Re: [OSL | CCIE_Voice] vRack Vouchers for sale

[WorkerBee]

Hi Everyone,

Me too! I got 25 voice vouchers for sale.

Cheers

On Sat, Jan 29, 2011 at 1:43 AM, Jones, Brett
 wrote:
> Hi Everyone,
>
> I have 10 voice rack vouchers for sale @ $30 each, please email if you're 
> interested?
>
> Thanks
>
>
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[OSL | CCIE_Voice] OT - 25 x 8-hours Voice Proctor labs for Sale

[WorkerBee]

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

Re: [OSL | CCIE_Voice] 3rd day in raw , Xcoder still wouldn't kick in.

[WorkerBee]

Do you have the xcoder resources assigned to the phones correctly? MRGL?

On Sat, May 16, 2009 at 3:55 AM, jeremy co  wrote:
> Hi Guys,
>
> I have this issue for first time, dsp resources will not perform their job.
> MTP registered,Router couple of times has been rebooted. I did "test dsp
> device 1 1 reset" couple of times
>
> Here is my config : (to simplify testing ,I make this gateway h323 ,rigion
> G729 , incoming dial-peer G711u)
>
>
> sccp local FastEthernet0/0.11
> sccp ccm 142.4.64.114 identifier 1
> sccp
> !
> sccp ccm group 1
>  bind interface FastEthernet0/0.11
>  associate ccm 1 priority 1
>  associate profile 1 register MTP000cce1db0c0
> !
> dspfarm profile 1 transcode
>  codec g711ulaw
>  codec g711alaw
>  codec g729ar8
>  codec g729abr8
>  codec g729r8
>  maximum sessions 4
>  associate application SCCP
> !
> !
> !
> dial-peer voice 1000 voip
>  incoming called-number .
>  codec g711ulaw
>  no vad
> !
> !
> !
> !
> !
> telephony-service
>  ip source-address 142.4.64.114 port 2000
>  sdspfarm units 1
>  sdspfarm transcode sessions 4
>  sdspfarm tag 1 MTP000cce1db0c0
>
> !
>
> C2600#  sh sdsp sessions sum
>
>  max-mtps:1, max-streams:8, alloc-streams:8, act-streams:0
>   ID   MTP  State  CallID confID Usage
> Codec/Duration
>  = == === == =
> ==
> 1    1 IDLE   0   0
> G711Ulaw64k /20ms
> 2    1 IDLE   0   0
> G711Ulaw64k /20ms
> 3    1 IDLE   0   0
> G711Ulaw64k /20ms
> 4    1 IDLE   0   0
> G711Ulaw64k /20ms
> 5    1 IDLE   -1  0
> G711Ulaw64k /20ms
> 6    1 IDLE   -1  0
> G711Ulaw64k /20ms
> 7    1 IDLE   -1  0
> G711Ulaw64k /20ms
> 8    1 IDLE   -1  0
> G711Ulaw64k /20ms
>
>
> C2600#sh sdspfarm units
>
> mtp-1 Device:MTP000cce1db0c0 TCP socket:[1]  REGISTERED
> actual_stream:8 max_stream 8 IP:142.4.64.114  44154  MTP YOKO keepalive 82
> Supported codec: G711Ulaw
>  G711Alaw
>  G729
>  G729a
>  G729ab
>
>
>
> if I change incoming dial peer to G729 every thing is fine . so I don't have
> call routing sisue.
>
> Anybody face this before?
>
>
> Jeremy
>
>
>
>
>

Re: [OSL | CCIE_Voice] ISDN CCBs

[WorkerBee]

CCB = call control block

With each call connected, the CCB counter will increase by 1.

Total Allocated ISDN CCBs = Number of ISDN call control blocks that
are allocated.

http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a0080094b78.shtml



On Fri, May 8, 2009 at 5:34 PM, Deepak sidana  wrote:
> Hi,
>
> Can someone explain what is CCBs in ISDN?
>
> I have identical routers running the same ios, but one of my router
> show when PSTN call are active as =CCB:callid=7D74, sapi=0, ces=0, B-chan=6,
> calltype=VOICE
>
> but it also show=CCB:callid=7D96, sapi=0, ces=0 calltype=SIGNALING(wat this
> mean)
>
> I am wondreing its not showing calltype=SIGNALING on another router with
> identical setup.
>
> here is the output of  l3 "sh isdn status"
>
>
> Layer 3 Status:
>>
>> 24 Active Layer 3 Call(s)
>>
>> CCB:callid=6407, sapi=0, ces=0, B-chan=8, calltype=VOICE
>>
>> CCB:callid=65EB, sapi=0, ces=0, B-chan=29, calltype=VOICE
>>
>> CCB:callid=65F4, sapi=0, ces=0, B-chan=3, calltype=VOICE
>>
>> CCB:callid=6611, sapi=0, ces=0, B-chan=12, calltype=VOICE
>>
>> CCB:callid=66AC, sapi=0, ces=0, B-chan=28, calltype=VOICE
>>
>> CCB:callid=682D, sapi=0, ces=0, B-chan=20, calltype=VOICE
>>
>> CCB:callid=6834, sapi=0, ces=0, B-chan=23, calltype=VOICE
>>
>> CCB:callid=6950, sapi=0, ces=0, B-chan=31, calltype=VOICE
>>
>> CCB:callid=6956, sapi=0, ces=0, B-chan=4, calltype=VOICE
>>
>> CCB:callid=695F, sapi=0, ces=0, B-chan=21, calltype=VOICE
>>
>> CCB:callid=6979, sapi=0, ces=0, B-chan=1, calltype=VOICE
>>
>> CCB:callid=6B8F, sapi=0, ces=0, B-chan=7, calltype=VOICE
>>
>> CCB:callid=E77C, sapi=0, ces=0, B-chan=22, calltype=VOICE
>>
>> CCB:callid=73AB, sapi=0, ces=0, B-chan=27, calltype=VOICE
>>
>> CCB:callid=73AF, sapi=0, ces=0, B-chan=19, calltype=VOICE
>>
>> CCB:callid=E822, sapi=0, ces=0, B-chan=30, calltype=VOICE
>>
>> CCB:callid=79D8, sapi=0, ces=0, B-chan=13, calltype=VOICE
>>
>> CCB:callid=7A9A, sapi=0, ces=0, B-chan=24, calltype=VOICE
>>
>> CCB:callid=7AB6, sapi=0, ces=0, B-chan=25, calltype=VOICE
>>
>> CCB:callid=7B1F, sapi=0, ces=0, B-chan=9, calltype=VOICE
>>
>> CCB:callid=7BD6, sapi=0, ces=0, B-chan=15, calltype=VOICE
>>
>> CCB:callid=7C23, sapi=0, ces=0, B-chan=5, calltype=VOICE
>>
>> CCB:callid=7D07, sapi=0, ces=0, B-chan=10, calltype=VOICE
>>
>> CCB:callid=7D27, sapi=0, ces=0, B-chan=14, calltype=VOICE
>>
>> CCB:callid=7D2C, sapi=0, ces=0, B-chan=26, calltype=VOICE
>>
>> CCB:callid=7D73, sapi=0, ces=0, B-chan=2, calltype=VOICE
>>
>> CCB:callid=7D74, sapi=0, ces=0, B-chan=6, calltype=VOICE
>>
>> CCB:callid=7D96, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7D97, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7D98, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7D99, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7D9E, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7D9F, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA0, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA1, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA2, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA3, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA4, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA5, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA6, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA7, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA8, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DA9, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DAA, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DAB, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DAC, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DB9, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DBA, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DBB, sapi=0, ces=0, B-chan=11, calltype=VOICE
>>
>> CCB:callid=7DBC, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DBD, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DBE, sapi=0, ces=0 calltype=SIGNALING
>>
>> CCB:callid=7DBF, sapi=0, ces=0 calltype=SIGNALING
>>
>> Active dsl 0 CCBs = 53
>>
>> The Free Channel Mask:  0x8003
>>
>> Number of L2 Discards = 0, L2 Session ID = 0
>>
>
>
> Regards
> Deepak
>
> 
> Cricket on your mind? Visit the ultimate cricket website. Enter now!

Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?

[WorkerBee]

It seems like whenever I reboot the router, I always need to yank off
the "isdn layer 3 bind ccm-manager", and reapply
to get isdn into multiple frame established status.

On Tue, Apr 28, 2009 at 11:49 PM, Bill Talley  wrote:
> If you have TEI assigned,  call manager won't send the call to the gateway 
> and you wouldn't see q931 messages, would you?  Port won't be registered in 
> call manager and call would fail at the route list.
>
> Sent from a mobile phone with very tiny touchscreen keys.  Please excuse my 
> typos.
>
> -Original Message-
> From: Cliff McGlamry 
> Sent: Tuesday, April 28, 2009 10:39 AM
> To: WorkerBee 
> Cc: ccie_voice@onlinestudylist.com 
> Subject: Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>
> Few things to try
>
> 1.  If you're telneted into the router (as opposed to being on a console
> connection), make sure you issue the terminal monitor command or you won't
> see the output.
>
> 2.  Did you check / reset the route list that the route pattern is pointed
> to in CCM?  If the route list is in a flakey state, nothing will go out, but
> incoming calls may work.
>
> 3.  Try an inbound call.  See what shows up in the traces then.
>
> This is where I would start.
>
>
> - Original Message -
> From: "WorkerBee" 
> To: "Cliff McGlamry" 
> Cc: "Tech Guy" ; 
> Sent: Tuesday, April 28, 2009 11:30 AM
> Subject: Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>
>
> Cliff, how do you check if route pattern is working? I use the Dial
> Analyzer, but it is not a real-time digit dialing capturing tool.
>
> I don't see any thing on the PSTN side. No debug traces output.
>
> I think it must be stuck within CCM. I have check
> CSS (calling device) and Partition (911). It seems to be correct and
> allowed.
>
> Any idea how to proceed?
>
> On Tue, Apr 28, 2009 at 10:59 PM, Cliff McGlamry  wrote:
>> What does the debug isdn q931 trace show when you attempt the call?
>>
>>
>> - Original Message -
>> From: "WorkerBee" 
>> To: "Tech Guy" 
>> Cc: 
>> Sent: Tuesday, April 28, 2009 10:47 AM
>> Subject: Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>>
>>
>> @Tech Guy. You got it spot on. It works. However, still no joy for
>> making a call out the PSTN.
>> I notice the dial-peer, OUT STAT is down. Is this correct?
>>
>> In the CCM, the route pattern 911 points to the MGCP gateway.
>>
>>
>> dial-peer voice 10 pots
>> service mgcpapp !-- if I omit this, OPER = down, this is not
>> necessary since I have bind l3 at serial interface
>> port 0/2/0:23
>>
>> HQ#show dial-peer voice summary
>> dial-peer hunt 0
>> AD PRE PASS
>> OUT
>> TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU
>> SESS-TARGET STAT PORT
>> 10 pots up up 0
>> down 0/2/0:23
>> HQ#
>>
>>
>> HQ#show isdn status
>> Global ISDN Switchtype = primary-ni
>>
>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
>> may not apply
>>
>> ISDN Serial0/2/0:23 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: 0x807F
>> Number of L2 Discards = 0, L2 Session ID = 0
>> Total Allocated ISDN CCBs = 0
>> HQ#
>>
>>
>>
>>
>>
>> On Tue, Apr 28, 2009 at 8:57 AM, Tech Guy  wrote:
>>> Unbind the D channel layer-3 from callmanager and re-bind it again.
>>> Multiple
>>> frame should be established after that, if not reboot the router.
>>>
>>> Tech Guy
>>>
>>> - Original Message - From: "WorkerBee" 
>>> To: 
>>> Sent: Monday, April 27, 2009 8:08 PM
>>> Subject: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>>>
>>>
>>>> Previously, I have configured for H323 and the ISDN Layer 2 is
>>>> working. I can make test call out to PSTN.
>>>> However, I have problem getting ISDN Layer 2 up with
>>>> MULTIPLE_FRAME_ESTABLISHED for MGCP config.
>>>>
>>>> CCM - 4.1.(3)sr8a
>>>> HQ Router - Cisco 2811 (C2800NM-ADVENTERPRISEK9_IVS_LI-M)
>>>>
>>>> HQ# debug isdn q921
>>>>
>>>> *Apr 28 01:20:39.327: ISDN Se0/2/0:23 Q921: U

Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?

[WorkerBee]

Cliff, how do you check if route pattern is working? I use the Dial
Analyzer, but it is not a real-time digit dialing capturing tool.

I don't see any thing on the PSTN side. No debug traces output.

I think it must be stuck within CCM. I have check
CSS (calling device) and Partition (911). It seems to be correct and allowed.

Any idea how to proceed?

On Tue, Apr 28, 2009 at 10:59 PM, Cliff McGlamry  wrote:
> What does the debug isdn q931 trace show when you attempt the call?
>
>
> - Original Message -
> From: "WorkerBee" 
> To: "Tech Guy" 
> Cc: 
> Sent: Tuesday, April 28, 2009 10:47 AM
> Subject: Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>
>
> @Tech Guy. You got it spot on. It works. However, still no joy for
> making a call out the PSTN.
> I notice the dial-peer, OUT STAT is down. Is this correct?
>
> In the CCM, the route pattern 911 points to the MGCP gateway.
>
>
> dial-peer voice 10 pots
>  service mgcpapp       !-- if I omit this, OPER = down, this is not
> necessary since I have bind l3 at serial interface
>  port 0/2/0:23
>
> HQ#show dial-peer voice summary
> dial-peer hunt 0
>             AD                                    PRE PASS
> OUT
> TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU
> SESS-TARGET    STAT PORT
> 10     pots  up   up                                0
>    down 0/2/0:23
> HQ#
>
>
> HQ#show isdn status
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
> may not apply
>
> ISDN Serial0/2/0:23 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:  0x807F
>    Number of L2 Discards = 0, L2 Session ID = 0
>    Total Allocated ISDN CCBs = 0
> HQ#
>
>
>
>
>
> On Tue, Apr 28, 2009 at 8:57 AM, Tech Guy  wrote:
>> Unbind the D channel layer-3 from callmanager and re-bind it again.
>> Multiple
>> frame should be established after that, if not reboot the router.
>>
>> Tech Guy
>>
>> - Original Message - From: "WorkerBee" 
>> To: 
>> Sent: Monday, April 27, 2009 8:08 PM
>> Subject: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>>
>>
>>> Previously, I have configured for H323 and the ISDN Layer 2 is
>>> working. I can make test call out to PSTN.
>>> However, I have problem getting ISDN Layer 2 up with
>>> MULTIPLE_FRAME_ESTABLISHED for MGCP config.
>>>
>>> CCM - 4.1.(3)sr8a
>>> HQ Router - Cisco 2811 (C2800NM-ADVENTERPRISEK9_IVS_LI-M)
>>>
>>> HQ# debug isdn q921
>>>
>>> *Apr 28 01:20:39.327: ISDN Se0/2/0:23 Q921: User RX <- SABMEp sapi=0
>>> tei=0
>>> *Apr 28 01:20:39.331: ISDN Se0/2/0:23 Q921: S4_SABME: BACKHAULED &
>>> vsc_wants_L2_up = FALSE
>>>
>>>
>>> = ISDN Status =
>>>
>>> HQ#show isdn status
>>> Global ISDN Switchtype = primary-ni
>>>
>>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
>>> may not apply
>>>
>>> ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
>>> Layer 3 Status:
>>> 0 Active Layer 3 Call(s)
>>> Active dsl 0 CCBs = 0
>>> The Free Channel Mask: 0x807F
>>> Number of L2 Discards = 0, L2 Session ID = 1
>>> Total Allocated ISDN CCBs = 0
>>> HQ#
>>>
>>>  CCM Layer 3 Backhaul Status =
>>>
>>> HQ#show ccm-manager
>>> MGCP Domain Name: HQ
>>> Priority Status Host
>>> 
>>> Primary Registered 10.34.200.20
>>> First Backup None
>>> Second Backup None
>>>
>>> Current active Call Manager: 10.34.200.20
>>> Backhaul/Redundant link port: 2428
>>> Failover Interval: 30 seconds
>>> Keepalive Interval: 15 seconds
>>> Last keepalive sent: 01:24:11 UTC Apr 28 2009 (elapsed
>>> time: 00:00:12)
>>> Last MGCP traffic time: 01:24:21 UTC Apr 28 2009 (elapsed
>>> time: 00:00:02)
>>> Las

Re: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?

[WorkerBee]

@Tech Guy. You got it spot on. It works. However, still no joy for
making a call out the PSTN.
I notice the dial-peer, OUT STAT is down. Is this correct?

In the CCM, the route pattern 911 points to the MGCP gateway.


dial-peer voice 10 pots
 service mgcpapp   !-- if I omit this, OPER = down, this is not
necessary since I have bind l3 at serial interface
 port 0/2/0:23

HQ#show dial-peer voice summary
dial-peer hunt 0
 ADPRE PASSOUT
TAGTYPE  MIN  OPER PREFIXDEST-PATTERN  FER THRU
SESS-TARGETSTAT PORT
10 pots  up   up0
down 0/2/0:23
HQ#


HQ#show isdn status
Global ISDN Switchtype = primary-ni

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

ISDN Serial0/2/0:23 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:  0x807F
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0
HQ#





On Tue, Apr 28, 2009 at 8:57 AM, Tech Guy  wrote:
> Unbind the D channel layer-3 from callmanager and re-bind it again. Multiple
> frame should be established after that, if not reboot the router.
>
> Tech Guy
>
> - Original Message - From: "WorkerBee" 
> To: 
> Sent: Monday, April 27, 2009 8:08 PM
> Subject: [OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?
>
>
>> Previously, I have configured for H323 and the ISDN Layer 2 is
>> working. I can make test call out to PSTN.
>> However, I have problem getting ISDN Layer 2 up with
>> MULTIPLE_FRAME_ESTABLISHED for MGCP config.
>>
>> CCM - 4.1.(3)sr8a
>> HQ Router - Cisco 2811 (C2800NM-ADVENTERPRISEK9_IVS_LI-M)
>>
>> HQ# debug isdn q921
>>
>> *Apr 28 01:20:39.327: ISDN Se0/2/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
>> *Apr 28 01:20:39.331: ISDN Se0/2/0:23 Q921: S4_SABME: BACKHAULED &
>> vsc_wants_L2_up = FALSE
>>
>>
>> = ISDN Status =
>>
>> HQ#show isdn status
>> Global ISDN Switchtype = primary-ni
>>
>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
>> may not apply
>>
>> ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
>>   Layer 3 Status:
>>       0 Active Layer 3 Call(s)
>>   Active dsl 0 CCBs = 0
>>   The Free Channel Mask:  0x807F
>>   Number of L2 Discards = 0, L2 Session ID = 1
>>   Total Allocated ISDN CCBs = 0
>> HQ#
>>
>>  CCM Layer 3 Backhaul Status =
>>
>> HQ#show ccm-manager
>> MGCP Domain Name: HQ
>> Priority        Status                   Host
>> 
>> Primary         Registered               10.34.200.20
>> First Backup    None
>> Second Backup   None
>>
>> Current active Call Manager:    10.34.200.20
>> Backhaul/Redundant link port:   2428
>> Failover Interval:              30 seconds
>> Keepalive Interval:             15 seconds
>> Last keepalive sent:            01:24:11 UTC Apr 28 2009 (elapsed
>> time: 00:00:12)
>> Last MGCP traffic time:         01:24:21 UTC Apr 28 2009 (elapsed
>> time: 00:00:02)
>> Last failover time:             None
>> Last switchback time:           None
>> Switchback mode:                Graceful
>> MGCP Fallback mode:             Not Selected
>> Last MGCP Fallback start time:  None
>> Last MGCP Fallback end time:    None
>> MGCP Download Tones:            Disabled
>> TFTP retry count to shut Ports: 2
>>
>> Backhaul Link info:
>>   Link Protocol:      TCP
>>   Remote Port Number: 2428
>>   Remote IP Address:  10.34.200.20
>>   Current Link State: OPEN
>>   Statistics:
>>       Packets recvd:   0
>>       Recv failures:   0
>>       Packets xmitted: 0
>>       Xmit failures:   0
>>   PRI Ports being backhauled:
>>       Slot 0, VIC 2, port 0
>> FAX mode: cisco
>> Configuration Error History:
>> HQ#
>>
>>
>>  MGCP Config on HQ Router =
>>
>>
>> card type t1 0 2
>> network-clock-participate wic 2
>>
>> isdn switch-type primary-ni
>

[OSL | CCIE_Voice] MGCP - L2 TEI_ASSIGNED?

[WorkerBee]

Previously, I have configured for H323 and the ISDN Layer 2 is
working. I can make test call out to PSTN.
However, I have problem getting ISDN Layer 2 up with
MULTIPLE_FRAME_ESTABLISHED for MGCP config.

CCM - 4.1.(3)sr8a
HQ Router - Cisco 2811 (C2800NM-ADVENTERPRISEK9_IVS_LI-M)

HQ# debug isdn q921

*Apr 28 01:20:39.327: ISDN Se0/2/0:23 Q921: User RX <- SABMEp sapi=0 tei=0
*Apr 28 01:20:39.331: ISDN Se0/2/0:23 Q921: S4_SABME: BACKHAULED &
vsc_wants_L2_up = FALSE


= ISDN Status =

HQ#show isdn status
Global ISDN Switchtype = primary-ni

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

ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask:  0x807F
Number of L2 Discards = 0, L2 Session ID = 1
Total Allocated ISDN CCBs = 0
HQ#

 CCM Layer 3 Backhaul Status =

HQ#show ccm-manager
MGCP Domain Name: HQ
PriorityStatus   Host

Primary Registered   10.34.200.20
First BackupNone
Second Backup   None

Current active Call Manager:10.34.200.20
Backhaul/Redundant link port:   2428
Failover Interval:  30 seconds
Keepalive Interval: 15 seconds
Last keepalive sent:01:24:11 UTC Apr 28 2009 (elapsed
time: 00:00:12)
Last MGCP traffic time: 01:24:21 UTC Apr 28 2009 (elapsed
time: 00:00:02)
Last failover time: None
Last switchback time:   None
Switchback mode:Graceful
MGCP Fallback mode: Not Selected
Last MGCP Fallback start time:  None
Last MGCP Fallback end time:None
MGCP Download Tones:Disabled
TFTP retry count to shut Ports: 2

Backhaul Link info:
Link Protocol:  TCP
Remote Port Number: 2428
Remote IP Address:  10.34.200.20
Current Link State: OPEN
Statistics:
Packets recvd:   0
Recv failures:   0
Packets xmitted: 0
Xmit failures:   0
PRI Ports being backhauled:
Slot 0, VIC 2, port 0
FAX mode: cisco
Configuration Error History:
HQ#


 MGCP Config on HQ Router =


card type t1 0 2
network-clock-participate wic 2

isdn switch-type primary-ni


controller T1 0/2/0
 cablelength long 0db
 pri-group timeslots 1-24 service mgcp


interface Serial0/2/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable

voice-port 0/2/0:23


ccm-manager mgcp

ccm-manager music-on-hold
ccm-manager config server 10.34.200.20
ccm-manager config

mgcp
mgcp call-agent 10.34.200.20 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 ecm
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0.640
mgcp bind media source-interface FastEthernet0/0.640

mgcp profile default

dial-peer voice 10 pots
 service mgcpapp
 port 0/2/0:23

Re: [OSL | CCIE_Voice] is there a phone remote control to test moh ios multicast br1 via vracks?

[WorkerBee]

Alternatively, you can log into the remote router and do "show ip
mroute" t see that multicast is flowing.Also look at IP IGMP messages
as well.

On Thu, Apr 23, 2009 at 9:31 AM, Duy Nguyen  wrote:
> Web into phone doesn't show mcast traffic.
>
> On Wed, Apr 22, 2009 at 7:01 PM,  wrote:
>>
>> i think you can just web into the phone ip, see if there is mcat traffic?
>>
>>
>> zamuel del Toro  wrote:
>>
>> Some bady know if exist a phone remote control to test moh ios on br2. I
>> need that the phone be behind br1 router.?
>> I probe with  voip integration phone remote control but not work on vrack.
>> on my own work fine
>>
>> thanks
>>
>> 
>> Windows Live Hotmail now works up to 70% faster. Sign up today.
>> 
>> Power up the Internet with Yahoo! Toolbar.
>
>

[OSL | CCIE_Voice] 2811 ISDN status

[WorkerBee]

For BR1/HQ site running Cisco 2811 router under MGCP, when you do a
show isdn status, what is considered the normal status?

TEI_ASSIGNED or MULTIPLE_FRAME_ESTABLISHED ?


When a "show ccm-manager" is issued, the Cisco 2811 is registered to the CCM.

Re: [OSL | CCIE_Voice] Version 3 Lab equipment...

[WorkerBee]

Probably phones maybe upgraded to 7961 series "Type B" that support
more presence features compare to "Type A" phones such as 7940, 7960.




On Wed, Apr 22, 2009 at 1:50 AM, Chris Parker  wrote:
> I would say another HWIC-4ESW, and seven 79XX - phones 3 for HQ, 2 for BR1
> and 2 for BR2
>
> Jonathan Charles wrote:
>>
>> Well, I have an NME-CUE, two 2821s, and a 2811, a 3750, HWIC-4ESW and a
>> pile of DL 380s... I am just curious what I am missing...
>>
>> On Mon, Apr 20, 2009 at 7:56 PM, Chris Parker > > wrote:
>>
>>
>>    I think BR2 will also be an NM-ESW type device. Suppose it could
>>    also be another 3750. I'm using HWIC-ESW for BR1/2 in my lab and a
>>    3560 at HQ. Not sure why they are using 3800's per the blue print.
>>    I think you can do everything with 2801/11. In my lab BR1/2 are
>>    2801 and HQ is 2811. The only trick with using 2801 is memory. You
>>    need at least 256MB for the 12.4(24)T IOS.
>>
>>    Chris
>>
>>
>>    Jonathan Charles wrote:
>>
>>        OK, I have abandoned my attempts at the V2 lab (just not
>>        enough time, and a little too retro for me...)
>>
>>        So, I am curious about the topology...
>>
>>        3750 at HQ, makes sense...
>>
>>        4 port HWIC at BR1... ok...
>>
>>        What is happening at BR2, switch-wise?
>>
>>        Also... why are the routers so beefy at HQ and BR1? Are there
>>        other modules in there we should be aware of?
>>
>>        What am I missing? I have about 20 phones here..
>>
>>        Also, when is IPExpert going to update their doco? I
>>        downloaded some spartan labs... but there is not a lot of info
>>        there... when the upgrade comes, how much? My lab is currently
>>        scheduled for 27 JULY
>>
>>
>>
>>        Jonathan
>>
>>
>>
>>
>
>

Re: [OSL | CCIE_Voice] One Way Audio issue

[WorkerBee]

Probably check out the routing path for the RTP streams of the endpoints.

On Fri, Mar 27, 2009 at 11:00 AM, Cristobal Priego
 wrote:
> No, the path from the PRI to the phone is clean, no firewalls, no ACLs, I
> was thinking of doing a sniffer trace on the agent phone. have the traces
> set to detailed and check the StationOpenReceiveChannel, and the ack. and
> then check that against the sniffer trace, also I'd like to sniff like to
> sniff the other end, but I don't know where since the call is coming from
> the PSTN.
> Sorry to bug you with so many questions, this happens when you are not an
> expert and you have the preassure of everybody else
>
>
>
> 2009/3/26 WorkerBee 
>>
>> Is there any firewall or ACL that sit between both parties?
>>
>> On Fri, Mar 27, 2009 at 10:25 AM, Cristobal Priego
>>  wrote:
>> > The agents are able to hear the caller, the caller cannot hear the
>> > agents.
>> > the stream is getting lost on the way out
>> >
>> > 2009/3/26 Cliff McGlamry 
>> >>
>> >> Which way is audio being lost?  Can the agents hear the caller or do
>> >> they
>> >> hear dead air?
>> >>
>> >> Bottom line, you're losing the media stream in one direction.  Media
>> >> stream is being lost between the the person who cannot be heard toward
>> >> the
>> >> person who can't hear.
>> >>
>> >> Identify who can't hear who, and we eliminate half of what needs to be
>> >> looked at.
>> >>
>> >> Cliff
>> >>
>> >>
>> >> - Original Message -
>> >> From: Cristobal Priego
>> >> To: OSL Group
>> >> Sent: Thursday, March 26, 2009 10:19 PM
>> >> Subject: [OSL | CCIE_Voice] One Way Audio issue
>> >> Hello Friends,
>> >>
>> >> I have an issue with one way audio for calls coming from the PSTN.
>> >>
>> >> PSTN->7200(PRI)->CCM/UCCX->switch->switch->switch->7961
>> >>
>> >> I check connectivity, I check the routing however some users are having
>> >> one way audio issues. all of them are Agents in UCCX
>> >>
>> >> thanks
>> >
>> >
>
>

Re: [OSL | CCIE_Voice] One Way Audio issue

[WorkerBee]

Is there any firewall or ACL that sit between both parties?

On Fri, Mar 27, 2009 at 10:25 AM, Cristobal Priego
 wrote:
> The agents are able to hear the caller, the caller cannot hear the agents.
> the stream is getting lost on the way out
>
> 2009/3/26 Cliff McGlamry 
>>
>> Which way is audio being lost?  Can the agents hear the caller or do they
>> hear dead air?
>>
>> Bottom line, you're losing the media stream in one direction.  Media
>> stream is being lost between the the person who cannot be heard toward the
>> person who can't hear.
>>
>> Identify who can't hear who, and we eliminate half of what needs to be
>> looked at.
>>
>> Cliff
>>
>>
>> - Original Message -
>> From: Cristobal Priego
>> To: OSL Group
>> Sent: Thursday, March 26, 2009 10:19 PM
>> Subject: [OSL | CCIE_Voice] One Way Audio issue
>> Hello Friends,
>>
>> I have an issue with one way audio for calls coming from the PSTN.
>>
>> PSTN->7200(PRI)->CCM/UCCX->switch->switch->switch->7961
>>
>> I check connectivity, I check the routing however some users are having
>> one way audio issues. all of them are Agents in UCCX
>>
>> thanks
>
>

Re: [OSL | CCIE_Voice] Understanding Transcoding

[WorkerBee]

MRG of the device, Region from CCM (hub/spoke), Dial-Peer codec
preference, phone 729/711 capabilities are part of the equation of who
is performing the transcoding.



On Thu, Mar 26, 2009 at 11:26 AM, Cliff McGlamry  wrote:
> Transcoding is invoked on the device where the mismatch occurs.  If CME has
> one dial peer that requires G711, and one with G729, then it will occur
> there (think of going to CUE).  But the dial-peer to CCM is hard coded to
> G729.
>
> CCM on the other hand has a variety of codecs, but typically the device pool
> for HQ is using G711.  The trunk it knows is hard coded to G729.  In that
> case, transcoding occurs on the CCM side.
>
> It may have to occur again (if the call forwards to CUE voicemail on the
> other end), but if it doesn't it isn't required.
>
>
>
> - Original Message -
> From: ccielab...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Sent: Wednesday, March 25, 2009 10:26 PM
> Subject: [OSL | CCIE_Voice] Understanding Transcoding
> I'm a little confused about how transcoding is invoked.
> Assuming I have an endpoint that makes a call (g.729) to another endpoint
> requiring g.711u.
> Is it ALWAYS the g.711 side that must allocate the transcoder resource?
>
>
>

Re: [OSL | CCIE_Voice] VMware ESXi for CCM servers

[WorkerBee]

Does NTP sync between Pub and Sub still an issue with ESXi server?

On Mon, Mar 23, 2009 at 9:04 AM, Jeff Garvas  wrote:
> It's windows 2003 Server.    You install one of the four DVDs and then
> install UCCx on top of that.
>
> On Sun, Mar 22, 2009 at 7:41 PM, Duy Nguyen  wrote:
>>
>> After install, you can give it limited memory and it will work.  Initial
>> is when it needs the resources.  Does UCCx 7 runs on Windows or is it
>> Linux.  If it's Linux just choose redhat 4/5.
>>
>> On Sun, Mar 22, 2009 at 4:16 PM, Jeff Garvas  wrote:
>>>
>>> I have a P4 3Ghz with 4gb running ESXi.   One thing I've read and
>>> experienced is that installing just about anything with less than the
>>> minimum disk/memory is asking for trouble.   In some unique instalaltions
>>> you'll get it to finish but it won't work right, services won't start right,
>>> etc.
>>>
>>> Has anyone here been successful installing UCCx 7 in ESXi?   If so, what
>>> platform server OS did you use or does it not matter?   I have the four
>>> HP/IBM CDs and I have not attempted this yet.
>>>
>>> Also, the OS won't install without 2gb of memory.   I tried to install it
>>> on a IBM box that only had a gig and figured that one out.   Anyone know if
>>> I can scale back the memory after installing everything or does it need that
>>> much to operate at all?
>>>
>>>
>>
>
>

Re: [OSL | CCIE_Voice] VMware ESXi for CCM servers

[WorkerBee]

How many virtual servers can u put in ESXi? There are quite a number
of servers required for the new blueprint.

On Sun, Mar 22, 2009 at 1:09 PM, Arun Kumar  wrote:
> during install select 73GB HDD and 2GB RAM after install you can reduce the
> amount of memory.
>
> On Sun, Mar 22, 2009 at 3:03 AM, J Hogan  wrote:
>>
>> I managed to get it running on 40 gig and 512 mem of ram
>>
>> On 3/21/09, Duy Nguyen  wrote:
>> > Make sure during the initial install that you allocate 74GB of hd and @
>> > least 2GB of ram.
>> >
>> > On Sat, Mar 21, 2009 at 2:35 PM,  wrote:
>> >
>> >> Any tricks to getting CUCM 7 working on ESXi ? My install keeps failing
>> >> .
>> >> I haven't invested too much time yet, as I'm trying hard to pass with
>> >> the
>> >> current blueprint :)
>> >>
>> >>
>> >>
>> >> On Sat, Mar 21, 2009 at 1:15 PM, Arun Kumar  wrote:
>> >>
>> >>> Hi
>> >>>
>> >>> I'm running CUCM 7 and Unity Connection 7 on ESXi with 6GB of RAM and
>> >>> 500GB of HDD and it's working fine. Not tested on Linux.
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>> On Sat, Mar 21, 2009 at 6:44 PM, WorkerBee  wrote:
>> >>>
>> >>>> Anyone has tried using ESXi with Quad core/8G ram instead of using
>> >>>> VMware server on a Linux?
>> >>>>
>> >>>> Does ESXi gives a better performance?
>> >>>>
>> >>>> Thanks.
>> >>>>
>> >>>
>> >>>
>> >>
>> >
>>
>> --
>> Sent from my mobile device
>>
>> J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI
>> Yahoo ID: jhogan552000
>> AIM ID: jhogan55
>> MSN ID: jhogan55
>> ICQ ID: 257599283
>>
>> Live Life And Do Not Kill Time.
>
>

[OSL | CCIE_Voice] VMware ESXi for CCM servers

[WorkerBee]

Anyone has tried using ESXi with Quad core/8G ram instead of using
VMware server on a Linux?

Does ESXi gives a better performance?

Thanks.

Re: [OSL | CCIE_Voice] gatekeeper registration

[WorkerBee]

Did you configure "h323-gateway voip bind srcaddr" for CUME to 172.25.102.1/32?


On Sat, Mar 14, 2009 at 10:48 PM, Josh Reinmann  wrote:
> When I have a gatekeeper set up without any security, I am able to get both
> CCM’s to register no problem.  Once I put security on, only 1 registers. I’m
> not sure how to get both to register.  My gatekeeper config is below:
>
>
>
> gatekeeper
>
>  zone local UCM ipexpert.com 172.25.100.1
>
>  zone local UCME ipexpert.com
>
>  no zone subnet UCM default enable
>
>  zone subnet UCM 10.25.200.20/32 enable
>
>  zone subnet UCM 10.25.200.21/32 enable
>
>  no zone subnet UCME default enable
>
>  zone subnet UCME 172.25.102.1/32 enable
>
>  no zone subnet UCME 10.25.202.1/32 enable
>
>  no shutdown
>
>
>
> Thanks,
>
>
>
> Josh Reinmann, MCSE/CCVP
> Fidelus Technologies, LLC
> 240 W. 35th St, 605
> New York, NY 10001
> Phone: 212-616-7840
> Fax: 212-616-7850
> Cell: 917-453-2312
> Email: jreinm...@fidelus.com
>
> “Erasing the Lines Between Voice & Data”
>
> P please consider the environment before printing this e-mail
>
>

Re: [OSL | CCIE_Voice] QOS Priority Queing question

[WorkerBee]

priority {bandwidth-kbps | percent percentage} [burst]

The default burst value, which is computed as 200 milliseconds of
traffic at the configured bandwidth rate.

http://www.cisco.com/en/US/docs/ios/qos/command/reference/qos_n1.html#wp1032158


0.2 x 156000 = 31200 = 3,900 bytes

Which is what you have configured.

I believe if you use the percentage, the burst value is taken the
default of 200 msec of the bw rate.



On Fri, Feb 13, 2009 at 2:11 AM, Cliff McGlamry  wrote:
> I've looked through the SRND, and can't find this one.
>
> If you configure the priority queue as bandwidth percent, then the outputs
> of a show policy-map look as expected.  But if I configure it as a fixed
> value, I'm getting something else:
>
> R1#show run | b policy-map
> policy-map WAN-EDGE-R1
>  class RTP
>   priority 156
>
>
> R1#sh policy-map WAN-EDGE-R1
>   Policy Map WAN-EDGE-R1
> Class RTP
>   Strict Priority
>   Bandwidth 156 (kbps) Burst 3900 (Bytes)
> Class SIGNAL
>   Bandwidth 73 (kbps) Max Threshold 64 (packets)
> Class SCAVENGER
>   Bandwidth 8 (kbps) Max Threshold 64 (packets)
> Class class-default
>   Flow based Fair Queueing
>   Bandwidth 0 (kbps) Max Threshold 64 (packets)
> R1#
>
> I've never noticed the burst value before, and the SRND seems to be silent
> about what, if any, setting should be made to it.  It has a minimum value
> that cannot be set to zero.  The same setting is available if you set it to
> a percent bandwidth, but the output looks different (see below):
>
> R1#show run | b policy-map
> policy-map WAN-EDGE-R1
>  class RTP
>   priority percent 33
>
> R1#sh policy-map WAN-EDGE-R1
>   Policy Map WAN-EDGE-R1
> Class RTP
>   Strict Priority
>   Bandwidth 33 (%)
>
> Has anyone on OSL got ANY ideas or input on this?
>
> I over thinking this one or is it something to be concerned about?  This
> burst rate looks to be about 2.5% of the configured bandwidth, but I've not
> ever heard this mentioned before.
>
> Cliff

Re: [OSL | CCIE_Voice] cme features

[WorkerBee]

If DN 3004 and 3004 are not available or busy, the call will be routed
to DN 53004.


On Sat, Jan 24, 2009 at 6:05 AM, omar itani  wrote:
> ephone-hunt 1 peer
> pilot 3210
> list 3004, 3003
> timeout 12
> final 53004
>
> (final 53004) what does it mean?
>
>
> 
> What can you do with the new Windows Live? Find out

Re: [OSL | CCIE_Voice] CME DialPeer Match

[WorkerBee]

By default, outbound calls are analyzed digit-by-digit unlike inbound
calls which buffer all the digits.

To resolve your overlapping outbound calls,

You could use this

 dial-peer voice 2 voip
  destination-pattern 2[^4]..$ !- route outbound calls other than 24xx


There are many other possible combinations. This is one of the possible
resolution.



On Mon, Dec 29, 2008 at 10:16 AM, kamal yousaf  wrote:
> Hi,
>
>  If we have 2 dial-peers,
>
> One with:
>
> dial-peer voice 1 voip
>  destination-pattern 24004001
>  session target ipv4:10.10.10.10
>  incoming called-number 24004001
>  codec g711ulaw
>  no vad
>  dtmf-relay h245-alpha
>
> 2nd:
>
> dial-peer voice 2 voip
>  destination-pattern 2...
>  session target ipv4:10.10.10.10
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
>
> Both with default preference 0. CME is matching dial-peer 2 (2...) always
> when i dial '24004001'. Why doesn' t it match specific one i.e dial-peer 1
> instead of dial-peer 2.
>
> please shed some light.
>
> Thanks
>
>

[OSL | CCIE_Voice] voice vlan dot1p

[WorkerBee]

## Config A - Voice traffic in vlan 110 with CoS 5 ##

mls qos

interface f0/1
 mls qos trust cos
 switchport mode access
 switchport voice vlan 110
 switchport access vlan 10

## Config B - Voice traffic in native vlan = vlan 10 with CoS 5?? ##

mls qos

interface f0/1
 mls qos trust cos
 switchport voice vlan dot1p
 switchport access vlan 10


In CCO example, isn't the default Vlan = 1? What is the concept of Vlan 0?


This example shows how to configure a port connected to an IP phone to
use the CoS value to classify ingress traffic, to use 802.1p priority
tagging for voice traffic, and to use the default native VLAN (VLAN 0)
to carry all traffic:

Switch(config)# interface gigabitethernet0/1
Switch(config-if)# mls qos trust cos
Switch(config-if)# switchport voice vlan dot1p
Switch(config-if)# end

dot1p—Configure the Cisco IP Phone to use 802.1p priority tagging for
voice traffic and to use the default native VLAN (VLAN 0) to carry all
traffic. By default, the Cisco IP Phone forwards the voice traffic
with an 802.1p priority of 5.

[OSL | CCIE_Voice] NM-HDV2 with PVDM2??

[WorkerBee]

Does it comes with PVDM2 when I purchase this module?

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a008039c333.shtml#table2

Thanks.

Re: [OSL | CCIE_Voice] The only CCIE voice certified FROG on this planet - yeah I finally passed it

[WorkerBee]

Well done Frog, get ready to pound on another track and finish it off~

On Thu, Jul 24, 2008 at 11:14 PM, FrogOnDSCP46EF <[EMAIL PROTECTED]> wrote:
>
> The only frog on this planet with CCIE#
>
> Not joking, frog went 4th time to the CCIE lab in Sydney and this time asked
> to the proctor;- hey, I have been to this lab for 3 times and u never been
> kind enough to gave me the #. This time he got serious and thought a while,
> then he tabbed frog's head and put his hand in his pocket and gave frog a
> brand new  CCIE#. Frog hopped in pound again! who's going to use his CCIE#?
> no its damn wastage !
>
> Seriously if someone want to hire frog in Sydney , PM him.
>
>
>
> I finally got my shiny new number after studying hard for the past 9 months
> with my full time job.
>
> In Oct, 2005 I started studying for R&S lab but after a few month of
> preparation I though its not enough, end of the day its vendor cert.
> I decided to leave CCIE for a while and gain a Masters Degree in
> "Networking" first. I took an admission in Charles Stuart University Sydney
> and then finished it in a year & a month [jan 2006 to feb 2007].
>
> Then I had nothing to do so I thought to go back and do CCIE cert. I took a
> few months break from study and then thought seriously about which CCIE
> track I go for. I finally made a decision do the toughest exam first and
> then easier ones, so I hopped on CCIE voice horse.
>
> At the beginning I started studying 2-6 hours per week for CCIE voice
> written after june2007. I passed the voice written exam in august 2007.
> After that it was a real challenge for me to arrange routers switches,
> servers and softwares for the home lab. It took 2 months to arrange routers,
> servers for unity, call manager etc.
>
> I started labbing from nov, 2007 and binded up my study on 20th of July,2008
> [ in 9 months]. I have logged about 1700+ hours of labbing and studying. 80%
> labbing and 20% studying on the forums, workbook, cisco.com/unvercd etc.
>
> To be honest, I didn't sleep properly for the last 4 months. I used to get
> back to home form work at 6pm, start labbing until 3-4am. If I remember
> correctly I didn't sleep for more than 3-4 hours per day, after that get up
> early morning7am, get ready and go to work. I found it worked for me but the
> drawback of this is - in middle of the week I used to collapse completely
> due to not enough sleep.
>
> but end of the day, hard work pays!
>
> Thanks to everyone who've helped me.
>
> Special thanks to my boss, and my employer for supporting me - yes they paid
> for my 4 attempts and about $30K for my lab gears [lucky me].
>
> Oh yeah, proctor at Sydney lab is very helpful [that doesn't mean that he
> gives you the solutions]. I have read many threads on GS or other forums
> saying that proctor changes config during lunch break or something like
> that. Hey, the fact is that the proctor goes with you to have his lunch as
> well so who is there to change the configs? all hoax! Also, in Sydney lab
> proctor were very helpful in provide the DocCD docs [the link which were not
> working]. Thanks Scott for your help and I won't forgot those nice eggs
> sandwiches.
>
> Now I will take a few weeks break before I start R&S track.
>
> Here is the list of my study materials;
> 
>
> This forum [voiceie.com] was very helpful and was the main tool [along with
> cisco.com/univercd] for my study.
>
> IM study mates; this is a must tool for everyone, make your MSN/yahoo study
> partner. This helps when u run into trouble and your available resources
> gets short. U can instant ask the question to your study mates. I had 2-3
> good full time 24/7 study mates. without them I couldn't have done it.
>
> FYI, I never attended any bootcamps or institute for VoIP training. I am
> working in IT for the past 8 years [mainly in R&S, firwalls, iptel stuffs]
> so it was easy for me to nail all topics of voice exam.
>
> I have used a MOC lab from Robert Hockley [kiwi guy]
> http://www.ccievoice-assessor.com/
> that was helpful and I came across knowing many thing which I wouldn't have
> normally picked up by myself. its was about $450 bucks but after I did my 8
> hours full lab, we [robert and I] went through all scenarios, questions and
> discussed on the phone for more than 5 hours. WoW u can ask him anything
> virtually..
>
> Apart from above, I used ccbootcamp's volume#1 and technology workbook. the
> technology workbook was helpful when i was starting to study. so for new
> aspirant its good to use ccbootcamp's technology workbook. Thanks Brad and
> Avner for such a wonderful quick reference book.
>
> COD – Fasial khans and Mark snow/Vik's free online class. Take free online
> from these experts, its worth. Specially Vik's IPMA tricks [ a must see].
> How to do ipma using single partion. Thanks VIK for that, really helpful.
>
> I also used IEmentor's voice workbook as a reference and to simulate
> different scenario.
>
> Also b

Re: [OSL | CCIE_Voice] Translation Rule for ephone-dn

[WorkerBee]

Is working now.

For ephone-dn, apply the translation-profile as incoming call leg.

I got the order messed up previously.

Thanks.

HQ(config-ephone-dn)#translation-profile ?
  incoming  Translation Profile for incoming call leg
  outgoing  Translation Profile for outgoing call leg

HQ#debug translation detail
xrule detail tracing is enabled
HQ#
*Jun 19 00:23:01.015: xrule_checking
*Jun 19 00:23:01.015: xrule_checking calling 8001, called 028
<-- map HELP to 028



On Thu, Jun 19, 2008 at 6:53 AM, Rimon Vallavanatt Jr.
<[EMAIL PROTECTED]> wrote:
> Try incoming instead of outgoing on ephone
>
>  ephone-dn  1
>  number 8001
>  translation-profile incoming HELP
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee
> Sent: Wednesday, June 18, 2008 4:06 PM
> To: OSL CCIE Voice Lab Exam; [EMAIL PROTECTED]
> Subject: Re: [OSL | CCIE_Voice] Translation Rule for ephone-dn
>
> Numexp will work but I want to use the newer voice translation-profile
> method applied to ephone-dn instead.
>
> Under ephone-dn, it does support translation-profile but it doesn't
> seems to work.
>
> On Thu, Jun 19, 2008 at 5:02 AM, Derrick Shumake <[EMAIL PROTECTED]>
> wrote:
>> Try using numexp
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee
>> Sent: Wednesday, June 18, 2008 1:58 PM
>> To: OSL CCIE Voice Lab Exam
>> Subject: [OSL | CCIE_Voice] Translation Rule for ephone-dn
>>
>> I would like to perform digit manipulation at ephone-dn level.
>>
>> When a user using phone DN 8001 call HELP->4357, it will automatically
>> translate to 028.
>>
>> I tried on voice translation-profile, it does not work but it works if
> I
>> use the
>> older translate command.
>>
>> == not working ==
>>
>> voice translation-rule 2
>>  rule 1 /4357/ /028/
>>
>> voice translation-profile HELP
>>  translate called 2
>>
>> ephone-dn  1
>>  number 8001
>>  translation-profile outgoing HELP
>>
>> == working ==
>>
>> translation-rule 2
>>  Rule 0 4357 028
>>
>> ephone-dn  1
>>  number 8001
>>  translate called 2
>>
>>
>> Any idea? Thanks.
>>
>

Re: [OSL | CCIE_Voice] Translation Rule for ephone-dn

[WorkerBee]

Numexp will work but I want to use the newer voice translation-profile
method applied to ephone-dn instead.

Under ephone-dn, it does support translation-profile but it doesn't
seems to work.

On Thu, Jun 19, 2008 at 5:02 AM, Derrick Shumake <[EMAIL PROTECTED]> wrote:
> Try using numexp
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee
> Sent: Wednesday, June 18, 2008 1:58 PM
> To: OSL CCIE Voice Lab Exam
> Subject: [OSL | CCIE_Voice] Translation Rule for ephone-dn
>
> I would like to perform digit manipulation at ephone-dn level.
>
> When a user using phone DN 8001 call HELP->4357, it will automatically
> translate to 028.
>
> I tried on voice translation-profile, it does not work but it works if I
> use the
> older translate command.
>
> == not working ==
>
> voice translation-rule 2
>  rule 1 /4357/ /028/
>
> voice translation-profile HELP
>  translate called 2
>
> ephone-dn  1
>  number 8001
>  translation-profile outgoing HELP
>
> == working ==
>
> translation-rule 2
>  Rule 0 4357 028
>
> ephone-dn  1
>  number 8001
>  translate called 2
>
>
> Any idea? Thanks.
>

[OSL | CCIE_Voice] Translation Rule for ephone-dn

[WorkerBee]

I would like to perform digit manipulation at ephone-dn level.

When a user using phone DN 8001 call HELP->4357, it will automatically
translate to 028.

I tried on voice translation-profile, it does not work but it works if I use the
older translate command.

== not working ==

voice translation-rule 2
 rule 1 /4357/ /028/

voice translation-profile HELP
 translate called 2

ephone-dn  1
 number 8001
 translation-profile outgoing HELP

== working ==

translation-rule 2
 Rule 0 4357 028

ephone-dn  1
 number 8001
 translate called 2


Any idea? Thanks.

Re: [OSL | CCIE_Voice] IP Communicator and CME

[WorkerBee]

Do a "debug ephone register" and does it show the any error output?




On Tue, Jun 10, 2008 at 2:39 AM, Ahmed Hamed <[EMAIL PROTECTED]> wrote:
>
>
> Hi,
>
> I am trying to register IP communicator with CME but... for some reason IP 
> communicator keeps on registering, unregistering continuously...!
>
> All settings are in place:
>
> ephone type is CIPC
> ephone dn is configured
>
> any idea?
>
> AH
>
>
>
>
>

Re: [OSL | CCIE_Voice] Happy Birthday Wayne

[WorkerBee]

OOoooh..probably Wayne is giving out special discount to this
community on IPexpert voice products.

BIG 40 - Celebrate with 40% discount  :D

Dream on~

On Tue, Jun 3, 2008 at 3:05 AM, David L. Blair <[EMAIL PROTECTED]> wrote:
> I hear through the "grapevine" Wayne is getting a year older on 06-08-08.
> Could it be the BIG 40?
>
>
> David

Re: [OSL | CCIE_Voice] Voice Translation Rule

[WorkerBee]

Actually, I am trying to understand IPexpert PSTN config.

>From what I understand, there are 02 x T1 and 01 x E1 interfaces configured
as PSTN switch.

02 x T1, 01 x E1 - All has one common "translate called 1".

Each of the interfaces has a different - "translation-profile incoming 11x"

Hence, I think IPexpert uses "translated called 1" as a common translation rule
for all PSTN interfaces and a unique "translation-profile incoming"
per respective
interface for digit manipulation.



On Tue, May 20, 2008 at 4:03 AM, Gregory Jost (grjost) <[EMAIL PROTECTED]> 
wrote:
> The commands are different.  IPExpert recommends avoiding
> "translation-rule" (old method).  In your example, "translation-profile
> incoming 110" does nothing, as there isn't any "voice
> translation-profile 110" (e.g. this command doesn't apply to
> "translation-rule 110").
>
>
>
>
> For (new method):
> voice translation-rule
> voice translation-profile
>
> Use:
> translation-profile in/out
>
> ==
>
> For (old method):
> translation-rule
>
> Use:
> translate calling/called
>
>
>
>
> Greg Jost
> Network Consulting Engineer
> Unified Communications Practice
> Cisco Systems, Inc.
> 214-274-1922
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee
> Sent: Sunday, May 18, 2008 10:06 PM
> To: ccievoice1
> Cc: ccievoice
> Subject: Re: [OSL | CCIE_Voice] Voice Translation Rule
>
> So what is the purpose of attaching translation-profile directly under
> the voice port
> as compared to "translate command which also call upon the Translation
> rule".
>
> Both seems to perform the same function to me in this example.
>
> voice-port 0/2/0:23
>  translation-profile incoming 110   < Direct Profile
>  translate called 1
>
> On Mon, May 19, 2008 at 11:00 AM, ccievoice1 <[EMAIL PROTECTED]>
> wrote:
>> Translation profile allowed you to bind multiple translation-rule
> within.
>>
>> translation-profile CISCO
>>  translate called 1
>>  translate calling 2
>>
>> voice-port 0/2/0:23
>>  translation-profile incoming 110
>>
>> HTH
>>
>>
>> On 5/19/08, WorkerBee <[EMAIL PROTECTED]> wrote:
>>>
>>> What is the difference between the following?
>>> Can I say that 'translate' command only strictly applied to
> voice-port
>>> inbound only?
>>> Whereas translation-profile is more flexible (inbound/outbound)?
>>>
>>> In the example below, is there a priority of which command to invoke?
>>>
>>> translation-rule 110
>>> Rule 0 ^222 21
>>>
>>> translation-rule 1
>>> Rule 0 ^222 21
>>>
>>>
>>> voice-port 0/2/0:23
>>> translation-profile incoming 110
>>> translate called 1
>>>
>>>
>>> = CCO ==
>>>
>>> translation-profile (voice-port)
>>> - Use the translation-profile command to assign a predefined
>>> translation profile to a voice port.
>>>
>>> translate (voice-port)
>>> - To apply a translation rule to manipulate dialed digits on an
>>> inbound POTS call leg, use the translate command in voice-port
>>>   configuration mode.
>>
>>
>

Re: [OSL | CCIE_Voice] PSTN WAN - IPexpert Config

[WorkerBee]

I believe it is a "catch-all" like gateway of last resort in IP world to avoid
hitting the default dial-peer properties where it may not be desirable.



On Mon, May 19, 2008 at 12:46 PM, WorkerBee <[EMAIL PROTECTED]> wrote:
> There are 2 dial-peer posted by Vik on Re: Initial / Final
> Configurations #29921 - 09/19/07 05:00 PM
>
>
>
> dial-peer voice 1 pots
>  incoming called-number .
>  direct-inward-dial
>
>
> dial-peer voice 9876 voip
>  voice-class codec 1
>  incoming called-number .
>  no vad
>
> There are no physical voice port associate to dial-peer 1 and
> no session target of dial-peer 9876.
>
> What is the meaning of those dial-peer? How is it being used?
>
> Thanks.
>

[OSL | CCIE_Voice] PSTN WAN - IPexpert Config

[WorkerBee]

There are 2 dial-peer posted by Vik on Re: Initial / Final
Configurations #29921 - 09/19/07 05:00 PM



dial-peer voice 1 pots
 incoming called-number .
 direct-inward-dial


dial-peer voice 9876 voip
 voice-class codec 1
 incoming called-number .
 no vad

There are no physical voice port associate to dial-peer 1 and
no session target of dial-peer 9876.

What is the meaning of those dial-peer? How is it being used?

Thanks.

Re: [OSL | CCIE_Voice] Voice Translation Rule

[WorkerBee]

So what is the purpose of attaching translation-profile directly under
the voice port
as compared to "translate command which also call upon the Translation rule".

Both seems to perform the same function to me in this example.

voice-port 0/2/0:23
 translation-profile incoming 110   < Direct Profile
 translate called 1

On Mon, May 19, 2008 at 11:00 AM, ccievoice1 <[EMAIL PROTECTED]> wrote:
> Translation profile allowed you to bind multiple translation-rule within.
>
> translation-profile CISCO
>  translate called 1
>  translate calling 2
>
> voice-port 0/2/0:23
>  translation-profile incoming 110
>
> HTH
>
>
> On 5/19/08, WorkerBee <[EMAIL PROTECTED]> wrote:
>>
>> What is the difference between the following?
>> Can I say that 'translate' command only strictly applied to voice-port
>> inbound only?
>> Whereas translation-profile is more flexible (inbound/outbound)?
>>
>> In the example below, is there a priority of which command to invoke?
>>
>> translation-rule 110
>> Rule 0 ^222 21
>>
>> translation-rule 1
>> Rule 0 ^222 21
>>
>>
>> voice-port 0/2/0:23
>> translation-profile incoming 110
>> translate called 1
>>
>>
>> = CCO ==
>>
>> translation-profile (voice-port)
>> - Use the translation-profile command to assign a predefined
>> translation profile to a voice port.
>>
>> translate (voice-port)
>> - To apply a translation rule to manipulate dialed digits on an
>> inbound POTS call leg, use the translate command in voice-port
>>   configuration mode.
>
>

[OSL | CCIE_Voice] Voice Translation Rule

[WorkerBee]

What is the difference between the following?
Can I say that 'translate' command only strictly applied to voice-port
inbound only?
Whereas translation-profile is more flexible (inbound/outbound)?

In the example below, is there a priority of which command to invoke?

translation-rule 110
 Rule 0 ^222 21

translation-rule 1
 Rule 0 ^222 21


voice-port 0/2/0:23
 translation-profile incoming 110
 translate called 1


= CCO ==

translation-profile (voice-port)
 - Use the translation-profile command to assign a predefined
translation profile to a voice port.

translate (voice-port)
 - To apply a translation rule to manipulate dialed digits on an
inbound POTS call leg, use the translate command in voice-port
   configuration mode.

Re: [OSL | CCIE_Voice] PSTN connectivity to BR1 via PRI

[WorkerBee]

It changed to MULTIPLE_FRAME_ESTABLISHED after I remove MGCP from the
controller.

Thanks for Jon archive.
http://threebit.net/mail-archive/cisco-voip/msg07554.html



On Sun, May 11, 2008 at 9:26 AM, WorkerBee <[EMAIL PROTECTED]> wrote:
> My PRI ISDN Layer 2 shows TEI_ASSIGNED instead of MULITPLE_FRAME_ESTABLISHED.
>
> I have attached PSTN (poor man budge instead of 2811) and BR1 config.
>
> Anyone has any idea what is the issue here? The CCM cannot registered BR1 as
> MGCP gateway.
>
>
> BR1#show isdn status
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
> may not apply
>
> ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
>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 = 1
>Total Allocated ISDN CCBs = 0
>
>
>
>
> PSTN Switch 1760-V
> ==
>
> The reason I am using TDM clocking is because 1760V cannot support > 2
> PRi-Group.
>
> tdm clock E1 0/0 voice import onboard internal   <-- this is using E1
> R2 semi for workaround fix.
> tdm clock T1 1/0 both import onboard internal
> tdm clock T1 1/1 both import onboard internal
>
>
> controller T1 1/1
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-3,24
>  description "** To BR1 **"
>
> interface Serial1/1:23
>  description "** BR1 **"
>  no ip address
>  encapsulation hdlc
>  no logging event link-status
>  isdn switch-type primary-ni
>  isdn protocol-emulate network
>  isdn incoming-voice voice
>  no cdp enable
>
>
> BR1
> ===
>
> controller T1 0/2/0
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-3,24 service mgcp
>  description "** To PSTN **"
>
>
> interface Serial0/2/0:23
>  description "** To PSTN **"
>  no ip address
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
>

Re: [OSL | CCIE_Voice] PSTN connectivity to BR1 via PRI

[WorkerBee]

Further troubleshooting reveals, my pstn Layer 2 ISDN status is flipping between

 State = AWAITING_ESTABLISHMENT and TEI_ASSIGNED.


On Sun, May 11, 2008 at 9:26 AM, WorkerBee <[EMAIL PROTECTED]> wrote:
> My PRI ISDN Layer 2 shows TEI_ASSIGNED instead of MULITPLE_FRAME_ESTABLISHED.
>
> I have attached PSTN (poor man budge instead of 2811) and BR1 config.
>
> Anyone has any idea what is the issue here? The CCM cannot registered BR1 as
> MGCP gateway.
>
>
> BR1#show isdn status
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output
> may not apply
>
> ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
>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 = 1
>Total Allocated ISDN CCBs = 0
>
>
>
>
> PSTN Switch 1760-V
> ==
>
> The reason I am using TDM clocking is because 1760V cannot support > 2
> PRi-Group.
>
> tdm clock E1 0/0 voice import onboard internal   <-- this is using E1
> R2 semi for workaround fix.
> tdm clock T1 1/0 both import onboard internal
> tdm clock T1 1/1 both import onboard internal
>
>
> controller T1 1/1
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-3,24
>  description "** To BR1 **"
>
> interface Serial1/1:23
>  description "** BR1 **"
>  no ip address
>  encapsulation hdlc
>  no logging event link-status
>  isdn switch-type primary-ni
>  isdn protocol-emulate network
>  isdn incoming-voice voice
>  no cdp enable
>
>
> BR1
> ===
>
> controller T1 0/2/0
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-3,24 service mgcp
>  description "** To PSTN **"
>
>
> interface Serial0/2/0:23
>  description "** To PSTN **"
>  no ip address
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
>

Re: [OSL | CCIE_Voice] PSTN connectivity to BR1 via PRI

[WorkerBee]

I have done some basic troubleshooting, I notice Line Code Violations
at both ends.

The config is attached below.  My guess is the PSTN (1760) tdm
clocking is giving the problem?



BR1#show controllers T1
T1 0/2/0 is up.
  Applique type is Channelized T1
  Cablelength is long gain36 0db
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20060707, FPGA: 13, spm_count = 0
  Framing is ESF, Line Code is B8ZS, Clock Source is Line.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (585 seconds elapsed):
 0 Line Code Violations, 0 Path Code Violations
 4 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
 4 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
  Total Data (last 1 15 minute intervals):
 5745 Line Code Violations, 2 Path Code Violations,
 4 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
 4 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 205 Unavail Secs
BR1#


PSTN#show controllers T1 1/1
T1 1/1 is up.
  Applique type is Channelized T1
  Cablelength is long gain36 0db
  Description: "** To BR1 **"
  No alarms detected.
  alarm-trigger is not set
  Soaking time: 3, Clearance time: 10
  AIS State:Clear  LOS State:Clear  LOF State:Clear
  Version info Firmware: 20071009, FPGA: 20, spm_count = 0
  Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (549 seconds elapsed):
 0 Line Code Violations, 0 Path Code Violations
 4 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
 4 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
  Data in Interval 1:
 1 Line Code Violations, 3 Path Code Violations
 3 Slip Secs, 0 Fr Loss Secs, 1 Line Err Secs, 1 Degraded Mins
 5 Errored Secs, 1 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
  Total Data (last 1 15 minute intervals):
 1 Line Code Violations, 3 Path Code Violations,
 3 Slip Secs, 0 Fr Loss Secs, 1 Line Err Secs, 1 Degraded Mins,
 5 Errored Secs, 1 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
PSTN#


y PRI ISDN Layer 2 shows TEI_ASSIGNED instead of MULITPLE_FRAME_ESTABLISHED.

I have attached PSTN (poor man budge instead of 2811) and BR1 config.

Anyone has any idea what is the issue here? The CCM cannot registered BR1 as
MGCP gateway.


BR1#show isdn status
Global ISDN Switchtype = primary-ni

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

ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
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 = 1
Total Allocated ISDN CCBs = 0




PSTN Switch 1760-V
==

The reason I am using TDM clocking is because 1760V cannot support > 2
PRi-Group.

tdm clock E1 0/0 voice import onboard internal   <-- this is using E1
R2 semi for workaround fix.
tdm clock T1 1/0 both import onboard internal
tdm clock T1 1/1 both import onboard internal


controller T1 1/1
 framing esf
 linecode b8zs
 pri-group timeslots 1-3,24
 description "** To BR1 **"

interface Serial1/1:23
 description "** BR1 **"
 no ip address
 encapsulation hdlc
 no logging event link-status
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable


BR1
===

controller T1 0/2/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-3,24 service mgcp
 description "** To PSTN **"


interface Serial0/2/0:23
 description "** To PSTN **"
 no ip address
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable

[OSL | CCIE_Voice] PSTN connectivity to BR1 via PRI

[WorkerBee]

My PRI ISDN Layer 2 shows TEI_ASSIGNED instead of MULITPLE_FRAME_ESTABLISHED.

I have attached PSTN (poor man budge instead of 2811) and BR1 config.

Anyone has any idea what is the issue here? The CCM cannot registered BR1 as
MGCP gateway.


BR1#show isdn status
Global ISDN Switchtype = primary-ni

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

ISDN Serial0/2/0:23 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 = TEI_ASSIGNED
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 = 1
Total Allocated ISDN CCBs = 0




PSTN Switch 1760-V
==

The reason I am using TDM clocking is because 1760V cannot support > 2
PRi-Group.

tdm clock E1 0/0 voice import onboard internal   <-- this is using E1
R2 semi for workaround fix.
tdm clock T1 1/0 both import onboard internal
tdm clock T1 1/1 both import onboard internal


controller T1 1/1
 framing esf
 linecode b8zs
 pri-group timeslots 1-3,24
 description "** To BR1 **"

interface Serial1/1:23
 description "** BR1 **"
 no ip address
 encapsulation hdlc
 no logging event link-status
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable


BR1
===

controller T1 0/2/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-3,24 service mgcp
 description "** To PSTN **"


interface Serial0/2/0:23
 description "** To PSTN **"
 no ip address
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable

Re: [OSL | CCIE_Voice] Still Having Problems with IP Blue?

[WorkerBee]

I have not tried using VPN. But I think you should use the VPN adaptor
instead of
the loopback for sending out the traffic.

If it works, test it with Wireless/Ethernet (with VPN).

I am also curious to find out.



On Tue, May 6, 2008 at 4:25 PM, Ahmed Hamed <[EMAIL PROTECTED]> wrote:
> Hi HTH,
>
> Thanks for the reply. I already got the distinct profiles (with different
> mac addresses) configured as you said, but when choosing the "Network
> Interface" I chose the "Microsoft Loopback Interface #" instead of "PC
> Physical".
> I am using VPN over wireless connection. Should I use the "VPN Interface" or
> the "Wireless Interface"? Do you recommend using wired Ethernet connection
> instead (with VPN of course)?
>
>
> AH
>
>
>
> WorkerBee <[EMAIL PROTECTED]> wrote:
> In your IP Blue icon, set the properties as follows:
>
> "C:\Program Files\IP blue\VTGO\VTGO-PC.exe" /d
>
> /d - To run multiple instance
>
> Create multiple MS loopback interfaces. Manually change the mac address of
> each loopback to a unique address. By default, multiple loopback interfaces
> are created with the same mac address. This will create issue when
> registering
> to the CCM.
>
> Go the the IP Blue settings:
>
> - Station Mac Address : Select the unique loopback MAC (unique for
> each ip blue)
>
> - Network Interface : Select your PC physical.VMware Interface card
> (Can be the same)
>
> Remember to save each profile as different name. When you launch
> multiple instance,
> select a different profile for each instance. Hence you have unique
> mac address for each
> IP Blue instance.
>
> HTH
>
>
> On Tue, May 6, 2008 at 3:47 PM, Ahmed Hamed wrote:
> > Hi,
> >
> > I am still having problems running multiple instances of IP Blue!
> > For a while, the configuration of different profiles for HQ, BR1, and BR2
> > seemed to be stable, following the hints on
> >
> >
> http://www.certificationtalk.com:81/showflat.php?Cat//Board/VGen/Number/29126/page/10/view/collapsed/sb/5/o//fpart/1
> >
> >
> > Yesterday, after launching IP communicator, I started getting weird
> > symptoms: IP blue won't launch until I disable a couple of the Microsoft
> > loopback interfaces!
> > This is limiting me to three instances ( I am not sure if it is going to
> be
> > less in the next session!).
> >
> > If I am going to use both IP Blue and Cisco IP Communicator, what mac/
> > interface should I use on Cisco IP communicator to avoid conflicts?
> >
> > What is the best practice when having both softphones? Which to use
> Where??
> >
> > Any ideas for alternative soft phones?
> >
> > I have a Vista business laptop and am using IP blue vtgo lite v.2.10.1 for
> > xp and 2000.. Are there any compatibility issues?
> >
> > Appreciate your advice.
> >
> > AH
> >
> >
> >
>
>
>
>
>  
> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it
> now.

Re: [OSL | CCIE_Voice] Still Having Problems with IP Blue?

[WorkerBee]

In your IP Blue icon, set the properties as follows:

"C:\Program Files\IP blue\VTGO\VTGO-PC.exe" /d

/d - To run multiple instance

Create multiple MS loopback interfaces. Manually change the mac address of
each loopback to a unique address. By default, multiple loopback interfaces
are created with the same mac address. This will create issue when registering
to the CCM.

Go the the IP Blue settings:

- Station Mac Address : Select the unique loopback MAC  (unique for
each ip blue)

- Network Interface : Select your PC physical.VMware Interface card
(Can be the same)

Remember to save each profile as different name. When you launch
multiple instance,
select a different profile for each instance. Hence you have unique
mac address for each
IP Blue instance.

HTH


On Tue, May 6, 2008 at 3:47 PM, Ahmed Hamed <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am still having problems running multiple instances of IP Blue!
> For a while, the configuration of different profiles for HQ, BR1, and BR2
> seemed to be stable, following the hints on
>
> http://www.certificationtalk.com:81/showflat.php?Cat//Board/VGen/Number/29126/page/10/view/collapsed/sb/5/o//fpart/1
>
>
> Yesterday, after launching IP communicator, I started getting weird
> symptoms: IP blue won't launch until I disable a couple of the Microsoft
> loopback interfaces!
> This is limiting me to three instances ( I am not sure if it is going to be
> less in the next session!).
>
> If I am going to use both IP Blue and Cisco IP Communicator, what mac/
> interface should I use on Cisco IP communicator to avoid conflicts?
>
> What is the best practice when having both softphones? Which to use Where??
>
> Any ideas for alternative soft phones?
>
> I have a Vista business laptop and am using IP blue vtgo lite v.2.10.1 for
> xp and 2000.. Are there any compatibility issues?
>
> Appreciate your advice.
>
> AH
>
>
>
>  
> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it
> now.

Re: [OSL | CCIE_Voice] Difference between COR and CCM CSS/PT

[WorkerBee]

Thanks for the crystal clear explanation. This much clearer than the
proctor guide or CCO.





On Tue, May 6, 2008 at 5:00 AM, Vik Malhi <[EMAIL PROTECTED]> wrote:
> A COR List is applied to the inbound and the outbound dial-peer. The router
>  makes a comparision of the two. If the COR list assigned to the incoming
>  dial-peer contains everything that is in the COR list applied to the
>  outgoing dial-peer then the call will not be denied. If the outgoing COR
>  list has a member that does not feature in the incoming COR list then the
>  call will be denied. If the COR list is missing on either the inbound or
>  outbound dial-peer then the call will not be denied.
>
>  So in summary- COR List has to be applied on BOTH call legs.
>
>  CSS, which is only used on CCM, is applied to the caller (inbound call leg
>  if you like) and the Route Pattern (outgoing call leg) is assigned a
>  Partion.
>
>  The statement used in the Proctor Guide is making a comparison of COR and
>  CSS.
>
>
>  Vik Malhi - CCIE #13890
>  Senior Technical Instructor - IPexpert, Inc.
>
>  Telephone: +1.810.326.1444
>  Fax: +1.810.454.0130
>  Mailto: [EMAIL PROTECTED]
>
>  Join our free online support and peer group communities:
>  http://www.IPexpert.com/communities
>
>  IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand
>  and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE
>  Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage
>  Lab Certifications.
>
>
>
>
>  -Original Message-
>  From: [EMAIL PROTECTED]
>  [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee
>  Sent: Monday, May 05, 2008 2:21 AM
>  To: CCIE Voice Online Study List
>  Subject: [OSL | CCIE_Voice] Difference between COR and CCM CSS/PT
>
>  In the IPexpert proctor guide, Task 3.2 states:
>
>  The difference between COR and CCM CSS/Partition:
>
>  COR (CME)
>  =
>  Apply CSS to the incoming AND outgoing leg of the call
>
>  using corlist incoming|outgoing ...
>
>
>  CCS (CCM)
>  =
>  Apply to the incoming call leg only
>
>
>  I thought CCS is applied to CCM for outgoing call only??
>
>

[OSL | CCIE_Voice] Difference between COR and CCM CSS/PT

[WorkerBee]

In the IPexpert proctor guide, Task 3.2 states:

The difference between COR and CCM CSS/Partition:

COR (CME)
=
Apply CSS to the incoming AND outgoing leg of the call

using corlist incoming|outgoing ...


CCS (CCM)
=
Apply to the incoming call leg only


I thought CCS is applied to CCM for outgoing call only??

Re: [OSL | CCIE_Voice] Inline Power HWIC-4E 2811

[WorkerBee]

I am running the latest image and yet it does not support this command...

BR1#show power inline
   ^
% Invalid input detected at '^' marker.

BR1#show version
Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9-M), Version
12.4(15)T5, RELEASE SOFTWARE (fc4)




On Sat, May 3, 2008 at 10:54 AM, WorkerBee <[EMAIL PROTECTED]> wrote:
> Despite plugging in the Inline Power Supply unit and attached the
>  power cable to the motherboard,
>  I still cannot power up my IP Phone. The port works for normal Data 
> connection.
>
>  Is there a command to enable PoE for HWIC?
>
>  NAME: "2811 chassis", DESCR: "2811 chassis"
>  PID: CISCO2811 , VID: V02 , SN:
>
>  NAME: "4 Port FE Switch on Slot 0 SubSlot 0", DESCR: "4 Port FE Switch"
>  PID: HWIC-4ESW , VID: VN/A, SN:
>
>  NAME: "WIC/VIC/HWIC 0 Power Daughter Card", DESCR: "4-Port HWIC-ESW
>  Power Daughter Card"
>  PID: ILPM-4, VID: VN/A, SN:
>

[OSL | CCIE_Voice] Inline Power HWIC-4E 2811

[WorkerBee]

Despite plugging in the Inline Power Supply unit and attached the
power cable to the motherboard,
I still cannot power up my IP Phone. The port works for normal Data connection.

Is there a command to enable PoE for HWIC?

NAME: "2811 chassis", DESCR: "2811 chassis"
PID: CISCO2811 , VID: V02 , SN:

NAME: "4 Port FE Switch on Slot 0 SubSlot 0", DESCR: "4 Port FE Switch"
PID: HWIC-4ESW , VID: VN/A, SN:

NAME: "WIC/VIC/HWIC 0 Power Daughter Card", DESCR: "4-Port HWIC-ESW
Power Daughter Card"
PID: ILPM-4, VID: VN/A, SN:

Re: [OSL | CCIE_Voice] 7940s or 7960s ?

[WorkerBee]

Go for 7960G..there must be a reason for something  ;)

On Sat, Apr 26, 2008 at 11:53 AM, Mike Brooks <[EMAIL PROTECTED]> wrote:
> IP Blue is good.  But I would rather use hard phones.  I am not sure which
> ones to get. 7940s are cheaper ;-)
>
>
> Mike Brooks
> CCIE# 16027 (R&S)
>
>
>
> On 4/25/08, WorkerBee <[EMAIL PROTECTED]> wrote:
> > You may want to use IP Blue to simulate 7960 instead of hard phones.
> >
> > On Sat, Apr 26, 2008 at 11:36 AM, Mike Brooks <[EMAIL PROTECTED]> wrote:
> > > Hi All,
> > >
> > > For the labs in the workbook would I run into any issues if I used 7940s
> > > rather than the 7960s for my home lab ?
> > >
> > > Thank you,
> > >
> > > Mike Brooks
> > > CCIE# 16027 (R&S)
> >
>
>

Re: [OSL | CCIE_Voice] Setting up PSTN switch on 1760V

[WorkerBee]

Hi Mike,

I have modified my pod to overcome setting up the PSTN/Voice backbone network.

1760V
  * VWIC2-1MFT-E1 - ds0-group r2-digital semi-compelled ani   (since
1760 cannot support > 2 pri-group)
  * VWIC-2MFT-T1 - ISDN PRI pri-group, isdn incoming-voice voice
  * CME

2621XM
  * FR switching
  * GK

I am wondering are you using 1760 as your branch router?

Infact, due to budget, I use 2621XM with NM-HDV-T1 as a substitute for
6608. (Poor man fix)

Thanks.

On Mon, Apr 21, 2008 at 11:38 PM, Michael Gross <[EMAIL PROTECTED]> wrote:
> GK will not work on a 1760.  I have two 1760 routers in my home lab.  You
> can use a Dynagen/Dynamips instance of say a 3725 as a GK.
>
>
> On Sun, Apr 20, 2008 at 9:13 AM, WorkerBee <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > Anyone out there has successfully configured PSTN switch using Cisco
> > 1760 (VWIC-2MFT-T1, VWIC2-1MFT-T1/E1)?
> >
> > I would like to convert IPexpert PSTN + WAN config and port PSTN and GK to
> 1760.
> >
> > I have another router running as FR function.
> >
> > Thanks.
> >
>
>

[OSL | CCIE_Voice] Channelized T1 vs PRI ISDN T1

[WorkerBee]

I can configure either Channelized T1 or PRI ISDN T1 for handling voice calls.

So what is the main difference between the two? What I can see is DS-0 has
a type field which can select a range of signaling protocol such as
e&m, r2-digital, etc.




controller T1 slot/port
 ds0-group  timeslots  type 

-> automatic create

 voice-port slot/port:ds0-group-no




controller T1
 pri-group timeslots 

interface slot/number:23
 isdn incoming voice-modem

[OSL | CCIE_Voice] Setting up PSTN switch on 1760V

[WorkerBee]

Hi,

Anyone out there has successfully configured PSTN switch using Cisco
1760 (VWIC-2MFT-T1, VWIC2-1MFT-T1/E1)?

I would like to convert IPexpert PSTN + WAN config and port PSTN and GK to 1760.

I have another router running as FR function.

Thanks.

[OSL | CCIE_Voice] IPexpert PSTN config

[WorkerBee]

What is the meaning of % in the rule? I can't find a translation rule
with symbol '%'?

translation-rule 1
Rule 0 ^1212.% 212

Anyone can explain the IPexpert PSTN config? I am confused with the additional
config such as telephony-service, gatekeeper, etc that form the PSTN switch.

Please contact me offline.

Thanks.

[OSL | CCIE_Voice] QoS Mapping on 3550

[WorkerBee]

Am I right with the following explanation:

Assuming a COS marked packet (dot1q) enters and leaves QoS enabled
3550 switchports.


Steps


1. Incoming dot1q traffic to 3550 ingress port is mapped into
"internal DSCP" using COS-TO-DSCP map
   if under the interface:

   [ingress port]

   mls qos map cos-dscp 0 8 16 26 32 46 48 56

interface f0/1
  mls qos trust cos

2. 3550 uses cos-dscp map to convert COS to "internal DSCP" value
marked in Step 1 for internal switching

3. When the traffic exit the physical egress port, DSCP-TO-COS map is
used to convert "internal DSCP" value
marked in Step 1 to COS value.

4. COS value determined in Step 3 send the packet to one of the four
egress queue based on the
COS value. (default : Queue 1 : 0,1 Queue 2: 2,3 Queue 3 : 4,5
Queue 4 : 6,7)

[egress port]

mls qos map dscp-cos 8 16 26 32 48 to 0

interface f0/10

 wrr-queue cos-map 1 0 1  !-- DSCP-COS-MAP for L2 queueing
 wrr-queue cos-map 2 2 3
 wrr-queue cos-map 3 6 7
 wrr-queue cos-map 4 5

 priority-queue out !- RTP packets with COS 5 send
out priority queue

Re: [OSL | CCIE_Voice] Regions and Locations

[WorkerBee]

To add-on,

CCM Location
==

Audio field : Audio only
Video field : Audio + Video   (<- this one create confusion)




On Thu, Mar 27, 2008 at 2:59 PM, WorkerBee <[EMAIL PROTECTED]> wrote:
> If I am not mistaken, setting bandwidth in Location field (include
>  both audio +  video).
>
>
>
>  On Thu, Mar 27, 2008 at 2:33 PM, ccievoice1 <[EMAIL PROTECTED]> wrote:
>  > Hi paul & bobs,
>  >
>  > You sure you can configure AUDIO bandwidth in Region? You only allowed to
>  > choose the AUDIO codec in Region and configure VIDEO Call bandwidth in
>  > Region. However, you can configure AUDIO bandwidth in Location.
>  >
>  > So, if you select audio codec g711 in Region and configure audio bandwidth
>  > to 80 in Location, then you are allowing only 1 x g711 call.
>  >
>  > HTH,
>  >
>  >
>  >
>  > On Thu, Mar 27, 2008 at 2:21 PM, Paul and Bobs <[EMAIL PROTECTED]>
>  > wrote:
>  >
>  > > Whats the difference between setting the audio bandwidth to 80 in the
>  > region and setting the location bandwidth to 80 for bandwidth management
>  > over the WAN
>  > >
>  >
>  >
>

Re: [OSL | CCIE_Voice] Regions and Locations

[WorkerBee]

If I am not mistaken, setting bandwidth in Location field (include
both audio +  video).

On Thu, Mar 27, 2008 at 2:33 PM, ccievoice1 <[EMAIL PROTECTED]> wrote:
> Hi paul & bobs,
>
> You sure you can configure AUDIO bandwidth in Region? You only allowed to
> choose the AUDIO codec in Region and configure VIDEO Call bandwidth in
> Region. However, you can configure AUDIO bandwidth in Location.
>
> So, if you select audio codec g711 in Region and configure audio bandwidth
> to 80 in Location, then you are allowing only 1 x g711 call.
>
> HTH,
>
>
>
> On Thu, Mar 27, 2008 at 2:21 PM, Paul and Bobs <[EMAIL PROTECTED]>
> wrote:
>
> > Whats the difference between setting the audio bandwidth to 80 in the
> region and setting the location bandwidth to 80 for bandwidth management
> over the WAN
> >
>
>

Re: [OSL | CCIE_Voice] cRTP Header size

[WorkerBee]

The extra 2 bytes if you include the checksum.

On Wed, Mar 19, 2008 at 9:57 PM, William George
<[EMAIL PROTECTED]> wrote:
> When using compressed RTP, we know that we can reduce IP/UDP/RTP from 40
>  down to "2 to 4" Bytes.   I'm wondering if someone can explain why some
>  packet headers compress to 2 bytes and others only compress to 4 bytes.
>  What makes up the extra two bytes in the larger header?
>
>  Thank you
>

Re: [OSL | CCIE_Voice] Question on H323 Connection

[WorkerBee]

>From design concept, non-gk-controlled ICT trunk requires full mesh if
you have multiple sites. For H323 GW if you are connecting to H323
network. Afaik, ICT is used between CCMs.

I think is not which way is better, is rather which design approach is
the question is looking for.


On Sat, Mar 15, 2008 at 8:34 AM, Devildoc <[EMAIL PROTECTED]> wrote:
>
>  Say... configuring for an H323 connection between 2 sites (CCM and an IOS
> router), when is it appropriate to register an IOS router as an H323 gateway
> in CM as apposed to an H323 Non-GK-Controlled ICT link between an IOS router
> and CCM?  Is there a difference in functionality?  I see it in the WB that
> when setting an IPIPGW on an IOS router, sometimes the WB say configure an
> ICT link from the CCM to the IOS router and sometimes it says to register
> the router as an H323 gateway.  So which way is preferred?  Which way is
> better?  What's the recommendation?  Thanks for any input.
>
>  JD
>
> 
> Need to know the score, the latest news, or you need your Hotmail(R)-get your
> "fix". Check it out.