Addr
src EP: gk_trunk_25001
CallSignalAddr  Port  RASSignalAddr   Port
172.1.10.20 38233 172.1.10.20 32795
  Endpt(s): Alias E.164Addr
dst EP: R33003
CallSignalAddr  Port  RASSignalAddr   Port
172.3.254.1 1720  172.3.254.1 49395

 GATEKEEPER ENDPOINT REGISTRATION
 
CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
172.1.10.10 47142 172.1.10.10 32838 UCM   VOIP-GW
 H323-ID: gk_trunk_1
 Voice Capacity Max.=  Avail.=  Current.= 0
172.1.10.20 38233 172.1.10.20 32795 UCM   VOIP-GW
 H323-ID: gk_trunk_2
 Voice Capacity Max.=  Avail.=  Current.= 0
172.1.254.1 1720  172.1.254.2 56974 VIA   H323-GW
 H323-ID: R1
 Voice Capacity Max.=  Avail.=  Current.= 0
172.3.254.1 1720  172.3.254.1 49395 VIA   H323-GW
 H323-ID: R3
 Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 4

R1(config-if)#do sh gatek gw
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*(Default gateway-technology)
   Zone UCM master gateway list:
 172.1.10.20:38233 gk_trunk_2
 172.1.10.10:47142 gk_trunk_1
   Zone VIA master gateway list:
 172.3.254.1:1720 R3
 172.1.254.2:1720 R1

BR2 Router (R2)

interface Loopback0
  ip address 172.3.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R3
  h323-gateway voip tech-prefix 1#
  h323-gateway voip bind srcaddr 172.3.254.1

dial-peer voice 10 voip
  incoming called-number 3...
  dtmf-relay rtp-nte
  codec g711ulaw
!

CUCM Trunk
the trunk was assign a separate DP with a region that using G729 when
calling HQ and BR2.



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

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


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

2011-04-09 Thread Shrini

Also configure voice service on BR2 router

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-conn sip to sip
allow-conn sip to h323
h323

Thanks
Shrini

On 4/9/2011 10:42 AM, Alex Goh wrote:

Hi Guys,

I'm trying to get the solutions for question 4.2 to work, but
apparently I'm missing something and hope someone can help.
I've search thru the list but doesn't really found a solution work for my case.

The issue I've encounter are when HQ phone 5001 calling BR2 phone
3003, 3003 ring, but when i tried to answered, the call drop.
I know it might be related to codec issue, but I've my HQ-RTR
configured with Xcoder which it is up and active but the call
still failing. I also did have the trunk in cucm Wait for Far End
H.245 Terminal Capability Set unchecked.

once things I notice is that, my call doesn't seems get re-originated
on the cube router to BR2 router, what I see during ringing state
my show gatekeeper endpoint show the call is directly from the CUCM
to BR2 It is only 2 call legs instead of 4 (see below).

hm, what have I missed?

Some Info:
HQ Router (R1)

interface Loopback0
  ip address 172.1.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R1
  h323-gateway voip bind srcaddr 172.1.254.1

gatekeeper
  zone local UCM 172.1.254.1
  zone local UCME outvia VIA
  zone local VIA
  zone prefix UCME 3...
  gw-type-prefix 1#* default-technology
  no shutdown

dial-peer voice 30 voip
  destination-pattern 3...
  session target ras
  codec g711ulaw
!
dial-peer voice 31 voip
  incoming called-number 3...

Total number of active calls = 1.
  GATEKEEPER CALL INFO
  
LocalCallIDAge(secs)   BW
511-32797  6   16(Kbps)
  Endpt(s): Alias E.164Addr
src EP: gk_trunk_25001
CallSignalAddr  Port  RASSignalAddr   Port
172.1.10.20 38233 172.1.10.20 32795
  Endpt(s): Alias E.164Addr
dst EP: R33003
CallSignalAddr  Port  RASSignalAddr   Port
172.3.254.1 1720  172.3.254.1 49395

 GATEKEEPER ENDPOINT REGISTRATION
 
CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
172.1.10.10 47142 172.1.10.10 32838 UCM   VOIP-GW
 H323-ID: gk_trunk_1
 Voice Capacity Max.=  Avail.=  Current.= 0
172.1.10.20 38233 172.1.10.20 32795 UCM   VOIP-GW
 H323-ID: gk_trunk_2
 Voice Capacity Max.=  Avail.=  Current.= 0
172.1.254.1 1720  172.1.254.2 56974 VIA   H323-GW
 H323-ID: R1
 Voice Capacity Max.=  Avail.=  Current.= 0
172.3.254.1 1720  172.3.254.1 49395 VIA   H323-GW
 H323-ID: R3
 Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 4

R1(config-if)#do sh gatek gw
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*(Default gateway-technology)
   Zone UCM master gateway list:
 172.1.10.20:38233 gk_trunk_2
 172.1.10.10:47142 gk_trunk_1
   Zone VIA master gateway list:
 172.3.254.1:1720 R3
 172.1.254.2:1720 R1

BR2 Router (R2)

interface Loopback0
  ip address 172.3.254.1 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
  h323-gateway voip h323-id R3
  h323-gateway voip tech-prefix 1#
  h323-gateway voip bind srcaddr 172.3.254.1

dial-peer voice 10 voip
  incoming called-number 3...
  dtmf-relay rtp-nte
  codec g711ulaw
!

CUCM Trunk
the trunk was assign a separate DP with a region that using G729 when
calling HQ and BR2.



Regards,
Alex
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[OSL | CCIE_Voice] CME SRST: calling name configuration

2011-04-09 Thread Miron Kobelski
Hello,

I was playing with CME SRST today and I encountered the same issue again. I
configured CME SRST with srst mode auto-provision all.
Phones reregistered to SRST correctly, ephone and ephone-dn configuration
appeared in the config. By default, each ephone-dn is configured with CUCM
external phone mask as a calling name. Is it possible to change ephone's
calling name to something other in CME SRST?

When I changed the name under ephone-dn and restarted the phone. It
reregistered, but DN didn't appear on the button (couldn't make any calls).
Is it normal/expected behaviour or I missed something?


best regards
kobel
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Re: [OSL | CCIE_Voice] System prompts in UCCX

2011-04-09 Thread voice boy

Hi,

I'm not using the upload to repository method from prompt management page

I create a prompt variable in the script and point to the prompt directory and 
it always work perfect with me as i do it in my work projects


Do they need specific method in the exam ??


thanks
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[OSL | CCIE_Voice] cisco product page down?

2011-04-09 Thread CCIE for Me

getting 403 forbidden access code to cisco.com pages, anyone else experiencing 
this who is studying?

just trying to troubleshoot


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Re: [OSL | CCIE_Voice] System prompts in UCCX

2011-04-09 Thread Alex
Hi Miron
I use the prompts from the system folder directly without specifying the full 
path, but just the prompt name. The only thing need to be modified is to put S 
before P in the path ( SP[promptname]). If the prompt u r after is in sys 
folder subfolder, then u have to add this too (SP[ACD\promptname]). Also, make 
sure u using the correct sys folder corresponding to the language u configured, 
such as en_US. 
I always save my recorded prompts to the sys folders too.
Hope this helped.
Alex
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: Miron Kobelski findko...@gmail.com
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Sat, 9 Apr 2011 18:25:22 
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] System prompts in UCCX

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Re: [OSL | CCIE_Voice] CME Firmware upgrade

2011-04-09 Thread Mark Padilla

He this worked great, however I was wondering if anyone had some assistance on 
getting the phone loads for SIP. The SCCP loads came across, but I am missing 
the SIP loads. 
 


Date: Mon, 4 Apr 2011 18:31:37 -0500
Subject: Re: [OSL | CCIE_Voice] CME Firmware upgrade
From: whl...@gmail.com
To: findko...@gmail.com
CC: clane...@hotmail.com; ccie_voice@onlinestudylist.com

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinstl.html


the firmware for the phones is included with the download



On Mon, Apr 4, 2011 at 5:20 PM, Miron Kobelski findko...@gmail.com wrote:

Hi,

you'll find the firmware version needed in Release Notes. firmware upgrade 
procedure can be found in CME admin guide. however, it's easiest done by 
autoregistering the ip phones to CUCM.

regards
kobel





On Mon, Apr 4, 2011 at 23:45, Mark Padilla clane...@hotmail.com wrote:


Anyone have a document on how to download and update the firmware in a CME 7.1 
on a 2811? I am trying to update the firmware on my phones, however not sure on 
what firmware to load into flash:. 
 
 From: ccie_voice-requ...@onlinestudylist.com
 Subject: CCIE_Voice Digest, Vol 62, Issue 22
 To: ccie_voice@onlinestudylist.com
 Date: Mon, 4 Apr 2011 14:12:30 -0400
 
 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: CDR Reports (CCIE For ME)
 2. Volume 2 Lab 4 - Dial plan question (Randall Crumm)
 
 
 --
 
 Message: 1
 Date: Mon, 4 Apr 2011 12:02:11 -0400
 From: CCIE For ME cciefo...@hotmail.com
 To: Mark Padilla clane...@hotmail.com,
 ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CDR Reports
 Message-ID: blu111-ds613ed5a850ea0684f1230b0...@phx.gbl
 Content-Type: text/plain; charset=iso-8859-1
 
 to customize you need to export with a 3rd party app or use something like 
 crystal reports.
 
 From: Mark Padilla 
 Sent: Monday, April 04, 2011 11:27 AM
 To: ccie_voice@onlinestudylist.com 
 Subject: [OSL | CCIE_Voice] CDR Reports
 
 When I have doign CDR reports, I am looking at the Calling Party Number, 
 however all I see if the users calling a VM port number and I know this 
 cannot be correct. How can I csutomize this report, so I can see all of the 
 information, including the number the user is calling. 
 
  From: ccie_voice-requ...@onlinestudylist.com
  Subject: CCIE_Voice Digest, Vol 62, Issue 20
  To: ccie_voice@onlinestudylist.com
  Date: Mon, 4 Apr 2011 10:38:48 -0400
  
  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: lab 5 every location with SIP TRUNK (Jonny Mendas)
  2. Re: CCIE_Voice Digest, Vol 62, Issue 19 (justin...@gmail.com)
  3. AA Tcl (Ccie Voice)
  
  
  --
  
  Message: 1
  Date: Mon, 4 Apr 2011 08:03:54 +
  From: Jonny Mendas jonny.men...@hotmail.com
  To: naou...@mhdinfotech.com
  Cc: ccie_voice@onlinestudylist.com,
  ccie_voice-boun...@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK
  Message-ID: col106-w150b658e8715647b12f44b8d...@phx.gbl
  Content-Type: text/plain; charset=iso-8859-1
  
  
  Yes naoufal,
  
  I am the best... being triple CCIE i know it... but struggling a lot !!! i 
  feel VOICE is really hard :)
  
  Regarding my knowledge people can give thier number and we will have direct 
  chat with him ah ha :)
  
  there are countries where people cannot even attempt for one time but 
  those guys knowledge is more then a CCIE
  
  so here we all need to pass regarding the job and other stuff u will get it 
  by experience which is with everyone..
  
  I have a question :)
  
  Why Cisco take CCIE interview :) when they say he is expert common guys... 
  everyone now a days knows thier responsibility abt what they should know in 
  terms of experience/job
  but if u will feel from your experience u will clear your CCIE then i think 
  u require your next birth even then u wont be able to do it :)
  
  In all institue people go make back to 

Re: [OSL | CCIE_Voice] CME SRST: calling name configuration

2011-04-09 Thread adam compton
I've had the same problem.  I never knew what to think of it.  I assumed
that's just how it works.

On Sat, Apr 9, 2011 at 2:57 PM, Miron Kobelski findko...@gmail.com wrote:

 Hello,

 I was playing with CME SRST today and I encountered the same issue again. I
 configured CME SRST with srst mode auto-provision all.
 Phones reregistered to SRST correctly, ephone and ephone-dn configuration
 appeared in the config. By default, each ephone-dn is configured with CUCM
 external phone mask as a calling name. Is it possible to change ephone's
 calling name to something other in CME SRST?

 When I changed the name under ephone-dn and restarted the phone. It
 reregistered, but DN didn't appear on the button (couldn't make any calls).
 Is it normal/expected behaviour or I missed something?


 best regards
 kobel

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


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[OSL | CCIE_Voice] Changing Network Settings on SIP Firmware 7961

2011-04-09 Thread Mann Chaddha
Hi All

I am running Lab 9  am facing this issue. My home phone 7961 has a SIP
firmware on it  is trying to register to the Br1-CME (Voice Register
Global). I need to edit the TFTP Server settings on the phone but itas
prompting for a username  password.

What are the defaults for these?

Thanks
Mann
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