Re: [OSL | CCIE_Voice] ntp

[Michael Luo]

Take a look at this link:
http://www.cisco.com/en/US/docs/ios/12_1/configfun/configuration/guide/fcd303.html#wp1001926

The two examples explain when you'll need "ntp master".

Hope this helps!
Michael

On Sun, Sep 18, 2011 at 10:14 AM, Ashraf Ayyash wrote:

> i have read this note but it didn't stated that the router can provide
> clock without master command , in here we have one ntp source in our network
> which is the HQ (from all the network entities ) and HQ is getting clock
> from external NTP ,
>
> Ray , can you please try what guys suggest here ?
>
> Ash
>
>
> On Sun, Sep 18, 2011 at 8:11 AM, Michael Luo  wrote:
>
>> We should not use "ntp master" on HQ router.  This command is only needed
>> if the router cannot reach any NTP source.  Since the HQ router can reach
>> the backbone (PSTN) router, it does not require "ntp master".
>>
>> If you look at the link, you'll notice this note:
>>
>> ===
>> Use this command with extreme caution. It is very easy to override valid
>> time sources using this command, especially if a low stratum number is
>> configured. Configuring multiple machines in the same network with the *ntp
>> master* command can cause instability in timekeeping if the machines do
>> not agree on the time.
>> ===
>>
>> Michael
>>
>>
>> On Sun, Sep 18, 2011 at 4:28 AM, Ashraf Ayyash wrote:
>>
>>> Hi all ,
>>>
>>>  if we took off the master command we will not be able to sync our
>>> internal network entities to the HQ router , please feel free to correct me
>>> if i am wrong ,
>>>
>>> the output you gave Ray for the first email and the config say that you
>>> have added the master command before you synced with the external ntp
>>> and so your internal router got the highest level ,
>>>
>>> to get rid of this you can configure the ntp server and then make sure
>>> that you are synced and then use the master command with lower *stratum
>>> level
>>> and this should do the trick for you ,
>>>
>>> below the link :
>>>
>>> http://www.cisco.com/en/US/docs/ios/12_1/configfun/configuration/guide/fcd303.html#wp1004877
>>>
>>> Ash
>>> *
>>>
>>> On Sat, Sep 17, 2011 at 11:35 PM, Ray  wrote:
>>>
>>>> i found the issue , i took out the ntp master and then shut down/ no
>>>> shut my f0/0 connecting to the UTC server, and the clock was syncing as
>>>> below.. good
>>>>
>>>>
>>>> hq#sh ntp ass
>>>>
>>>>   address ref clock   st   when   poll reach  delay  offset
>>>>   disp
>>>> *~157.26.1.100127.127.1.1 15 63 64   377  0.000 2901006
>>>>  3.641
>>>>  * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~
>>>> configured
>>>>
>>>> hq#sho run | s ntp
>>>> ntp source Loopback0
>>>> ntp server 157.26.1.100
>>>>
>>>> --- On *Sat, 9/17/11, Marko Milivojevic * wrote:
>>>>
>>>>
>>>> From: Marko Milivojevic 
>>>> Subject: Re: [OSL | CCIE_Voice] ntp
>>>> To: "Ray" 
>>>> Cc: "ccie voice" 
>>>> Date: Saturday, September 17, 2011, 11:44 PM
>>>>
>>>>
>>>> Why did you configure your router to be NTP master? When you did that,
>>>> on which stratum did your router operate? If it was less than 15, will
>>>> it sync to a server on stratum 15?
>>>>
>>>> Answer these questions and you'll know the answer to yours :-)
>>>>
>>>> --
>>>> Marko Milivojevic - CCIE #18427
>>>> Senior Technical Instructor - IPexpert
>>>>
>>>> FREE CCIE training: http://bit.ly/vLecture
>>>>
>>>> Mailto: mar...@ipexpert.com <http://mc/compose?to=mar...@ipexpert.com>
>>>> Telephone: +1.810.326.1444
>>>> Web: http://www.ipexpert.com/
>>>>
>>>> On Sat, Sep 17, 2011 at 18:14, Ray 
>>>> http://mc/compose?to=jonha...@yahoo.com>>
>>>> wrote:
>>>> > looking at the sho ntp ass below and the config below. I could not
>>>> make the
>>>> > Hq router syn its time from 1567.26.1.100. this question troubled me
>>>> when i
>>>> > took the exams. any idea!!!.. the UTC server at 157.26.1.100 was
>>>> set to
>>>> > str

Re: [OSL | CCIE_Voice] ntp

[Michael Luo]

We should not use "ntp master" on HQ router.  This command is only needed if
the router cannot reach any NTP source.  Since the HQ router can reach the
backbone (PSTN) router, it does not require "ntp master".

If you look at the link, you'll notice this note:

===
Use this command with extreme caution. It is very easy to override valid
time sources using this command, especially if a low stratum number is
configured. Configuring multiple machines in the same network with the *ntp
master* command can cause instability in timekeeping if the machines do not
agree on the time.
===

Michael

On Sun, Sep 18, 2011 at 4:28 AM, Ashraf Ayyash  wrote:

> Hi all ,
>
>  if we took off the master command we will not be able to sync our internal
> network entities to the HQ router , please feel free to correct me if i am
> wrong ,
>
> the output you gave Ray for the first email and the config say that you
> have added the master command before you synced with the external ntp
> and so your internal router got the highest level ,
>
> to get rid of this you can configure the ntp server and then make sure that
> you are synced and then use the master command with lower *stratum level
> and this should do the trick for you ,
>
> below the link :
>
> http://www.cisco.com/en/US/docs/ios/12_1/configfun/configuration/guide/fcd303.html#wp1004877
>
> Ash
> *
>
> On Sat, Sep 17, 2011 at 11:35 PM, Ray  wrote:
>
>> i found the issue , i took out the ntp master and then shut down/ no shut
>> my f0/0 connecting to the UTC server, and the clock was syncing as below..
>> good
>>
>>
>> hq#sh ntp ass
>>
>>   address ref clock   st   when   poll reach  delay  offset
>> disp
>> *~157.26.1.100127.127.1.1 15 63 64   377  0.000 2901006
>>  3.641
>>  * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~
>> configured
>>
>> hq#sho run | s ntp
>> ntp source Loopback0
>> ntp server 157.26.1.100
>>
>> --- On *Sat, 9/17/11, Marko Milivojevic * wrote:
>>
>>
>> From: Marko Milivojevic 
>> Subject: Re: [OSL | CCIE_Voice] ntp
>> To: "Ray" 
>> Cc: "ccie voice" 
>> Date: Saturday, September 17, 2011, 11:44 PM
>>
>>
>> Why did you configure your router to be NTP master? When you did that,
>> on which stratum did your router operate? If it was less than 15, will
>> it sync to a server on stratum 15?
>>
>> Answer these questions and you'll know the answer to yours :-)
>>
>> --
>> Marko Milivojevic - CCIE #18427
>> Senior Technical Instructor - IPexpert
>>
>> FREE CCIE training: http://bit.ly/vLecture
>>
>> Mailto: mar...@ipexpert.com 
>> Telephone: +1.810.326.1444
>> Web: http://www.ipexpert.com/
>>
>> On Sat, Sep 17, 2011 at 18:14, Ray 
>> http://mc/compose?to=jonha...@yahoo.com>>
>> wrote:
>> > looking at the sho ntp ass below and the config below. I could not make
>> the
>> > Hq router syn its time from 1567.26.1.100. this question troubled me
>> when i
>> > took the exams. any idea!!!.. the UTC server at 157.26.1.100 was set
>> to
>> > stratum 15 i think... so how can u make HQ syn time from the UTC
>> server... I
>> > was confused  here..
>> > ntp source Loopback0
>> > ntp master
>> > ntp server 157.26.1.100
>> >
>> > hq#sho ntp ass
>> >   address ref clock   st   when   poll reach  delay  offset
>> > disp
>> > *~127.127.1.1 .LOCL.   7 12 16   377  0.000   0.000
>> >  0.238
>> >  ~157.26.1.10078.85.76.76 16 34 64   376  0.000 2901016
>> >  2.591
>> >  * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~
>> configured
>> > ___
>> > For more information regarding industry leading CCIE Lab training,
>> please
>> > visit www.ipexpert.com
>> >
>> > Are you a CCNP or CCIE and looking for a job? Check out
>> > www.PlatinumPlacement.com
>> >
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] CUPS integration and CUPC time consuming

[Michael Luo]

You should be able to do it within 10 to 15 minutes.

Michael

On Sat, Jul 23, 2011 at 9:01 PM, Adil Shaikh  wrote:

> hi guys,
>
> i am finding it takes me 25 to 30 minutes to configure CUCM for CUPS
> (including 2 x CUPC and 1 x IPPM) and CUPS.
>
> just want to know, is it the time frame you guys are taking?
>
> thanks
> -adil
>
> --
>   .. . .
> _7___|___|_|_|adil.sha...@gmail.com
>
> . .
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] UCCX server install issue VMWARE

[Michael Luo]

It won't activate because the key is for OEM version.  When you try to
activate an OEM version Windows, it'll try to retrieve vendor info from BIOS
(VM BIOS).

You should just install a regular Windows (retail or volume license) and
activate it.  Then apply the registry hack to make it look like Cisco IPT
OS.  Google is your friend.

Michael

On Sun, Jul 17, 2011 at 5:48 PM, Michael Dietrich  wrote:

> Installing on ESXi
>
> I have installed base MS Server 2003 from Cisco IPT OS Disk
> Win SVr 2003 Teleco Svr App SW 3.0 1-4 CPU
> Product Key : QW438-46P6G-MHXTB-99XGG-XX7CG
>
> using as base OS for UCCX 7 install for CCIE-V home lab
>
> Problem is when I try to activate the server key (microsoft) it will not
> activate...I have internet access on the 2003 Server VM
> Anyone else run into this issue?
>
> thanks, Michael
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] CUC issue

[Michael Luo]

It looks like for user-b, the ANI does not match the mailbox extension.

To see what ANI was presented to CUC, you may use RTMT > Unity Connection >
Port Monitor.

Michael

On Sun, Jun 19, 2011 at 2:25 PM, John Anzar  wrote:

>  Hi Guys,
>
> i'm doing some testing on the unity connection. i have created the users at
> the CUCM and i have intergrate CUC with the CUCM. ports registered and
> everything is fine. But i have an strange issue. From user-a when i dial
> voicemail it will ask for the pin. But for the user-b its
> giving standard message
>
> "*Hello*: Cisco *Unity Connection Messaging System*. From *a touchtone
> phone you may dial* an extension at *any* time.
>
> But i have check the configuration of the both users. its 100% identical. I
> wonder what is this. Any idea ?
>
> Best Regards
> John
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] MVA hairpin @ BR1, call drop if call from HQ or PSTN

[Michael Luo]

>From my understanding, if you're doing hairpin on MVA, it would be easier to
troubleshoot with two different number.

1) DID number (e.g. 408-555-).  This number is for PSTN user to access
MVA.

Let say the significant digit was 4 for incoming MGCP GW, that'll make the
incoming called number .

2) In CUCM route pattern, we'll have a pattern  which point to RL/RG and
eventually point to a H.323 GW.  That's because the MVA VXML script needs to
be triggered on a H323 GW.

3) On H323 GW, you'll have an incoming dial-peer with "incoming
called-number ".  In the dial-peer you'll associate the MVA service
(e.g. "service MVA")

4) Once the incoming called-number was matched, H323 GW will activate the
MVA IVR, which is actually a VXML script on CUCM server.

5) The script will make call to MVA number (which you configured in service
parameters and media resource).  To avoid confusion, this should be a
different number than the DID, say .

In this scenario,  is for "internal use" only.  It was not exposed to
PSTN.  Nor it was dialable for internal phones.

Just my two cents.

Michael

On Wed, Jun 15, 2011 at 11:45 AM, Sam Park wrote:

> George;
>
> Thanks for post.  I've actually not read the SRND on MVA (haven't got to it
> yet)
>
> I think you are talking about SRND "best practice" not a "known good".
> I've been practicing IPX method for the Lab.  Same number in and out and
> back in, works everytime.  Sure, it's not "best practice" but if someone is
> familiar with IPX material, it is a "known good".  and anyway, I recommended
> that for his testing - not final config.
>
> Cheers
>
> Sam.
>
>
>
> On Wed, Jun 15, 2011 at 10:41 AM, George Goglidze wrote:
>
>> Hi Sam,
>>
>> The "known good" config is what Cisco suggests in their SRND, and they
>> suggest to have two different numbers:
>> Here's a snippet from SRND:
>>
>> *Note *When deploying Mobile Voice Access in hairpinning mode, Cisco
>> recommends configuring the Mobile Voice Access DID at the PSTN gateway and
>> the Mobile Voice Access Directory Number within Cisco Unified CM (under 
>> *Media
>> Resources* > *Mobile Voice Access*) as different numbers. A translation
>> pattern within Unified CM can then be used to translate the called number of
>> the Mobile Voice Access DID to the configured Mobile Voice Access directory
>> number. Because the Mobile Voice Access directory number configured within
>> Unified CM is visible to the administrator only, translation between the DID
>> and directory number will be invisible to the end user and there will be no
>> change in end-user dialing behavior. This is recommended in order to prevent
>> mobility call routing issues in multi-cluster environments. This
>> recommendation does not apply to Mobile Voice Access in non-hairpinning
>> mode.
>>
>>
>>
>> On Wed, Jun 15, 2011 at 3:02 PM, Sam Park wrote:
>>
>>> Adil;
>>>
>>> For testing, create a G711-region only device pool and put your H323 gw
>>> in that DP.  So all devices even PSTN should be able to make the call.
>>> If not than, there might be some other config error.
>>> Start from there to see where it's failing.  I got through testing
>>> yesterday but reset my rack.
>>>
>>> As suggested earlier make sure you don't user voice class codec in your
>>> Dialpeer.
>>>
>>> Also, your config is interesting, have to try it next time.
>>> But since a "known good" config (in this forum for Lab takers) is to make
>>> the Route Parttern number the same as the MVA number, you could try that,
>>> again just to see that it all works, and there isn't some other issue.
>>>
>>> Sam.
>>>
>>>
>>> On Wed, Jun 15, 2011 at 9:02 AM, Adil Shaikh wrote:
>>>
 Hi SinGei,

 I do not have translation pattern in CM. You call flow is aboslutely
 correct. Just for clarity. I am adding few things in your call flow as
 below:

 PSTN IN [4158881999]--> mgcp gw [4digit]--> route pattern[1999] -->
 h323 gw --> match incoming [1999] (service mva configure) dp {here service
 mva URL  talks to publisher and downloads the MVA number configured in CM -
 in my case it is 1998}-->match outgoing mva# dp [1998] --> UCM
 I hope this clarifies the setup.

 -adil

 On Wed, Jun 15, 2011 at 9:49 PM, ShinGei Yong 
 wrote:

> hi Adil,
>
> Apologize for the questions, because never try your setup before.
>
> How do you translate the mgcp incoming 1999 to 1998? Do you have any
> translation?
> If not, how are you going to match the DNIS#
>
> My understanding was:
>
> PSTN IN --> mgcp gw --> route pattern --> h323 gw --> match incoming
> mva# (service mva configure) dp-->match outgoing mva# dp --> UCM
>
> TIA
> Shingei
>
>
> On Wed, Jun 15, 2011 at 7:32 PM, Adil Shaikh wrote:
>
>> hi ShinGei,
>>
>> The MVA number is 1998.
>> Rather than using same number for the inbound to application service
>> mva and outbound to MVA number, two 

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

[Michael Luo]

MVA IVR support G.711 only.  If inter-region codec was set to G.729, you
need transcoder.

On Wed, Jun 15, 2011 at 1:21 AM, Adil Shaikh  wrote:

> Hi all,
>
> I have configured MVA hairpin on BR1 router which is an MGCP gateway.
> If the call is made to MVA route-pattern number [1999] from BR1 phone then
> i hear "Welcome to Cisco Unfiied Communications"
> If i call to MVA route-pattern number [1999] from HQ phone or PSTN phone
> then i do not hear "Welcome to Cisco Unfied Communications". debug shows the
> correct MVA dial-peer hits on BR1-GW.
>
> There is no transcoder configured on HQ or BR1 device pool.  To resolve my
> issue, if transcoder needs to be configured on HQ or BR1 device pool then i
> would like to know why.
>
>
> Here is the configuration:
> MVA directory number: 1998
> Route-pattern: 1999  > BR1-H323-GW in BR1 device pool
> BR1-MGCP-GW in BR1 devicepool
> BR1 and HQ region talks g729
>
> Configuration from BR1-RTR:
>
> interface Loopback0
>  ip address 10.10.110.2 255.255.255.255
>  ip ospf network point-to-point
>  h323-gateway voip bind srcaddr 10.10.110.2
>
>
> voice service voip
>  allow-connections h323 to h323
> !
> !
> !
> voice class codec 1
>  codec preference 1 g711ulaw bytes 160
>  codec preference 2 g729r8 bytes 20
> !
> !
> !
> voice class h323 1
>  h225 timeout tcp establish 3
> !
>
>
> voice translation-rule 1002
>  rule 1 /^1002$/ /4158884343/
> !
> !
> voice translation-profile 1002
>  translate calling 1002
> !
> !
> !
> application
>   service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
>   !
>
> dial-peer voice 1997 voip
>  destination-pattern 1998
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.11
>  dtmf-relay h245-alphanumeric
>  no vad
> !
> dial-peer voice 1998 voip
>  preference 1
>  destination-pattern 1998
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.10
>  dtmf-relay h245-alphanumeric
>  no vad
> !
> dial-peer voice 1999 voip
>  translation-profile incoming 1002
>  service mva
>  voice-class codec 1
>  incoming called-number 1999
>  dtmf-relay h245-alphanumeric
> !
> -
> output of debug voip dial-p
>
> Following output when called 1999 from BR1 extenstion 1002. I hear the MVA
> message:
> R1-RTR#
> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> BR1-RTR#
> *Jun 15 11:23:22.838: //-1/800CEF8C0B00/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
> BR1-RTR#
> BR1-RTR#
> BR1-RTR#
> BR1-RTR#
> BR1-RTR#
> Following output when called 1999 from HQ extenstion 5002. I do not hear
> the MVA message:
> BR1-RTR#
> BR1-RTR#
> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>Calling Number=5002, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>Calling Number=5002, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> BR1-RTR#
> *Jun 15 11:23:34.346: //-1/801A16940C00/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
> BR1-RTR#
> BR1-RTR#
> BR1-RTR#
>
> Following output when called 4158881999 from PSTN phone 4158884343. I do
> not hear the MVA message:
> BR1-RTR#
> *Jun 15 11:25:44.078: //-1/808792E10D00/DPM/dpAssociateIncomingPeerCore:
>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> *Jun 15 11:25:44.078: //-1/808792E10D00/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
> *Jun 15 11:25:44.078: //-1/808792E10D00/DPM/dpAssociateIncomingPeerCore:
>Calling Number=1002, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> *Jun 15 11:25:44.

[OSL | CCIE_Voice] re-read / re-grade

[Michael Luo]

Just ran across this post by accident:
https://cisco-support.hosted.jivesoftware.com/message/477914#477914


"Re-grade or re-read for CCIE Voice lab is actively being pursued as a
service to our candidates. It's more costly and administratively complex to
offer re-reads for Voice, but we see it as a necessity for candidates, and
therefore we are working hard to make it available soon.

 Assessor for voice, on the other hand, is also being worked on, although I
don't really have any solid dates to offer, hopefully we will have better
news for you towards the end of* this calendar year*."

Then I realize it was posted in 2006.
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[OSL | CCIE_Voice] Disable G.722?

[Michael Luo]

I heard the rumor that even if you specify G.711 as intra-region codec,
you'll have to disable G.722 in Enterprise Parameters and Service
Parameters.  Otherwise, G.722 will be chosen.

I tested it in the lab.  That was not the case.  If I explicitly specified
G.711 as intra-region codec, the phones will always use G.711.

Do we still need to disable G.722?

Thanks!
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[OSL | CCIE_Voice] "irrelevant" commands

[Michael Luo]

>From my observation, the biggest challenge in voice lab was - some of the
"irrelevant" commands actually can cost you points.

What I meant "irrelevant" was - without those commands, everything seems to
work properly.  But obviously, they were expected to show up in your config.

I was wonder if we can gather all those commands in this thread.  Here's
what I got so far:

1) Line code and Framing under t1/e1 controllers.
In home lab, t1/e1 controller works fine without those (default will work).
But you could lose points if you didn't type those commands (even if call
routing was working fine).

2) "isdn send-alerting" and "isdn sending-complete" under D-channel
interface.

3) Incoming VoIP dial-peer
Without it, incoming VoIP call will match the default (hidden) dial-peer.

4) "mwi relay" under "telephony-service" and "mwi sip" under ephone-dn (or
ephone-dn-template)

5) "transfer-pattern .T" under "telephony-service"

6) "mgcp dtmf-relay voip codec all mode out-of-band"

Please add more to the list.

Thanks!
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[OSL | CCIE_Voice] MGCP to H323 fallback

[Michael Luo]

Does anyone know what the purpose of "service alternate Default" command?

In my lab, I use "ccm-manager fallback-mgcp" only.  It seems to work well.
But Cisco recommend "service alternate Default".  Why was that?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] RSVP agent on LAN?

[Michael Luo]

The network topology is as below:
IP Phone --> LAN --> Router --> WAN --> Router --> LAN --> IP Phone.

I read the CUCM 7 SRND.  On page 127 of 1102, there's a note:
---
When RSVP is enabled on a router interface, all other interfaces in the
router will drop RSVP messages unless they are also enabled for RSVP. To
avoid dropping RSVP messages, enable RSVP on all interfaces through which
you expect RSVP signaling to transit. If call admission control is not
desired on an interface, set the bandwidth value to 75% of the interface
bandwidth.
---

1) If I enabled rsvp on LAN interface only (not on WAN interface), CUCM
always route the call via AAR as if there's not enough bandwidth.

2) If I enable rsvp on both LAN and WAN interfaces, I saw that rsvp only
reserved bandwidth on WAN interface based on the output of "show ip rsvp
interface".

Can anyone explain why it didn't reserve bandwidth on LAN interface?  If it
didn't reserve bandwidth on LAN interface, what's the point of enabling rsvp
there?

Thanks!
Michael

On Thu, Jun 9, 2011 at 1:58 PM, Randall Saborio  wrote:

> As I understand, this should work. The requirement is that devices must be
> on separate locations and that each device must have a separate RSVP agent
> on its own.
> Then make sure "ip rsvp bandwidth" is configured on the interfaces between
> the RSVP agents.
>
> If CUCM tries to use same RSVP agent for both devices, reservation fails.
>
> For what you ask (which is not typical), then you must have probably two
> ISRs with RSVP agent on same LAN.
>
>
>  On Thu, Jun 9, 2011 at 12:56 PM, Michael Luo  wrote:
>
>>  When we do RSVP CAC, we usually put the RSVP agent on WAN interface.
>>
>> What if we were asked to put the RSVP agent on LAN interface or voice
>> VLAN?  I was not able to get it worked.
>>
>> Any hints?
>>
>> Thanks!
>>  Michael
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
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>
>
>
> --
> Randall "da ill" Saborio
> CCIE Voice Wannabe #10054675811
>
>
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[OSL | CCIE_Voice] RSVP agent on LAN?

[Michael Luo]

When we do RSVP CAC, we usually put the RSVP agent on WAN interface.

What if we were asked to put the RSVP agent on LAN interface or voice VLAN?
I was not able to get it worked.

Any hints?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] I just got my Fail result

[Michael Luo]

Customized prompt was required.  Thus I recorded my prompt.

Set agent to not ready as required.

For CSQ, there's no specific requirement.  Thus I created my own CSQ.

I'm pretty familiar with UCCX.  I can do whatever they ask me to do.  I just
need to understand their grading philosophy so I don't miss the trivial
things and lose the whole portion.

Michael

On Wed, Jun 8, 2011 at 8:54 AM, Ki Wi  wrote:

> Maybe you missed out some requirement?
>
> Such as they want you to use system recording but you recorded your own?
>
> They want you to set agent to not ready after the call ended? Set the agent
> to work state after the call ended?
>
> Queue the call to special CSQ, etc etc.
>
> UCCX is a nightmare.!
>
> On Wed, Jun 8, 2011 at 10:37 PM, Michael Luo  wrote:
>
>> I meant - "I've done everything right in UCCX".
>>
>>
>> On Wed, Jun 8, 2011 at 7:37 AM, Michael Luo  wrote:
>>
>>> I also got my report.  That's very confusing to me as well.
>>>
>>> For example, I'm pretty sure I made a mistake on QoS.  But I got 100%.
>>>
>>> Then I'm pretty sure I've don't everything right in UCCX but got 38%.
>>>
>>>
>>> On Wed, Jun 8, 2011 at 4:02 AM, adam compton  wrote:
>>>
>>>> I know what you mean.  I thought I nailed QOS 100% and got a zero.
>>>>
>>>> On Wed, Jun 8, 2011 at 3:04 AM, Chris Green wrote:
>>>>
>>>>>  Hi All,
>>>>>
>>>>> I just got my fail result, and what it shocked me is the Call Routing
>>>>> mark for 40% while I am sure I done what was asked for and I tested as 
>>>>> well.
>>>>>
>>>>> I also got 100% in 4 different section but what it killed me is Call
>>>>> Routing which I was quite confident about that section :(
>>>>>
>>>>> Chris
>>>>>
>>>>> ___
>>>>> For more information regarding industry leading CCIE Lab training,
>>>>> please visit www.ipexpert.com
>>>>>
>>>>> Are you a CCNP or CCIE and looking for a job? Check out
>>>>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>>>>
>>>>
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
>>>> please visit www.ipexpert.com
>>>>
>>>> Are you a CCNP or CCIE and looking for a job? Check out
>>>> www.PlatinumPlacement.com
>>>>
>>>
>>>
>>
>> ___
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>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] I just got my Fail result

[Michael Luo]

I meant - "I've done everything right in UCCX".

On Wed, Jun 8, 2011 at 7:37 AM, Michael Luo  wrote:

> I also got my report.  That's very confusing to me as well.
>
> For example, I'm pretty sure I made a mistake on QoS.  But I got 100%.
>
> Then I'm pretty sure I've don't everything right in UCCX but got 38%.
>
>
> On Wed, Jun 8, 2011 at 4:02 AM, adam compton  wrote:
>
>> I know what you mean.  I thought I nailed QOS 100% and got a zero.
>>
>> On Wed, Jun 8, 2011 at 3:04 AM, Chris Green  wrote:
>>
>>>  Hi All,
>>>
>>> I just got my fail result, and what it shocked me is the Call Routing
>>> mark for 40% while I am sure I done what was asked for and I tested as well.
>>>
>>> I also got 100% in 4 different section but what it killed me is Call
>>> Routing which I was quite confident about that section :(
>>>
>>> Chris
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] I just got my Fail result

[Michael Luo]

I also got my report.  That's very confusing to me as well.

For example, I'm pretty sure I made a mistake on QoS.  But I got 100%.

Then I'm pretty sure I've don't everything right in UCCX but got 38%.

On Wed, Jun 8, 2011 at 4:02 AM, adam compton  wrote:

> I know what you mean.  I thought I nailed QOS 100% and got a zero.
>
> On Wed, Jun 8, 2011 at 3:04 AM, Chris Green  wrote:
>
>>  Hi All,
>>
>> I just got my fail result, and what it shocked me is the Call Routing mark
>> for 40% while I am sure I done what was asked for and I tested as well.
>>
>> I also got 100% in 4 different section but what it killed me is Call
>> Routing which I was quite confident about that section :(
>>
>> Chris
>>
>> ___
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>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] UCCX grading

[Michael Luo]

The one I got actually has three tasks - two for scripts and one for agent
clients.

But I tested the call flow by calling the route points and watch the agents
behavior.  It was all functioning as required.  So I don't understand why
lost points.

On Wed, Jun 8, 2011 at 12:28 AM, Victor Malyuga wrote:

> I wonder how we can get 38% if it is only 1 task. It can be 0 or 100. There
> is no partial mark - we were told!
>
> --- On *Wed, 8/6/11, Michael Luo * wrote:
>
>
> From: Michael Luo 
> Subject: [OSL | CCIE_Voice] UCCX grading
> To: ccie_voice@onlinestudylist.com
> Date: Wednesday, 8 June, 2011, 5:40
>
>
> I got 38% on UCCX, which I didn't expect.
>
> I tested made test calls to verify the workflow.  What could be the
> possible traps / tricks that cause loss of points?
>
> Thanks!
> Michael
>
> -Inline Attachment Follows-
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
>
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[OSL | CCIE_Voice] UCCX grading

[Michael Luo]

I got 38% on UCCX, which I didn't expect.

I tested made test calls to verify the workflow.  What could be the possible
traps / tricks that cause loss of points?

Thanks!
Michael
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[OSL | CCIE_Voice] Auto QoS - trust or not?

[Michael Luo]

When configure auto QoS for WAN (on router), shall we use "trust" keyword or
not?  What's the best practice?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] "reserved threshold" and }threshold 3"

[Michael Luo]

Well, from my point of view, I can see it, I can set it with the command
below:

SW(config)#mls qos queue-set output 1 threshold 2 60 80 ?
  <1-100>  enter *reserved threshold* 1-100

That confused me.
Michael

On Sun, Jun 5, 2011 at 2:05 PM, Ki Wi  wrote:

> Means it's default, you can't see it and can't set it. If do you, you will
> see a value call threshold 3.
>
> That's why you always see threshold 1 and threshold 2 around in auto qos.
> =)
>
>
> On Mon, Jun 6, 2011 at 1:55 AM, Michael Luo  wrote:
>
>> Also per
>> http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1163863
>> ,
>>
>> "The drop-threshold percentage for threshold 3 is predefined. *It is set
>> to the queue-full state. *"
>>
>> If it is set to the queue-full state, shouldn't it be 100 instead of 50?
>>
>> Thanks!
>> Michael
>>
>>
>> On Sun, Jun 5, 2011 at 12:51 PM, Michael Luo  wrote:
>>
>>> Can somebody help me understand the statement on
>>> http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1284809
>>> :
>>>
>>> "Each queue has three threshold values. The QOS label is determines which
>>> of the three threshold values is subjected to the frame. Of the three
>>> thresholds, *two are configurable (explicit) and one is not (implicit)*
>>> ."
>>>
>>> I'm curious about the "implicit" threshold that was NOT configurable.
>>>
>>> When I used the command "show mls qos queue-set", I got the following.
>>>
>>> Queueset: 1
>>> Queue :   1   2   3   4
>>> --
>>> buffers   :  25  25  25  25
>>> threshold1: 100 200 100 100
>>> threshold2: 100 200 100 100
>>> * reserved  :  50  50  50  50*
>>> maximum   : 400 400 400 400
>>>
>>> Was the line "reserved" corresponding to the "implicit" threshold?  If
>>> yes, I think it's configurable with the following command:
>>>
>>> SW(config)#mls qos queue-set output 1 threshold 2 60 80 ?
>>>   <1-100>  enter *reserved threshold* 1-100
>>>
>>> Did I misunderstand it?
>>>
>>> Thanks!
>>> Michael
>>>
>>
>>
>> ___
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>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] "reserved threshold" and }threshold 3"

[Michael Luo]

That's why I got confused.  If it's not configurable, whey the command
option was there (to set threshold 3).

On Sun, Jun 5, 2011 at 1:31 PM, Miron Kobelski  wrote:

> This is what I was writing in my previous email. Reserved is something
> different. It sepcifies the percentage of the shared buffer, which is
> reserved for this specific queue. Please correct me if I'm wrong.
>
> best regards
> kobel
>
> On Sun, Jun 5, 2011 at 19:55, Michael Luo  wrote:
>
>> Also per
>> http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1163863
>> ,
>>
>> "The drop-threshold percentage for threshold 3 is predefined. *It is set
>> to the queue-full state. *"
>>
>> If it is set to the queue-full state, shouldn't it be 100 instead of 50?
>>
>> Thanks!
>> Michael
>>
>>
>> On Sun, Jun 5, 2011 at 12:51 PM, Michael Luo  wrote:
>>
>>> Can somebody help me understand the statement on
>>> http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1284809
>>> :
>>>
>>> "Each queue has three threshold values. The QOS label is determines which
>>> of the three threshold values is subjected to the frame. Of the three
>>> thresholds, *two are configurable (explicit) and one is not (implicit)*
>>> ."
>>>
>>> I'm curious about the "implicit" threshold that was NOT configurable.
>>>
>>> When I used the command "show mls qos queue-set", I got the following.
>>>
>>> Queueset: 1
>>> Queue :   1   2   3   4
>>> --
>>> buffers   :  25  25  25  25
>>> threshold1: 100 200 100 100
>>> threshold2: 100 200 100 100
>>> * reserved  :  50  50  50  50*
>>> maximum   : 400 400 400 400
>>>
>>> Was the line "reserved" corresponding to the "implicit" threshold?  If
>>> yes, I think it's configurable with the following command:
>>>
>>> SW(config)#mls qos queue-set output 1 threshold 2 60 80 ?
>>>   <1-100>  enter *reserved threshold* 1-100
>>>
>>> Did I misunderstand it?
>>>
>>> Thanks!
>>> Michael
>>>
>>
>>
>> ___
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>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] "reserved threshold" and }threshold 3"

[Michael Luo]

Also per
http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1163863
,

"The drop-threshold percentage for threshold 3 is predefined. *It is set to
the queue-full state. *"

If it is set to the queue-full state, shouldn't it be 100 instead of 50?

Thanks!
Michael

On Sun, Jun 5, 2011 at 12:51 PM, Michael Luo  wrote:

> Can somebody help me understand the statement on
> http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1284809
> :
>
> "Each queue has three threshold values. The QOS label is determines which
> of the three threshold values is subjected to the frame. Of the three
> thresholds, *two are configurable (explicit) and one is not (implicit)*."
>
> I'm curious about the "implicit" threshold that was NOT configurable.
>
> When I used the command "show mls qos queue-set", I got the following.
>
> Queueset: 1
> Queue :   1   2   3   4
> --
> buffers   :  25  25  25  25
> threshold1: 100 200 100 100
> threshold2: 100 200 100 100
> * reserved  :  50  50  50  50*
> maximum   : 400 400 400 400
>
> Was the line "reserved" corresponding to the "implicit" threshold?  If yes,
> I think it's configurable with the following command:
>
> SW(config)#mls qos queue-set output 1 threshold 2 60 80 ?
>   <1-100>  enter *reserved threshold* 1-100
>
> Did I misunderstand it?
>
> Thanks!
> Michael
>
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[OSL | CCIE_Voice] "reserved threshold" and }threshold 3"

[Michael Luo]

Can somebody help me understand the statement on
http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_46_se/configuration/guide/swqos.html#wp1284809
:

"Each queue has three threshold values. The QOS label is determines which of
the three threshold values is subjected to the frame. Of the three
thresholds, *two are configurable (explicit) and one is not (implicit)*."

I'm curious about the "implicit" threshold that was NOT configurable.

When I used the command "show mls qos queue-set", I got the following.

Queueset: 1
Queue :   1   2   3   4
--
buffers   :  25  25  25  25
threshold1: 100 200 100 100
threshold2: 100 200 100 100
* reserved  :  50  50  50  50*
maximum   : 400 400 400 400

Was the line "reserved" corresponding to the "implicit" threshold?  If yes,
I think it's configurable with the following command:

SW(config)#mls qos queue-set output 1 threshold 2 60 80 ?
  <1-100>  enter *reserved threshold* 1-100

Did I misunderstand it?

Thanks!
Michael
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[OSL | CCIE_Voice] Frame Relay Traffic Shaping

[Michael Luo]

I was trying to understand if the two configuration below achieve the same
thing.  If not, what's the difference?

Configuration A:

map-class frame-relay FRTS
 frame-relay cir 768000
 frame-relay bc 7680
 frame-relay be 0
 frame-relay mincir 768000

Configuration B:

map-class frame-relay FRTS
 service-policy out policy_shape
!
policy-map policy_shape
 class class-default
shape average 768000 7680 0


Thanks!
Michael
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[OSL | CCIE_Voice] transcoder for voicemail ports

[Michael Luo]

I'm testing with Unity Connection 7.0.

Let say if the voicemail ports are in region A.  IP phones and voice gateway
are in region B.  Cross-region codec was G.729.

IP phones can call VM pilot just fine (with G.729).

However, if PSTN calls IP phone and rollover (CFNA/CFB) to voicemail, I got
fast busy (reorder tone).  By looking at CCM trace, it seems to be codec
issue:

06/02/2011 23:07:43.042 CCM|MediaResourceManager::sendAllocationResourceErr
- ERROR - no transcoder device configured.

My question is: if I configure transcoder to fix this problem, shall the
transcoder be in the voicemail's mgrl or the voice gateway's mrgl?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] CUBE H.323 to H.323

[Michael Luo]

Hi George,

You were right.

On the destination GW, the incoming dial-peer was hard coded to G.711.  Thus
causing the codec mismatch.

Thank you very much for the tips!

Michael

On Mon, May 23, 2011 at 9:51 PM, George Goglidze  wrote:

> Hi Michael,
>
> What is the destination phone? I guess it's CUCME?  but is it SCCP Phone or
> SIP?
>
> You've showed the dial-peers on CUBE, but what about dialpeers on the
> destination CUCME?
> What dial-peer is it matching? and what codec does that dial-peer have?
>
> Do the following on the CUCME:
> debug voice dialpeer inout
> check which dial-peer is incoming, and then check the codec on that
> dial-peer.
> Then if the endpoint is SCCP it should work, but if it's SIP then its
> different. You would have to check the codec under voice register pool.
>
> Regards,
>
> On Mon, May 23, 2011 at 7:03 PM, Michael Luo  wrote:
>
>> I was testing CUBE feature (H.323 to H.323).  The call can ring the
>> destination.  But when called party pick up, the call got disconnected with
>> fast busy.
>>
>> debug cch323 h225 shows "//31/8087591D0900/H323/cch323_h225_send_release:
>> Cause = 65".
>>
>> Per the symptom and debug, it seems to be a codec issue.  But I've checked
>> the following:
>>
>> 1) "allow h323 to h323"
>> 2) incoming dial-peer was g.729r8
>> 3) outgoing dial-peer was g.729r8
>> 4) "Wait for Far End H.245 Terminal Capability Set" was unchecked
>>
>> What else shall I check?
>>
>> Thanks!
>> Michael
>>
>> ___
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Re: [OSL | CCIE_Voice] CUBE H.323 to H.323

[Michael Luo]

On the destination it was SCCP.

I lab this before.  It used to work.  Then I wipe out the config and redo
it.  This time it didn't work.

The CUBE kept sending "Bearer capability not implemented" to CUCM.

Thanks!
Michael

On Mon, May 23, 2011 at 9:51 PM, George Goglidze  wrote:

> Hi Michael,
>
> What is the destination phone? I guess it's CUCME?  but is it SCCP Phone or
> SIP?
>
> You've showed the dial-peers on CUBE, but what about dialpeers on the
> destination CUCME?
> What dial-peer is it matching? and what codec does that dial-peer have?
>
> Do the following on the CUCME:
> debug voice dialpeer inout
> check which dial-peer is incoming, and then check the codec on that
> dial-peer.
> Then if the endpoint is SCCP it should work, but if it's SIP then its
> different. You would have to check the codec under voice register pool.
>
> Regards,
>
> On Mon, May 23, 2011 at 7:03 PM, Michael Luo  wrote:
>
>> I was testing CUBE feature (H.323 to H.323).  The call can ring the
>> destination.  But when called party pick up, the call got disconnected with
>> fast busy.
>>
>> debug cch323 h225 shows "//31/8087591D0900/H323/cch323_h225_send_release:
>> Cause = 65".
>>
>> Per the symptom and debug, it seems to be a codec issue.  But I've checked
>> the following:
>>
>> 1) "allow h323 to h323"
>> 2) incoming dial-peer was g.729r8
>> 3) outgoing dial-peer was g.729r8
>> 4) "Wait for Far End H.245 Terminal Capability Set" was unchecked
>>
>> What else shall I check?
>>
>> Thanks!
>> Michael
>>
>> ___
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>> visit www.ipexpert.com
>>
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>>
>
>
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[OSL | CCIE_Voice] CUBE H.323 to H.323

[Michael Luo]

I was testing CUBE feature (H.323 to H.323).  The call can ring the
destination.  But when called party pick up, the call got disconnected with
fast busy.

debug cch323 h225 shows "//31/8087591D0900/H323/cch323_h225_send_release:
Cause = 65".

Per the symptom and debug, it seems to be a codec issue.  But I've checked
the following:

1) "allow h323 to h323"
2) incoming dial-peer was g.729r8
3) outgoing dial-peer was g.729r8
4) "Wait for Far End H.245 Terminal Capability Set" was unchecked

What else shall I check?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] MVA with Hairpinning

[Michael Luo]

It looks like a tanscoder is required if we want to do G.729 between MGCP
and H323 GWs.

MVA IVR only supports G.711.

Michael

On Fri, May 20, 2011 at 10:34 AM, George Goglidze wrote:

> Hi Michael,
>
> Have you tried to put g729 on the dial-peer?
> Because if the h323 Gateway and MGCP gateway are in different regions and
> use g729 between them, and if on incoming dial-peer on the h323 gateway
> you're trying to negotiate g711, the negotiation will fail.
>
> Try putting:
>
> voice class codec 1
>  codec preference 1 g711u
>  codec preference 2 g729r
>
>
> dial-peer voice 2888 voip
>  service mva
>  incoming called-number 2888
>  dtmf-relay h245-alphanumeric
>  voice-class codec 1
>
>  no vad
> !
> dial-peer voice 2999 voip
>  destination-pattern 2999
>  session target ipv4:10.10.210.10 <-- CUCM Publisher
>  dtmf-relay h245-alphanumeric
>  voice-class codec 1
>  no vad
>
> Let me know if it works, as g729 should definitely work with MVA.
>
> Regards,
>
> On Fri, May 20, 2011 at 4:09 PM, Michael Luo  wrote:
>
>> I'm testing MVA with Hairpinning (
>> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1126351
>> )
>>
>> Call flow as below:
>>
>> PSTN Phone -> MGCP GW -> CUCM -> H323 GW -> MVA application
>>
>> I have no problem getting it worked if everything was in the same region
>> (G711u was used).
>>
>> But if I put gateways into different region and try to use G.729.  The
>> incoming call from PSTN just siliently disconnected.  I was expecting to
>> hear the MVA welcome prompts.
>>
>> Where I can find more documents regarding codec handling in MVA
>> hairpinning scenario?  I'd like to know if the VXML app on CUCM accepts
>> G.729 or not.  If not, how do we configure codec in incoming and outgoing
>> dial-peers?
>>
>> H323 GW configuration as below.  2888 is the incoming DID.  2999 is the
>> MVA number.
>>
>> dial-peer voice 2888 voip
>>  service mva
>>  incoming called-number 2888
>>  dtmf-relay h245-alphanumeric
>>  codec g711ulaw
>>  no vad
>> !
>> dial-peer voice 2999 voip
>>  destination-pattern 2999
>>  session target ipv4:10.10.210.10 <-- CUCM Publisher
>>  dtmf-relay h245-alphanumeric
>>  codec g711ulaw
>>  no vad
>>
>> voice service voip
>>  allow h323 to h323
>>
>> application
>>  service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
>>
>> Thanks!
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] Passed CCIE#28970!!!!!!!

[Michael Luo]

Rogers,

Congratulations!

Very nice number by the way.  :)

Michael

On Fri, May 20, 2011 at 1:38 AM, Rogers Ochieng wrote:

> Freshly minted! I took my exam yesterday in Bangalore and the good news is
> here!!!
>
> Thanks to my study partners Michael, Fatai and Rahul!
>
> CCIE#28970 - Voice
> Rogers Ochieng
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] MVA with Hairpinning

[Michael Luo]

I'm testing MVA with Hairpinning (
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1126351
)

Call flow as below:

PSTN Phone -> MGCP GW -> CUCM -> H323 GW -> MVA application

I have no problem getting it worked if everything was in the same region
(G711u was used).

But if I put gateways into different region and try to use G.729.  The
incoming call from PSTN just siliently disconnected.  I was expecting to
hear the MVA welcome prompts.

Where I can find more documents regarding codec handling in MVA hairpinning
scenario?  I'd like to know if the VXML app on CUCM accepts G.729 or not.
If not, how do we configure codec in incoming and outgoing dial-peers?

H323 GW configuration as below.  2888 is the incoming DID.  2999 is the MVA
number.

dial-peer voice 2888 voip
 service mva
 incoming called-number 2888
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 2999 voip
 destination-pattern 2999
 session target ipv4:10.10.210.10 <-- CUCM Publisher
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

voice service voip
 allow h323 to h323

application
 service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml

Thanks!
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Re: [OSL | CCIE_Voice] CME presence - BLF speed dial vs. monitor button

[Michael Luo]

I tried.  Both BLF and monitor button can monitor DND.

On Wed, May 18, 2011 at 11:16 AM, Nirvair Sahota <
nirvair.sah...@sbcglobal.net> wrote:

> Michael,
>
> I believe, when you use "BLF Speed Dial"  and enble DND (press DND button),
> it will not show on the monitoring phone, where as monitoring "button 3m1"
> will work with DND. Try it.
>
> Nirvair
>
> --- On *Wed, 5/18/11, Michael Luo * wrote:
>
>
> From: Michael Luo 
> Subject: [OSL | CCIE_Voice] CME presence - BLF speed dial vs. monitor
> button
> To: ccie_voice@onlinestudylist.com
> Date: Wednesday, May 18, 2011, 8:30 AM
>
>
> On CME phone, we have different ways to monitor another phone's status
> (Ringing, Offhook, DND).
>
> I was trying to understand the difference between BLF speed dial and the
> monitor button (e.g. "button 3m1").  They seem to have the same feature.
> What are the difference?
>
> Thanks!
> Michael
>
> -Inline Attachment Follows-
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
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[OSL | CCIE_Voice] CME presence - BLF speed dial vs. monitor button

[Michael Luo]

On CME phone, we have different ways to monitor another phone's status
(Ringing, Offhook, DND).

I was trying to understand the difference between BLF speed dial and the
monitor button (e.g. "button 3m1").  They seem to have the same feature.
What are the difference?

Thanks!
Michael
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[OSL | CCIE_Voice] transcoder registration consideration

[Michael Luo]

Call flow is as below:

IP Phone -> CUCM -> GK-controlled Trunk(WAN) -> CME -> CUE voicemail pilot

Requirements:
1) CUE only accept G.711
2) To save bandwidth, we want G.729 over the WAN

Solution:
Configure transcoder on CME to translate between G.729 and G.711.

Questions:
>From the test, I found out that I can register the transcoder to either CUCM
or CME.  Which one is a better solution in real life scenario?  And why?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] How to troubleshoot CME transcoding

[Michael Luo]

never mind, missing incoming dial-peer

On Tue, May 17, 2011 at 6:52 PM, Michael Luo  wrote:

> I have configured transcoder on CME.  But it's not working.  Incoming VoIP
> call got fast busy.  I've checked all config.  What debug commands we can
> use to troubleshoot this?
>
> telephony-service
>  sdspfarm units 3
>  sdspfarm transcode sessions 3
>  sdspfarm tag 1 sc-cfb
>  sdspfarm tag 2 sc-xcd
>
>
>
> #sh dspf dsp all
> SLOT DSP VERSION  STATUS CHNL USE   TYPERSC_ID BRIDGE_ID PKTS_TXED
> PKTS_RXED
>
> 03   24.3.4   UP N/A  FREE  xcode  1  - -
> -
> 03   24.3.4   UP N/A  FREE  xcode  1  - -
> -
> 03   24.3.4   UP N/A  FREE  xcode  1  - -
> -
> 07   24.3.4   UP N/A  FREE  conf   2  - -
> -
> 08   24.3.4   UP N/A  FREE  conf   2  - -
> -
> 08   24.3.4   UP N/A  FREE  conf   2  - - -
>
> #sh sccp ccm group
> CCM Group Identifier: 1
>  Description: None
>  Binded Interface: None
>  Associated CCM Id: 1, Priority in this CCM Group: 1
>  Associated Profile: 1, Registration Name: sc-cfb
>  Associated Profile: 2, Registration Name: sc-xcd
>
> #sh dspfarm profile
> Dspfarm Profile Configuration
>
>  Profile ID = 2, Service = TRANSCODING, Resource ID = 1
>  Profile Description :
>  Profile Service Mode : Non Secure
>  Profile Admin State : UP
>  Profile Operation State : ACTIVE
>  Application : SCCP   Status : ASSOCIATED
>  Resource Provider : FLEX_DSPRM   Status : UP
>  Number of Resource Configured : 3
>  Number of Resource Available : 3
>  Codec Configuration
>  Codec : g711ulaw, Maximum Packetization Period : 30
>  Codec : g711alaw, Maximum Packetization Period : 30
>  Codec : g729ar8, Maximum Packetization Period : 60
>  Codec : g729abr8, Maximum Packetization Period : 60
> Dspfarm Profile Configuration
>
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[OSL | CCIE_Voice] How to troubleshoot CME transcoding

[Michael Luo]

I have configured transcoder on CME.  But it's not working.  Incoming VoIP
call got fast busy.  I've checked all config.  What debug commands we can
use to troubleshoot this?

telephony-service
 sdspfarm units 3
 sdspfarm transcode sessions 3
 sdspfarm tag 1 sc-cfb
 sdspfarm tag 2 sc-xcd



#sh dspf dsp all
SLOT DSP VERSION  STATUS CHNL USE   TYPERSC_ID BRIDGE_ID PKTS_TXED
PKTS_RXED

03   24.3.4   UP N/A  FREE  xcode  1  - -
-
03   24.3.4   UP N/A  FREE  xcode  1  - -
-
03   24.3.4   UP N/A  FREE  xcode  1  - -
-
07   24.3.4   UP N/A  FREE  conf   2  - -
-
08   24.3.4   UP N/A  FREE  conf   2  - -
-
08   24.3.4   UP N/A  FREE  conf   2  - - -

#sh sccp ccm group
CCM Group Identifier: 1
 Description: None
 Binded Interface: None
 Associated CCM Id: 1, Priority in this CCM Group: 1
 Associated Profile: 1, Registration Name: sc-cfb
 Associated Profile: 2, Registration Name: sc-xcd

#sh dspfarm profile
Dspfarm Profile Configuration

 Profile ID = 2, Service = TRANSCODING, Resource ID = 1
 Profile Description :
 Profile Service Mode : Non Secure
 Profile Admin State : UP
 Profile Operation State : ACTIVE
 Application : SCCP   Status : ASSOCIATED
 Resource Provider : FLEX_DSPRM   Status : UP
 Number of Resource Configured : 3
 Number of Resource Available : 3
 Codec Configuration
 Codec : g711ulaw, Maximum Packetization Period : 30
 Codec : g711alaw, Maximum Packetization Period : 30
 Codec : g729ar8, Maximum Packetization Period : 60
 Codec : g729abr8, Maximum Packetization Period : 60
Dspfarm Profile Configuration
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Re: [OSL | CCIE_Voice] VoiceMail (Exam result FAILED)

[Michael Luo]

Why do we need to add tftp servers in cxn?  (since voicemail works without
it).

Thanks!
Michael

On Fri, May 13, 2011 at 2:53 PM,  wrote:

> Possible things:
>
> Wrong ip address for cue module
> Did not add tftp servers in cxn
> Left mwi on
> Did not choose correct timezone
> They were lazy and assumed you were wrong
> They stopped grading because you already failed
>
> -Original Message-
> From: Naoufal Kerboute 
> Sender: ccie_voice-boun...@onlinestudylist.com
> Date: Fri, 13 May 2011 18:45:06
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] VoiceMail (Exam result FAILED)
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] 3750 srr queue input vs. output?

[Michael Luo]

Hi,

I'm studying QoS on 3750 switch.

I'm always confused with SRR input and output queuing.  I know there are two
input queues and four output queues.  When do we use input queuing?  When do
we use output queuing?

Thanks!

Michael
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Re: [OSL | CCIE_Voice] h323-gateway voip interface

[Michael Luo]

I know.  But what's the point of putting "h323-gateway voip interface" under
an interface?

If it enable the interface to talk to the GK, shouldn't all the traffic
comes out from this interface?  Why we're seeing a different IP from GK
side?

Thanks!
Michael

On Wed, Apr 20, 2011 at 1:00 PM, Randall Crumm <
randall.cr...@flextronics.com> wrote:

> h323-gateway voip bind srcaddr is used if the gw is h.323 GW
>
> The other is to register to the GK
>
>
>
> HTH
>
>
>
> Randall
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Michael Luo
> *Sent:* Wednesday, April 20, 2011 10:01 AM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] h323-gateway voip interface
>
>
>
> I've read Cisco IOS command reference, but I'm still not sure about the
> purpose of "h323-gateway voip interface" command.
>
> Moreover, I was trying to understand the difference between "h323-gateway
> voip interface" and "h323-gateway voip bind srcaddr".
>
> Here's my config on the gateway router:
>
>
> ===
> interface Fa0/0
>  ip address 192.168.1.254 255.255.255.0
>  h323-gateway voip bind srcaddr 192.168.1.254
>
> interface loopback0
>  ip address 10.1.1.254 255.255.255.0
>  h323-gateway voip interface
>  h323-gateway voip id GK ipaddr 10.2.2.254
>  h323-gateway voip h323-id CUCME
>  h323-gateway voip tech-prefix 408
> ===
>
> From my understanding, the "h323-gateway voip interface" command should
> dictate the source address that being used to communicate with gatekeeper.
> i.e. it should be 10.1.1.254 (loopback0)
>
> However, when I do a "show gatekeeper endpoint" command on the gatekeeper
> router, it's showing the RAS address as 192.168.1.1.254 (Fa0/0).  How can I
> make the gateway router to use loopback0 communicate with GK, while using
> the Fa0/0 interface communicate with other H323 peers (such as CUCM)?
>
> ===
> GK#show gatekeeper endpoint
> GATEKEEPER ENDPOINT REGISTRATION
> 
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
> --- - --- - - -
> 192.168.1.254  1720  192.168.1.254  60745 GKVOIP-GW
> H323-ID: CUCME
> Voice Capacity Max.=  Avail.=  Current.= 0
> Total number of active registrations = 1
> ===
>
> - IOS image being used is c3825-adventerprisek9_ivs-mz.124-24.T4.bin
> - I also tried making the gateway router a "H323-GW" instead of "VOIP-GW"
> by adding "allow h323 to h323" command.  Same behavior.
>
>
> Thanks!
> Michael
> Legal Disclaimer: The information contained in this message may be
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Re: [OSL | CCIE_Voice] h323-gateway voip interface

[Michael Luo]

I rebooted the routers (both gatekeeper and the gateway).

On Wed, Apr 20, 2011 at 1:04 PM, Rogers Ochieng wrote:

> are you forcing the gateway to re-register after each change? no gateway
> then gateway?
>
> On 20 April 2011 20:00, Michael Luo  wrote:
>
>> I've read Cisco IOS command reference, but I'm still not sure about the
>> purpose of "h323-gateway voip interface" command.
>>
>> Moreover, I was trying to understand the difference between "h323-gateway
>> voip interface" and "h323-gateway voip bind srcaddr".
>>
>> Here's my config on the gateway router:
>>
>>
>> ===
>> interface Fa0/0
>>  ip address 192.168.1.254 255.255.255.0
>>  h323-gateway voip bind srcaddr 192.168.1.254
>>
>> interface loopback0
>>  ip address 10.1.1.254 255.255.255.0
>>  h323-gateway voip interface
>>  h323-gateway voip id GK ipaddr 10.2.2.254
>>  h323-gateway voip h323-id CUCME
>>  h323-gateway voip tech-prefix 408
>> ===
>>
>> From my understanding, the "h323-gateway voip interface" command should
>> dictate the source address that being used to communicate with gatekeeper.
>> i.e. it should be 10.1.1.254 (loopback0)
>>
>> However, when I do a "show gatekeeper endpoint" command on the gatekeeper
>> router, it's showing the RAS address as 192.168.1.1.254 (Fa0/0).  How can I
>> make the gateway router to use loopback0 communicate with GK, while using
>> the Fa0/0 interface communicate with other H323 peers (such as CUCM)?
>>
>> ===
>> GK#show gatekeeper endpoint
>> GATEKEEPER ENDPOINT REGISTRATION
>> 
>> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
>> Flags
>> --- - --- - - 
>> -
>> 192.168.1.254  1720  192.168.1.254  60745 GKVOIP-GW
>> H323-ID: CUCME
>> Voice Capacity Max.=  Avail.=  Current.= 0
>> Total number of active registrations = 1
>> ===
>>
>> - IOS image being used is c3825-adventerprisek9_ivs-mz.124-24.T4.bin
>> - I also tried making the gateway router a "H323-GW" instead of "VOIP-GW"
>> by adding "allow h323 to h323" command.  Same behavior.
>>
>>
>> Thanks!
>> Michael
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
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[OSL | CCIE_Voice] h323-gateway voip interface

[Michael Luo]

I've read Cisco IOS command reference, but I'm still not sure about the
purpose of "h323-gateway voip interface" command.

Moreover, I was trying to understand the difference between "h323-gateway
voip interface" and "h323-gateway voip bind srcaddr".

Here's my config on the gateway router:


===
interface Fa0/0
 ip address 192.168.1.254 255.255.255.0
 h323-gateway voip bind srcaddr 192.168.1.254

interface loopback0
 ip address 10.1.1.254 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id GK ipaddr 10.2.2.254
 h323-gateway voip h323-id CUCME
 h323-gateway voip tech-prefix 408
===

>From my understanding, the "h323-gateway voip interface" command should
dictate the source address that being used to communicate with gatekeeper.
i.e. it should be 10.1.1.254 (loopback0)

However, when I do a "show gatekeeper endpoint" command on the gatekeeper
router, it's showing the RAS address as 192.168.1.1.254 (Fa0/0).  How can I
make the gateway router to use loopback0 communicate with GK, while using
the Fa0/0 interface communicate with other H323 peers (such as CUCM)?

===
GK#show gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
192.168.1.254  1720  192.168.1.254  60745 GKVOIP-GW
H323-ID: CUCME
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 1
===

- IOS image being used is c3825-adventerprisek9_ivs-mz.124-24.T4.bin
- I also tried making the gateway router a "H323-GW" instead of "VOIP-GW" by
adding "allow h323 to h323" command.  Same behavior.


Thanks!
Michael
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[OSL | CCIE_Voice] Confusion about "priority queue" and "bandwidth shape"

[Michael Luo]

On LAN QoS, I've seen a configuration like below:

g0/1/0
>  srr-queue bandwidth share 1 30 40 30
>  srr-queue bandwidth shape 4 0 0 0
>  priority-queue out
>

>From my understanding, "priority-queue out" will make Q1 the priority queue,
which means, until Q1 is empty, there won't be any transmission on Q2, Q3
and Q4.

If that's the case, what's the point of shaping Q1's bandwidth with
"bandwidth shape 4 0 0 0"?  Shouldn't we give Q1 100% bandwidth (since it's
priority queue)?

Thanks!
Michael
___
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Re: [OSL | CCIE_Voice] CFUR for and by fields

[Michael Luo]

Have you enabled ISDN redirect-ie?  (both on CUCM and router).

On Thu, Mar 31, 2011 at 11:04 AM, adam compton  wrote:

> All,
>
> I am desperately seeking help on how to alter the For and By fields on a
> CFUR to an SRST phone and a branch site.  This is what I need to display on
> the phone:
>
> Forwarded 5001
> For: +16178631001 ( 1... )
> by : +16178631001 ( 1... )
>
> I've been trying for a month to figure out how to display this with no
> luck.  I've watched every video I know of in the IPexpert catalog, and
> nothing references it that I've seen.  Any help would be appreciated,
> because i am at my wits end on this one.
>
> Adam Compton
>
> ___
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> visit www.ipexpert.com
>
>
___
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[OSL | CCIE_Voice] CME: assign blf-speed-dial to a specific button?

[Michael Luo]

I'm using CME 7.1 and 7965 phone.

Let say, I want to configure a blf-speed-dial on the 5th button on the
phone.  How should I configure that?  I know the "blf-speed-dial" command.
But I couldn't figure out how to assign it to a specific button.

Thanks!
Michael
___
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Re: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen

[Michael Luo]

Here's what I found out from CCM traces:

1) When the call ring out.  CCM instructs the phone display the called
number as "5551212" which is the desired behavior.

2) CCM receives a H.225 NOTIFY message from the GW with "called number =
95551212".

3) CCM instructs the phone to update the screen as "To: 95551212".

Any way to change this behavior?

Thanks!
Michael

On Sun, Mar 27, 2011 at 1:40 PM, Tamer Ismail  wrote:

> Hello,
>
> All manipulations done in route pattern will not make your requirements.
>
> You have to pre-dot in route pattern, then prefix in route list.
>
>
>
> Best regards,
>
> Tamer Ismail
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Michael Luo
> *Sent:* Sunday, March 27, 2011 8:08 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Remove prefix 9 from called number on phone
> screen
>
>
>
> I'm using CUCM 7.1.2 with H.323 gateway.
>
> In H323 GW, the dial-peer for local number is 9 with 7-digit.  For example
> "destination-pattern 9...".
>
> When user make a local call, the IP phone will display the called number
> with prefix 9.  For example "To: 95551212".
>
> If we're not allowed to change dial-peer on GW, what's the easiest way to
> remove the prefix 9 from phone screen?
>
> I tried to use route pattern to discard the 9 then prefix 9 in route
> group.  But it's still showing up.  Do I have to do called-party transform
> pattern?  I was trying to avoid that.
>
> Thanks!
> Michael
>
___
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Re: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen

[Michael Luo]

Thanks for the reply.  That's exactly what I did:

1) In route pattern, use Pre-Dot to remove prefix 9
2) In associated route list > route group, prefix 9

But it's still showing the prefix 9 on phone's screen.

Thanks!
Michael

On Sun, Mar 27, 2011 at 1:40 PM, Tamer Ismail  wrote:

> Hello,
>
> All manipulations done in route pattern will not make your requirements.
>
> You have to pre-dot in route pattern, then prefix in route list.
>
>
>
> Best regards,
>
> Tamer Ismail
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Michael Luo
> *Sent:* Sunday, March 27, 2011 8:08 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Remove prefix 9 from called number on phone
> screen
>
>
>
> I'm using CUCM 7.1.2 with H.323 gateway.
>
> In H323 GW, the dial-peer for local number is 9 with 7-digit.  For example
> "destination-pattern 9...".
>
> When user make a local call, the IP phone will display the called number
> with prefix 9.  For example "To: 95551212".
>
> If we're not allowed to change dial-peer on GW, what's the easiest way to
> remove the prefix 9 from phone screen?
>
> I tried to use route pattern to discard the 9 then prefix 9 in route
> group.  But it's still showing up.  Do I have to do called-party transform
> pattern?  I was trying to avoid that.
>
> Thanks!
> Michael
>
___
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[OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen

[Michael Luo]

I'm using CUCM 7.1.2 with H.323 gateway.

In H323 GW, the dial-peer for local number is 9 with 7-digit.  For example
"destination-pattern 9...".

When user make a local call, the IP phone will display the called number
with prefix 9.  For example "To: 95551212".

If we're not allowed to change dial-peer on GW, what's the easiest way to
remove the prefix 9 from phone screen?

I tried to use route pattern to discard the 9 then prefix 9 in route group.
But it's still showing up.  Do I have to do called-party transform pattern?
I was trying to avoid that.

Thanks!
Michael
___
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[OSL | CCIE_Voice] encapsulation frame-relay cisco?

[Michael Luo]

I'm running ISO 12.4(24)T4.

When I tried to configure Auto QoS under the frame-relay DLCI, I got the
complain that the sub-interface encapsulation type was not "Cisco".

But I don't see "cisco" as an option while configuring encapsulation.  Only
ietf and MFR are available.  Can anyone help?

R1(config-if)#encapsulation frame-relay ?
  MFR   Multilink Frame Relay bundle interface
  ietf  Use RFC1490/RFC2427 encapsulation
  

Thanks!
Michael
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Re: [OSL | CCIE_Voice] CUE MWI "Outcalling"

[Michael Luo]

Never mind.  It's translation rules.

On Mon, Mar 21, 2011 at 2:39 PM, Michael Luo  wrote:

> There were three MWI delivery options for CUE - "Outcalling", "SIP
> Subscribe Notify" and "SIP Unsolicited Notify".  I wanted to test
> "Outcalling" with CUE 7.0.6 and CME 7.1.
>
> Per http://ciscovoiceguru.com/518/cue-mwi-notification-methods/,
> "Outcalling" need the following config.  But when I left a message, CUE will
> call the destination phone with two rings but didn't light up the lamp.  I
> was expecting no ring and lamp light up.
>
> I followed the "MWI Troubleshooting" doc on Cisco but couldn't find any
> thing significant.  Can anyone help?  Thanks!
>
> CME:
> dial-peer voice 3600 voip
> destination-pattern 3600
> session protocol sipv2
> session target ipv4:10.10.202.2
> incoming called-number 399[89]….
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
> !
> ephone-dn  11
> number 3998…. no-reg primary
> mwi off
> !
> ephone-dn  12
> number 3999…. no-reg primary
> mwi on
>
> CUE:
> ccn application ciscomwiapplication aa
> parameter "strMWI_OFF_DN" "3998"
> parameter "strMWI_ON_DN" "3999"
> end application
>
>
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[OSL | CCIE_Voice] CUE MWI "Outcalling"

[Michael Luo]

There were three MWI delivery options for CUE - "Outcalling", "SIP Subscribe
Notify" and "SIP Unsolicited Notify".  I wanted to test "Outcalling" with
CUE 7.0.6 and CME 7.1.

Per http://ciscovoiceguru.com/518/cue-mwi-notification-methods/,
"Outcalling" need the following config.  But when I left a message, CUE will
call the destination phone with two rings but didn't light up the lamp.  I
was expecting no ring and lamp light up.

I followed the "MWI Troubleshooting" doc on Cisco but couldn't find any
thing significant.  Can anyone help?  Thanks!

CME:
dial-peer voice 3600 voip
destination-pattern 3600
session protocol sipv2
session target ipv4:10.10.202.2
incoming called-number 399[89]….
dtmf-relay sip-notify
codec g711ulaw
no vad
!
ephone-dn  11
number 3998…. no-reg primary
mwi off
!
ephone-dn  12
number 3999…. no-reg primary
mwi on

CUE:
ccn application ciscomwiapplication aa
parameter "strMWI_OFF_DN" "3998"
parameter "strMWI_ON_DN" "3999"
end application
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[OSL | CCIE_Voice] CUE/CME MWI config

[Michael Luo]

Can anyone send me a link to CUE/CME MWI configuration example?

Whenever I left a message, CUE VM pilot will call the phone instead of
lighting up the phone.

Attached are the config and debug.

Thanks!
Michael
Mar 21 12:09:14.352 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:19984002@142.102.66.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
Max-Forwards: 70
To: 
From: ;tag=cue782f4c
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 INVITE
Content-Length: 182
Contact: 
Content-Type: application/sdp
Cisco-Gcid: D69A3781-012E-1000-4000-001125CUCE68
Call-Info: 
;method="NOTIFY;Event=telephone-event;Duration=2000"
Allow-Events: telephone-event

v=0
o=CiscoSystemsSIP-Workflow-App-UserAgent 1922 1922 IN IP4 142.102.66.253
s=SIP Call
c=IN IP4 142.102.66.253
t=0 0
m=audio 16912 RTP/AVP 0
a=rtpmap:0 pcmu/8000
a=ptime:20

Mar 21 12:09:14.356 HKT: //-1/D12D2B6F8048/SIP/Media/sipSPICopyPeerDataToCCB: 
Firewall traversal is not enabled
Mar 21 12:09:14.356 HKT: //29/D12D2B6F8048/SIP/Media/sipSPISetMediaSrcAddr: 
Media src addr for stream 1 = 142.102.66.254
Mar 21 12:09:14.356 HKT: //29/D12D2B6F8048/SIP/Media/sipSPIDoPtimeNegotiation: 
Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec 
g711ulaw
Mar 21 12:09:14.356 HKT: //29/D12D2B6F8048/SIP/Media/sipSPIUpdCallWithSdpInfo: 
Preferred Codec: g711ulaw, bytes :160
Preferred  DTMF relay  : sip-notify
Preferred NTE payload  : 101
Early Media: No
Delayed Media  : No
Bridge Done: No
New Media  : No
DSP DNLD Reqd  : No

Mar 21 12:09:14.356 HKT: //29/D12D2B6F8048/SIP/Media/sipSPISetMediaSrcAddr: 
Media src addr for stream 1 = 142.102.66.254
Mar 21 12:09:14.356 HKT: //29/D12D2B6F8048/SIP/Media/sipSPIUpdCallWithSdpInfo: 
  Stream type: voice-only
  Media line : 1
  State  : STREAM_ADDING (2)
  Stream address type: 1
  Callid : -1
  Negotiated Codec   : g711ulaw, bytes :160
  Nego. Codec payload: 0 (tx), 0 (rx)
  Negotiated DTMF relay  : sip-notify
  Negotiated NTE payload : 0 (tx), 0 (rx)
  Negotiated CN payload  : 0
  Media Srce Addr/Port   : [142.102.66.254]:0
  Media Dest Addr/Port   : [142.102.66.253]:16912

Mar 21 12:09:14.356 HKT: //-1//SIP/Media/sipSPIReserveRtpPort: 
reserved port 19284 for stream 1
Mar 21 12:09:14.360 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
From: ;tag=cue782f4c
To: 
Date: Mon, 21 Mar 2011 04:09:14 GMT
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Mar 21 12:09:14.368 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
From: ;tag=cue782f4c
To: ;tag=1498EB8-1467  
Date: Mon, 21 Mar 2011 04:09:14 GMT
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "SC Phone2" 
;party=called;screen=no;privacy=off
Contact: 
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Mar 21 12:09:19.356 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Received: 
CANCEL sip:19984002@142.102.66.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
Max-Forwards: 70
To: 
From: ;tag=cue782f4c
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 CANCEL
Content-Length: 0


Mar 21 12:09:19.360 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
From: ;tag=cue782f4c
To: 
Date: Mon, 21 Mar 2011 04:09:19 GMT
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 CANCEL
Content-Length: 0


Mar 21 12:09:19.364 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
  
From: ;tag=cue782f4c
To: ;tag=1498EB8-1467
Date: Mon, 21 Mar 2011 04:09:19 GMT
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0


Mar 21 12:09:19.376 HKT: //-1//SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:19984002@142.102.66.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 142.102.66.253:5060;branch=z9hG4bKAJiPb8i3BCCzYXDkIAI37g~~7
Max-Forwards: 70
To: ;tag=1498EB8-1467
From: ;tag=cue782f4c
Call-ID: 130068055436913@142.102.66.253
CSeq: 1 ACK
Content-Length: 0version 12.4
service timestamps debug datetime msec localtime sh

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

[Michael Luo]

I was trying to understand this part - "The br2 phone need to be registered
with cucm. Not with cme."

Why was that?

>From my troubleshooting, it looks like the CME phone was listening to the
correct IP address/port - 239.1.1.1 16384.  But didn't receive any MOH
music.

I was under the impression that the music was continuously being played by
the router.  The CME phones just need to "tune" to the right channel to
receive music.

Thanks!
Michael

On Sun, Mar 20, 2011 at 3:24 PM, Jimmy  wrote:

> Mate
>
>Coorect me if I am wrong.  Doesn't matter if it's pstn or gatekeeper
> trunk.
>
> When cucm phone press hold button. The phone at cme site will will receive
> moh from cucm.
>
> For moh from flash to work. The br2 phone need to be registered with cucm.
> Not with cme.
>
> I hope I am making sense. I am not good at writing emails.
>
>
>
> Regards
>
>
> Sent from my iPad
>
> On Mar 21, 2011, at 8:17 AM, Michael Luo  wrote:
>
> One thing I noticed was the word "stopped" in show ip mroute output.  This
> seems to be the problem by comparing with a working one.  But I don't know
> how to fix it.  Thanks!
>
> R3#sh ip mro
> IP Multicast Routing Table
> Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
> Connected,
>L - Local, P - Pruned, R - RP-bit set, F - Register flag,
>T - SPT-bit set, J - Join SPT, M - MSDP created entry,
>X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,
>U - URD, I - Received Source Specific Host Report,
>Z - Multicast Tunnel, z - MDT-data group sender,
>Y - Joined MDT-data group, y - Sending to MDT-data group,
>V - RD & Vector, v - Vector
> Outgoing interface flags: H - Hardware switched, A - Assert winner
>  Timers: Uptime/Expires
>  Interface state: Interface, Next-Hop or VCD, State/Mode
>
> (*, 239.1.1.1), 00:01:51/*stopped*, RP 0.0.0.0, flags: DCL
>   Incoming interface: Null, RPF nbr 0.0.0.0
>   Outgoing interface list:
> Vlan502, Forward/Dense, 00:00:14/00:00:00
>
>
>
> On Sun, Mar 20, 2011 at 1:48 PM, Michael Luo < 
> hout...@gmail.com> wrote:
>
>>
>> Call flow as below:
>>
>> CME phone -> CME (H.323 GW) -> GK -> GK-controlled Trunk -> CUCM -> UCM
>> Phone
>>
>> Problem: When UCM Phone press the "Hold" button, CME phone hears silence.
>> Expected behavior: CME phone hears MOH from router flash -
>> music-on-hold.au.
>>
>> Troubleshooting done:
>> 1) Verified MOH works as expected between CME phones. (CME phone can hear
>> MOH)
>> 2) Verified MOH works as expected with PSTN phones (PSTN phone can hear
>> MOH)
>>
>> Config highlights:
>>
>> =
>> ip multicast-routing
>> !
>> interface vlan502
>>  description e-phone vlan
>>  ip address 142.102.66.254 255.255.255.0
>>  ip pim dense
>>
>> interface Loopback0
>>  ip address 142.1.66.254 255.255.255.255
>>
>> ccm-manager music-on-hold
>>
>> telephony-service
>>  ip source-address 142.102.66.254 port 2000
>>  moh <http://music-on-hold.au>music-on-hold.au
>>  multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
>> =
>>
>> debug highlights:
>> =
>> R3#sh ccm mus
>> Current active multicast sessions : 1
>>  Multicast   RTP port   Packets   Call   CodecIncoming
>>  Address number in/outid  Interface
>> ===
>> 239.1.1.1 16384   1545/154595   g711ulaw  Vl502
>>
>>
>> R3#sh call active voice compact
>>A/O FAX T Codec   typePeer Address   IP
>> R:
>> Total call-legs: 2
>> 94 ANS T52g711ulawVOIPP2001
>> 239.1.1.1:16384
>> 95 ORG T52g711ulawTELEP4001
>> =
>>
>> detailed debug attached.
>>
>> Thanks!
>> Michael
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit <http://www.ipexpert.com>www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] MOH across GK-controlled trunk?

[Michael Luo]

One thing I noticed was the word "stopped" in show ip mroute output.  This
seems to be the problem by comparing with a working one.  But I don't know
how to fix it.  Thanks!

R3#sh ip mro
IP Multicast Routing Table
Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C - Connected,
   L - Local, P - Pruned, R - RP-bit set, F - Register flag,
   T - SPT-bit set, J - Join SPT, M - MSDP created entry,
   X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,
   U - URD, I - Received Source Specific Host Report,
   Z - Multicast Tunnel, z - MDT-data group sender,
   Y - Joined MDT-data group, y - Sending to MDT-data group,
   V - RD & Vector, v - Vector
Outgoing interface flags: H - Hardware switched, A - Assert winner
 Timers: Uptime/Expires
 Interface state: Interface, Next-Hop or VCD, State/Mode

(*, 239.1.1.1), 00:01:51/*stopped*, RP 0.0.0.0, flags: DCL
  Incoming interface: Null, RPF nbr 0.0.0.0
  Outgoing interface list:
Vlan502, Forward/Dense, 00:00:14/00:00:00



On Sun, Mar 20, 2011 at 1:48 PM, Michael Luo  wrote:

>
> Call flow as below:
>
> CME phone -> CME (H.323 GW) -> GK -> GK-controlled Trunk -> CUCM -> UCM
> Phone
>
> Problem: When UCM Phone press the "Hold" button, CME phone hears silence.
> Expected behavior: CME phone hears MOH from router flash -
> music-on-hold.au.
>
> Troubleshooting done:
> 1) Verified MOH works as expected between CME phones. (CME phone can hear
> MOH)
> 2) Verified MOH works as expected with PSTN phones (PSTN phone can hear
> MOH)
>
> Config highlights:
>
> =
> ip multicast-routing
> !
> interface vlan502
>  description e-phone vlan
>  ip address 142.102.66.254 255.255.255.0
>  ip pim dense
>
> interface Loopback0
>  ip address 142.1.66.254 255.255.255.255
>
> ccm-manager music-on-hold
>
> telephony-service
>  ip source-address 142.102.66.254 port 2000
>  moh music-on-hold.au
>  multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
> =
>
> debug highlights:
> =
> R3#sh ccm mus
> Current active multicast sessions : 1
>  Multicast   RTP port   Packets   Call   CodecIncoming
>  Address number in/outid  Interface
> ===
> 239.1.1.1 16384   1545/154595   g711ulaw  Vl502
>
>
> R3#sh call active voice compact
>A/O FAX T Codec   typePeer Address   IP
> R:
> Total call-legs: 2
> 94 ANS T52g711ulawVOIPP2001
> 239.1.1.1:16384
> 95 ORG T52g711ulawTELEP4001
> =
>
> detailed debug attached.
>
> Thanks!
> Michael
>
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Re: [OSL | CCIE_Voice] MOH across GK-controlled trunk?

[Michael Luo]

Call flow as below:

CME phone -> CME (H.323 GW) -> GK -> GK-controlled Trunk -> CUCM -> UCM
Phone

Problem: When UCM Phone press the "Hold" button, CME phone hears silence.
Expected behavior: CME phone hears MOH from router flash - music-on-hold.au.

Troubleshooting done:
1) Verified MOH works as expected between CME phones. (CME phone can hear
MOH)
2) Verified MOH works as expected with PSTN phones (PSTN phone can hear MOH)

Config highlights:

=
ip multicast-routing
!
interface vlan502
 description e-phone vlan
 ip address 142.102.66.254 255.255.255.0
 ip pim dense

interface Loopback0
 ip address 142.1.66.254 255.255.255.255

ccm-manager music-on-hold

telephony-service
 ip source-address 142.102.66.254 port 2000
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
=

debug highlights:
=
R3#sh ccm mus
Current active multicast sessions : 1
 Multicast   RTP port   Packets   Call   CodecIncoming
 Address number in/outid  Interface
===
239.1.1.1 16384   1545/154595   g711ulaw  Vl502


R3#sh call active voice compact
   A/O FAX T Codec   typePeer Address   IP
R:
Total call-legs: 2
94 ANS T52g711ulawVOIPP2001
239.1.1.1:16384
95 ORG T52g711ulawTELEP4001
=

detailed debug attached.

Thanks!
Michael
version 12.4
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
!
hostname R3
!
boot-start-marker
boot-end-marker
!
card type e1 0 0
logging message-counter syslog
logging buffered 512000
enable secret 5 $1$mwEV$3uqLkfGBScOF9BIOcFFLd1
!
no aaa new-model
clock timezone HKT 8
network-clock-participate wic 0 
!
!
!
dot11 syslog
ip source-route
ip cef
!
!
ip dhcp excluded-address 142.102.66.1 142.102.66.9
ip dhcp excluded-address 142.102.66.31 142.102.66.254
!
ip dhcp pool CME
   network 142.102.66.0 255.255.255.0
   default-router 142.102.66.254 
   option 150 ip 142.102.66.254 
!
!
no ip domain lookup
ip multicast-routing 
no ipv6 cef
!
multilink bundle-name authenticated
! 
!
!
!
isdn switch-type primary-net5
!
voice-card 0
!
!
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
!
!
voice class h323 1
  h225 timeout tcp establish 3
  h225 timeout setup 3
!
! 
!
!
!
!
!
!
!
!
!
voice translation-rule 1
 rule 1 /^.*\(4...\)/ /\1/
!
voice translation-rule 2
 rule 1 /^852\(4...\)/ /\1/
!
voice translation-rule 11
 rule 1 /^4...$/ /2404&/ type any subscriber plan any isdn
!
voice translation-rule 12
 rule 1 /^4...$/ /8522404&/ type any international plan any isdn
!
voice translation-rule 21
 rule 1 // // type any subscriber plan any isdn
!
voice translation-rule 22
 rule 1 // // type any international plan any isdn
!
!
voice translation-profile gk_in
 translate called 1
!
voice translation-profile int_out
 translate calling 12
 translate called 22
!
voice translation-profile loc_out
 translate calling 11
 translate called 21
!
voice translation-profile pstn_in
 translate called 1
!
!
voip-incoming translation-profile gk_in
!
! 
! 
!
!
archive
 log config
  hidekeys
!
!
controller E1 0/0/0
 pri-group timeslots 1-6,16
 description PSTN 0/1/0
!
!
!
!
!
interface Loopback0
 ip address 142.1.66.254 255.255.255.255
!
interface GigabitEthernet0/0
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
!
interface FastEthernet0/2/0
 description SW G1/0/22 SiteC SoftPhone
 switchport access vlan 502
!
interface FastEthernet0/2/1
 description SiteC Phone1
 switchport access vlan 602
 switchport voice vlan 502
 spanning-tree portfast
!
interface FastEthernet0/2/2
 description SiteC Phone2
 switchport access vlan 602
 switchport voice vlan 502
 spanning-tree portfast
!
interface FastEthernet0/2/3
 shutdown
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn outgoing display-ie
 no cdp enable
!
interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 ip ospf mtu-ignore
 no fair-queue
 frame-relay lmi-type ansi
!
interface Serial0/1/0.301 point-to-point
 description DLCI 301 to R1 DLCI 103
 ip address 156.201.26.2 255.255.255.0
 frame-relay interface-dlci 301   
!
interface Service-Engine1/0
 no ip address
 shutdown
!
interface Vlan1
 no ip address
 shutdown
!
interface Vlan502
 ip address 142.102.66.254 255.255.255.0
 ip pim dense-mode
 h323-gateway voip interface
 h323-gateway voip id GK ipaddr 142.1.64.254 1719
 h323-gateway voip h323-id CME-HKG
 h323-gateway voip tech-prefix 852
 h323-gateway voip bind srcaddr 142.102.66.254
!
interface Vlan602
 ip address 142.202.66.254 255.255.255.0
! 
router ospf 1
 router-i

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

[Michael Luo]

Call flow as below:

ePhone -> CME (H.323 GW) -> GK -> GK-controlled Trunk -> CUCM -> UCM Phone

When UCM Phone press the "Hold" button, how the MOH stream flow to ePhone?

Would multicast MoH work in this scenario?

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

Re: [OSL | CCIE_Voice] CME phone hears beep-on-hold on PSTN call

[Michael Luo]

Could you elaborate more?

Did you mean I have to configure moh on the PSTN Router (PSTN / Frame-Relay
switch) instead of the CME router?

Thanks!

On Sun, Mar 20, 2011 at 10:20 AM, CCIE Voice  wrote:

> Sounds like your pstn router is not configured for moh.
>
> --
>
>
> On Mar 20, 2011, at 8:34, Michael Luo  wrote:
>
> > I was trying to configure a CME router to stream music on hold to its
> e-phones.  The CME router is also the voice gateway to PSTN.
> >
> > Here are the results:
> >
> > Scenario #1 Calls between e-phones.  e-phones can hear MOH.
> >
> > Scenario #2 Calls between e-phone and PSTN phones.  PSTN phones can hear
> MOH.
> >
> > Scenario #3 Calls between e-phone and PSTN phones.  e-phones hear
> beep-on-hold (beep-beep-beep).
> >
> > I was trying to troubleshoot Scenario #3.
> >
> > Here are the configure highlights:
> >
> > =
> > ip multicast-routing
> > !
> > interface vlan502
> >  description e-phone vlan
> >  ip address 142.102.66.254 255.255.255.0
> >  ip pim dense
> >
> > interface Loopback0
> >  ip address 142.1.66.254 255.255.255.255
> >
> > ccm-manager music-on-hold
> >
> > telephony-service
> >  ip source-address 142.102.66.254 port 2000
> >  moh music-on-hold.au
> >  multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
> > =
> >
> > Attached are the full router config and debug output.
> >
> > Any help is appreciated!
> >
> > Michael
> > 
> > 
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] CME phone hears beep-on-hold on PSTN call

[Michael Luo]

I was trying to configure a CME router to stream music on hold to its
e-phones.  The CME router is also the voice gateway to PSTN.

Here are the results:

Scenario #1 Calls between e-phones.  e-phones can hear MOH.

Scenario #2 Calls between e-phone and PSTN phones.  PSTN phones can hear
MOH.

Scenario #3 Calls between e-phone and PSTN phones.  e-phones hear
beep-on-hold (beep-beep-beep).

I was trying to troubleshoot Scenario #3.

Here are the configure highlights:

=
ip multicast-routing
!
interface vlan502
 description e-phone vlan
 ip address 142.102.66.254 255.255.255.0
 ip pim dense

interface Loopback0
 ip address 142.1.66.254 255.255.255.255

ccm-manager music-on-hold

telephony-service
 ip source-address 142.102.66.254 port 2000
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 142.1.66.254 142.102.66.254
=

Attached are the full router config and debug output.

Any help is appreciated!

Michael
version 12.4
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
!
hostname R3
!
boot-start-marker
boot-end-marker
!
card type e1 0 0
logging message-counter syslog
logging buffered 512000
enable secret 5 $1$mwEV$3uqLkfGBScOF9BIOcFFLd1
!
no aaa new-model
clock timezone HKT 8
network-clock-participate wic 0 
!
!
!
dot11 syslog
ip source-route
ip cef
!
!
ip dhcp excluded-address 142.102.66.1 142.102.66.9
ip dhcp excluded-address 142.102.66.31 142.102.66.254
!
ip dhcp pool CME
   network 142.102.66.0 255.255.255.0
   default-router 142.102.66.254 
   option 150 ip 142.102.66.254 
!
!
no ip domain lookup
ip multicast-routing 
no ipv6 cef
!
multilink bundle-name authenticated
! 
!
!
!
isdn switch-type primary-net5
!
voice-card 0
!
!
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
!
!
voice class h323 1
  h225 timeout tcp establish 3
  h225 timeout setup 3
!
! 
!
!
!
!
!
!
!
!
!
voice translation-rule 1
 rule 1 /^.*\(4...\)/ /\1/
!
voice translation-rule 2
 rule 1 /^852\(4...\)/ /\1/
!
voice translation-rule 11
 rule 1 /^4...$/ /2404&/ type any subscriber plan any isdn
!
voice translation-rule 12
 rule 1 /^4...$/ /8522404&/ type any international plan any isdn
!
voice translation-rule 21
 rule 1 // // type any subscriber plan any isdn
!
voice translation-rule 22
 rule 1 // // type any international plan any isdn
!
!
voice translation-profile gk_in
 translate called 1
!
voice translation-profile int_out
 translate calling 12
 translate called 22
!
voice translation-profile loc_out
 translate calling 11
 translate called 21
!
voice translation-profile pstn_in
 translate called 1
!
!
voip-incoming translation-profile gk_in
!
! 
! 
!
!
archive
 log config
  hidekeys
!
!
controller E1 0/0/0
 pri-group timeslots 1-6,16
 description PSTN 0/1/0
!
!
!
!
!
interface Loopback0
 ip address 142.1.66.254 255.255.255.255
!
interface GigabitEthernet0/0
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
!
interface FastEthernet0/2/0
 description SW G1/0/22 SiteC SoftPhone
 switchport access vlan 502
!
interface FastEthernet0/2/1
 description SiteC Phone1
 switchport access vlan 602
 switchport voice vlan 502
 spanning-tree portfast
!
interface FastEthernet0/2/2
 description SiteC Phone2
 switchport access vlan 602
 switchport voice vlan 502
 spanning-tree portfast
!
interface FastEthernet0/2/3
 shutdown
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn outgoing display-ie
 no cdp enable
!
interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 ip ospf mtu-ignore
 no fair-queue
 frame-relay lmi-type ansi
!
interface Serial0/1/0.301 point-to-point
 description DLCI 301 to R1 DLCI 103
 ip address 156.201.26.2 255.255.255.0
 frame-relay interface-dlci 301   
!
interface Service-Engine1/0
 no ip address
 shutdown
!
interface Vlan1
 no ip address
 shutdown
!
interface Vlan502
 ip address 142.102.66.254 255.255.255.0
 ip pim dense-mode
 h323-gateway voip interface
 h323-gateway voip id GK ipaddr 142.1.64.254 1719
 h323-gateway voip h323-id CME-HKG
 h323-gateway voip tech-prefix 852
 h323-gateway voip bind srcaddr 142.102.66.254
!
interface Vlan602
 ip address 142.202.66.254 255.255.255.0
! 
router ospf 1
 router-id 142.1.66.254
 log-adjacency-changes
 network 0.0.0.0 255.255.255.255 area 0
!
ip forward-protocol nd
ip http server
no ip http secure-server
!
!
!
!
!
!
!
!
tftp-server flash:Desktops/320x212x16/List.xml
tftp-server flash:Desktops/320x212x16/Sunrise.png
tftp-server flash:Desktops/320x212x16/Sunrise-TN.png
!
control-plane
!
! 
!
voice-port 0/0/0:15
 translation-profile incoming pstn_in
!
ccm-manager music-on-hold
!
mgcp fax t38 ecm
!
!
!
dial-peer voice 1 pots
 incoming called-number .
 direct-inward-dial
!
dial-peer v

[OSL | CCIE_Voice] What's the purpose of calling Type/Plan?

[Michael Luo]

We usually set the called party Type (Subscriber, National, International),
so provider and do appropriate routing.

What's the purpose of setting it on the calling party?

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

Re: [OSL | CCIE_Voice] Attendant Console hunt group member busy?

[Michael Luo]

Thanks for the info.  But it doesn't apply to my scenario.

1) I'm not getting fast busy.  I'm getting slow busy.

2) I didn't specify user/line pairs.  I specify DN only, which does NOT
require user login.

Thanks for your help.



On Tue, Feb 8, 2011 at 11:52 AM, ccieid1ot  wrote:

> Q. Why do I get a fast busy when I call the pilot point?
>
>  *A. *The pilot point and/or the controlled phones are not in the
> controlled device list of the 'ac' user. You must create a user with id 'ac'
> from the Cisco CallManager user administration page, and associate all pilot
> points and attendant phones with this user. Make sure that this user checks
> *Enable CTI Application Use*. The default password for this user should be
> *12345*. If you want to change the password, see How do I change the
> password for the 'ac' 
> user?<http://www.cisco.com/en/US/products/ps7282/products_qanda_item09186a00800b0a20.shtml#qa6>
>
>  Q. Why do I still get a fast busy when I call the pilot point after I
> have associated the devices with the ac user?
>
>  *A. *It can be that the user(s) specified in the user/line pair(s) in the
> hunt-group associated with the pilot point is not online. Log in and go
> online from the Cisco CallManager Attendant Console GUI. If this does not
> work, you may be running into Cisco bug ID 
> CSCdw86252<http://www.cisco.com/pcgi-bin/Support/Bugtool/onebug.pl?bugid=CSCdw86252>(
> registered <http://tools.cisco.com/RPF/register/register.do> customers
> only) - 'User Device Association is Lost Sometimes in Device Association
> page'. Go to the Cisco CallManager Administration pages and re-associate the
> relevant devices that are no longer associated with the user.
>
>
>
> On Tue, Feb 8, 2011 at 10:20 AM, Michael Luo  wrote:
>
>> Do we need to get users logged in if we chose device members?
>>
>> I thought it was only required when we choose "user member".
>>
>> see
>> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsccmac.html#wp1225737
>>
>> Maybe I misunderstood it?
>>
>> Thanks!
>> Michael
>>
>>
>>
>> On Tue, Feb 8, 2011 at 10:13 AM, ccieid1ot  wrote:
>>
>>> If you hear busy then none of the users are logged in to ac.
>>>
>>> duy
>>> ccie #27737 voice
>>>
>>> tmobile g2
>>> On Feb 8, 2011 10:00 AM, "Michael Luo"  wrote:
>>> > I'm using the native Attendant Console on CUCM 7.1.2. Information as
>>> below:
>>> >
>>> > Hunt Pilot: 5500
>>> >
>>> > Member: 5001, 1001
>>> >
>>> > Calling party: 5002
>>> >
>>> > Time: 02/07 18:06:58
>>> >
>>> >
>>> >
>>> > Symptom:
>>> >
>>> > 1) If I have "Queuing Enable" option checked, caller heard music.
>>> >
>>> > 2) If I have "Queuing Enable" option unchecked, caller heard busy.
>>> >
>>> >
>>> >
>>> > By looking at AC logs, it looks like AC thought members were busy. But
>>> in
>>> > fact, they weren't.
>>> >
>>> >
>>> >
>>> > I've tried restart Attendant Console service, restart CTIManager,
>>> restart
>>> > CUCM, reset phones. Symptom persists.
>>> >
>>> > In AC logs: "208[Thread-87]02/07 22:14:09.921 INFO Thread-87 >>>
>>> ACPilotRP:
>>> > 5500: CallID: 33633439 No lines active to route the call".
>>> >
>>> > Any pointer would be appreciated. Attached are the configuration and AC
>>> > logs.
>>> >
>>> > Thanks!
>>>
>>
>>
>
>
> --
> duy
> CCIE #27737 Voice
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Attendant Console hunt group member busy?

[Michael Luo]

Do we need to get users logged in if we chose device members?

I thought it was only required when we choose "user member".

see
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsccmac.html#wp1225737

Maybe I misunderstood it?

Thanks!
Michael


On Tue, Feb 8, 2011 at 10:13 AM, ccieid1ot  wrote:

> If you hear busy then none of the users are logged in to ac.
>
> duy
> ccie #27737 voice
>
> tmobile g2
> On Feb 8, 2011 10:00 AM, "Michael Luo"  wrote:
> > I'm using the native Attendant Console on CUCM 7.1.2. Information as
> below:
> >
> > Hunt Pilot: 5500
> >
> > Member: 5001, 1001
> >
> > Calling party: 5002
> >
> > Time: 02/07 18:06:58
> >
> >
> >
> > Symptom:
> >
> > 1) If I have "Queuing Enable" option checked, caller heard music.
> >
> > 2) If I have "Queuing Enable" option unchecked, caller heard busy.
> >
> >
> >
> > By looking at AC logs, it looks like AC thought members were busy. But in
> > fact, they weren't.
> >
> >
> >
> > I've tried restart Attendant Console service, restart CTIManager, restart
> > CUCM, reset phones. Symptom persists.
> >
> > In AC logs: "208[Thread-87]02/07 22:14:09.921 INFO Thread-87 >>>
> ACPilotRP:
> > 5500: CallID: 33633439 No lines active to route the call".
> >
> > Any pointer would be appreciated. Attached are the configuration and AC
> > logs.
> >
> > Thanks!
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Attendant Console goes into member's voicemail?

[Michael Luo]

I was doing IPExpert Vol2 Lab1 Task 8.2 regarding Attendant Console.  The
requirement was "If both 1002 and 5002 are busy, the caller should hear
music".

I followed the proctor guide to set up AC pilot point, member, queue size,
queue hold time.

However, if one of the members were busy, the call went into that member's
voicemail.  It's because on that DN, I have call forward busy to voicemail
set (per previous task's requirement).  So there's no way for the caller to
be queued and hear music.

Did I configure it the wrong way?

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

Re: [OSL | CCIE_Voice] How to troubleshoot CUBE call?

[Michael Luo]

Fixed it.

It was "Wait for Far End H.245 Terminal Capability Set" on CUCM needs to be
unchecked.  Any documentation explain why this was needed for CUBE?

Thanks!
Michael

On Tue, Feb 1, 2011 at 10:16 PM, Cristobal Priego  wrote:

> when you make the call, does it actually uses you dsp resources ?
>
> when you make a call, what is the codec used on the hq phone (i'm assuming
> it is g.729) and on cme it has to be g.711.
>
> when you hit the answer softkey you can issue on the cme router
> show sdsp sessions active
>
> and confirm that you're invoking your transcoder.
>
> on the cme router while the phone is ringing you can also use
>
> show voice active call brief or something like that and it will show the
> dial-peer used to confirm your call flow
>
> 2011/2/1 Michael Luo 
>
>> On HQ-RTR, debug voip ccapi inout:
>>
>> Feb  1 18:53:08.822 EST: //30/00AB25620700/CCAPI/ccCallDisconnect:
>>Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0,
>> Disconnect Cause=47)
>>
>> It looks like a codec issue.  But I have configure transcoder per the
>> proctor guide:
>>
>> HQ-RTR#sh sdspfarm units
>>
>> mtp-1 Device:hq-xcoder TCP socket:[1]  REGISTERED in SCCP ver 17/10
>> actual_stream:6 max_stream 6 IP:10.10.200.3  35686  MTP Dixieland
>> keepalive 151
>> Supported codec:
>>  G711Ulaw
>>  G711Alaw
>>  G729a
>>  G729ab
>>
>>  max-mtps:1, max-streams:6, alloc-streams:6, act-streams:0
>>
>> HQ-RTR#sh run | b teleph
>> telephony-service
>>  sdspfarm units 1
>>  sdspfarm transcode sessions 3
>>  sdspfarm tag 1 hq-xcoder
>>  max-ephones 1
>>  max-dn 1
>>  ip source-address 10.10.200.3 port 2000
>>
>> How does CUBE know where to look for XCoder resource?
>>
>> Thanks!
>> Michael
>>
>>
>>
>>
>>
>> On Tue, Feb 1, 2011 at 5:51 PM, Michael Luo  wrote:
>>
>>> I was doing IPExpert vol2 lab 1 task 4.2.
>>>
>>> When HQ(5001) calls BR2(3001), phone rings.  But when even if I pressed
>>> "answer" on BR2 phone, HQ phone kept ringing for a couple seconds, then
>>> disconnected.
>>>
>>> I've read some thread like
>>> http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17555.html.
>>> But it didn't see to apply to my case.
>>>
>>> What's the systematic way to debug this issue?  What debug commands we
>>> could use?
>>>
>>> Thank you very much!
>>> Michael
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] How to troubleshoot CUBE call?

[Michael Luo]

On HQ-RTR, debug voip ccapi inout:

Feb  1 18:53:08.822 EST: //30/00AB25620700/CCAPI/ccCallDisconnect:
   Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0,
Disconnect Cause=47)

It looks like a codec issue.  But I have configure transcoder per the
proctor guide:

HQ-RTR#sh sdspfarm units

mtp-1 Device:hq-xcoder TCP socket:[1]  REGISTERED in SCCP ver 17/10
actual_stream:6 max_stream 6 IP:10.10.200.3  35686  MTP Dixieland keepalive
151
Supported codec:
 G711Ulaw
 G711Alaw
 G729a
 G729ab

 max-mtps:1, max-streams:6, alloc-streams:6, act-streams:0

HQ-RTR#sh run | b teleph
telephony-service
 sdspfarm units 1
 sdspfarm transcode sessions 3
 sdspfarm tag 1 hq-xcoder
 max-ephones 1
 max-dn 1
 ip source-address 10.10.200.3 port 2000

How does CUBE know where to look for XCoder resource?

Thanks!
Michael




On Tue, Feb 1, 2011 at 5:51 PM, Michael Luo  wrote:

> I was doing IPExpert vol2 lab 1 task 4.2.
>
> When HQ(5001) calls BR2(3001), phone rings.  But when even if I pressed
> "answer" on BR2 phone, HQ phone kept ringing for a couple seconds, then
> disconnected.
>
> I've read some thread like
> http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17555.html.
> But it didn't see to apply to my case.
>
> What's the systematic way to debug this issue?  What debug commands we
> could use?
>
> Thank you very much!
> Michael
>
___
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[OSL | CCIE_Voice] How to troubleshoot CUBE call?

[Michael Luo]

I was doing IPExpert vol2 lab 1 task 4.2.

When HQ(5001) calls BR2(3001), phone rings.  But when even if I pressed
"answer" on BR2 phone, HQ phone kept ringing for a couple seconds, then
disconnected.

I've read some thread like
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17555.html.
But it didn't see to apply to my case.

What's the systematic way to debug this issue?  What debug commands we could
use?

Thank you very much!
Michael
___
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Re: [OSL | CCIE_Voice] IE Delivery

[Michael Luo]

You are sending calling name to 911

Display i = 'HQ Phone 1 2001'

Didn't they see it?

On Fri, Jan 28, 2011 at 6:19 AM, Ccie Voice  wrote:

> Hi this is the debug of isdn q931 ( Call form the IP phone to PSTN)
>
>
>  Jan 28 04:13:38.179: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref =
> 0x0002
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98383
> Exclusive, Channel 3
> Display i = 'HQ Phone 1 2001'
> Calling Party Number i = 0x0081, '2001'
> Plan:Unknown, Type:Unknown
> Called Party Number i = 0x80, '911'
> Plan:Unknown, Type:Unknown
> Jan 28 04:13:38.203: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref
> = 0x8002
> Channel ID i = 0xA98383
> Exclusive, Channel 3
> HQ-R1#
> Jan 28 04:13:38.207: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8  callref =
> 0x8002
> Progress Ind i = 0x8188 - In-band info or appropriate now available
>
> HQ-R1#
> Jan 28 04:13:49.831: ISDN Se0/0/0:23 Q931: RX <- CONNECT pd = 8  callref =
> 0x8002
> Jan 28 04:13:49.835: ISDN Se0/0/0:23 Q931: TX -> CONNECT_ACK pd = 8
> callref = 0x0002
> =
> =
>
>
> and this the debug from PSTN to IP phone ( I can see in the IP phone form
> Emergency Services)
>
> Jan 28 04:17:42.003: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref =
> 0x0087
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA18381
> Preferred, Channel 1
> Facility i =
> 0x9F8B0100A11A0201090201008012456D657267656E6379205365727669636573
> Protocol Profile =  Networking Extensions
> 0xA11A0201090201008012456D657267656E6379205365727669636573
> Component = Invoke component
> Invoke Id = 9
> Operation = CallingName
> Name Presentation Allowed Extended
> Name = Emergency Services
> Progress Ind i = 0x8583 - Origination address is non-ISDN
> HQ-R1#
> Display i = 'Emergency Services'
> Calling Party Number i = 0x0080, '911'
> Plan:Unknown, Type:Unknown
> Called Party Number i = 0xA1, '7752012001'
> Plan:ISDN, Type:National
> Jan 28 04:17:42.039: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref
> = 0x8087
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> Jan 28 04:17:42.039: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8  callref =
> 0x8087
> Progress Ind i = 0x8088 - In-band info or appropriate now available
>
> HQ-R1#
> Jan 28 04:18:05.275: ISDN Se0/0/0:23 Q931: TX -> CONNECT pd = 8  callref =
> 0x8087
> Display i = 'HQ Phone 1 2001'
> Jan 28 04:18:05.283: ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8
> callref = 0x0087
>
>
>
>
> Regards,
>
> --
> *From:* Michael Luo 
> *To:* Ccie Voice 
> *Cc:* CCIE Study 
> *Sent:* Fri, January 28, 2011 6:57:58 AM
> *Subject:* Re: [OSL | CCIE_Voice] IE Delivery
>
> You may use "debug isdn q931" to review IE.
>
> For "Redirecting IE", you must need to set call forwarding on the phone.
>
> Michael
>
> On Thu, Jan 27, 2011 at 7:19 PM, Ccie Voice  wrote:
>
>> Hi All,
>>
>> I am  sorry for this silly question but I really need your help to
>> understand when I should click these check boxes and how to test the
>> functionality:
>>
>> Display IE Delivery
>> Redirecting Number IE Delivery-Outbound
>>
>> Regards,
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] IE Delivery

[Michael Luo]

If you run "debug isdn q931" on the MGCP gateway, what do you see?

On Fri, Jan 28, 2011 at 5:00 AM, Ccie Voice  wrote:

> Thank you Shrini and Micheal for your reply,
>
> Actually I tried to test it before asking.
>
> I have the following setup:
>
> Phone registered to CUCM >> MGCP Gateway >> connected to PSTN via T1.
>
> I checked the display IE but the same result I am not receiving the Calling
> Name.
>
> Regards,
>
>
> --
> *From:* Shrini 
> *To:* Ccie Voice ; CCIE Study <
> ccie_voice@onlinestudylist.com>
> *Sent:* Fri, January 28, 2011 6:51:29 AM
> *Subject:* RE: [OSL | CCIE_Voice] IE Delivery
>
> Nothing is silly here :-)
>
> Display ie - means when a user calls PSTN line it will display the calling
> party name. Same applies to redirecting when a call is redirected to PSTN.
>
> To test:
>
> assign name to a local phone. make a call to PSTN and see the difference
> checking and unchecking. For redirecting number you may use call forward
> feature or cfur on the line profile page.
>
> Thanks
> Shrini
>
>  --
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ccie Voice
> *Sent:* Thursday, January 27, 2011 5:20 PM
> *To:* CCIE Study
> *Subject:* [OSL | CCIE_Voice] IE Delivery
>
>  Hi All,
>
> I am  sorry for this silly question but I really need your help to
> understand when I should click these check boxes and how to test the
> functionality:
>
> Display IE Delivery
> Redirecting Number IE Delivery-Outbound
>
> Regards,
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] AAR to voicemail?

[Michael Luo]

I got a quick question.

Phone A calls phone B and ring no answer to voicemail.  Due to insufficient
bandwidth on the WAN, AAR was triggered to get to the voicemail pilot.

In this case, which CSS is taking effect?

A) Phone A's AAR CSS?
B) Phone B's Call Forwarding CSS?
C) Phone B's AAR CSS?

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

Re: [OSL | CCIE_Voice] IE Delivery

[Michael Luo]

You may use "debug isdn q931" to review IE.

For "Redirecting IE", you must need to set call forwarding on the phone.

Michael

On Thu, Jan 27, 2011 at 7:19 PM, Ccie Voice  wrote:

> Hi All,
>
> I am  sorry for this silly question but I really need your help to
> understand when I should click these check boxes and how to test the
> functionality:
>
> Display IE Delivery
> Redirecting Number IE Delivery-Outbound
>
> Regards,
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] RSVP reservation bandwidth

[Michael Luo]

Are you sure we can use "show voice call status" to show a IP-to-IP call?  I
doubt it.

On Thu, Jan 27, 2011 at 1:17 PM, Tanner Ezell wrote:

> My apologies, I missed that part of the email. 24k sounds correct for
> G.729. Still, verify with the show voice call status command what
> codec is being used.
>
> On Thu, Jan 27, 2011 at 11:07 AM, Michael Luo  wrote:
> > Well if it was set to G.711, why reservation bandwidth would be 24k after
> > the call was connected?  (see item B).
> >
> > Thanks!
> > Michael
> >
> > On Thu, Jan 27, 2011 at 12:37 PM, Tanner Ezell 
> > wrote:
> >>
> >> Well, most likely the region settings are set incorrectly and the call
> >> is going across as G.711. Try issuing a "show voice call status" on
> >> the gateway while the call is in progress to verify
> >>
> >> On Thu, Jan 27, 2011 at 8:00 AM, Michael Luo  wrote:
> >> > I was doing IPexpert Vol 1 lab 10.1 - RSVP bandwidth reservation.
> >> >
> >> > I configured RSVP agent for HQ and BR1 routers.  Then I made test
> calls
> >> > from
> >> > HQ to BR1.  Per the region settings, it should use G.729 for the call.
> >> >
> >> > I use "show ip rsvp reservation" command while making the call.  I
> >> > noticed
> >> > that:
> >> >
> >> > A) When the call is in ringing state the reservation is 96k.  If I
> >> > configure
> >> > the reservation to be lower than 96k, the call got routed via AAR.
> >> > B) When the call is in connected state, the reservation is 24k.
> >> >
> >> > I have question on item A above.  Per the proctor guide and CUCM SRND,
> >> > item
> >> > A should be 40k (for G.729) calls.  Why was it 96K in my lab?
> >> >
> >> > Thanks!
> >> > Michael
> >> >
> >> > ___
> >> > For more information regarding industry leading CCIE Lab training,
> >> > please
> >> > visit www.ipexpert.com
> >> >
> >> >
> >
> >
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] RSVP reservation bandwidth

[Michael Luo]

Well if it was set to G.711, why reservation bandwidth would be 24k after
the call was connected?  (see item B).

Thanks!
Michael

On Thu, Jan 27, 2011 at 12:37 PM, Tanner Ezell wrote:

> Well, most likely the region settings are set incorrectly and the call
> is going across as G.711. Try issuing a "show voice call status" on
> the gateway while the call is in progress to verify
>
> On Thu, Jan 27, 2011 at 8:00 AM, Michael Luo  wrote:
> > I was doing IPexpert Vol 1 lab 10.1 - RSVP bandwidth reservation.
> >
> > I configured RSVP agent for HQ and BR1 routers.  Then I made test calls
> from
> > HQ to BR1.  Per the region settings, it should use G.729 for the call.
> >
> > I use "show ip rsvp reservation" command while making the call.  I
> noticed
> > that:
> >
> > A) When the call is in ringing state the reservation is 96k.  If I
> configure
> > the reservation to be lower than 96k, the call got routed via AAR.
> > B) When the call is in connected state, the reservation is 24k.
> >
> > I have question on item A above.  Per the proctor guide and CUCM SRND,
> item
> > A should be 40k (for G.729) calls.  Why was it 96K in my lab?
> >
> > Thanks!
> > Michael
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> >
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] RSVP reservation bandwidth

[Michael Luo]

I was doing IPexpert Vol 1 lab 10.1 - RSVP bandwidth reservation.

I configured RSVP agent for HQ and BR1 routers.  Then I made test calls from
HQ to BR1.  Per the region settings, it should use G.729 for the call.

I use "show ip rsvp reservation" command while making the call.  I noticed
that:

A) When the call is in ringing state the reservation is 96k.  If I configure
the reservation to be lower than 96k, the call got routed via AAR.
B) When the call is in connected state, the reservation is 24k.

I have question on item A above.  Per the proctor guide and CUCM SRND, item
A should be 40k (for G.729) calls.  Why was it 96K in my lab?

Thanks!
Michael
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] B-ACD not playing welcome prompt

[Michael Luo]

Ok, I got it fixed but don't quite understand why it worked this way.

The incoming call was from a SIP trunk to CME.  With dialed number = 3500,
there could be two potential matched dial-peers:

dial-peer voice 3000 voip
 voice-class codec 1
 incoming called-number 3...$
 *dtmf-relay rtp-nte*
!
dial-peer voice 222 voip
 service aa
 destination-pattern 3500
 session target ipv4:10.10.110.3
 incoming called-number 3500
 *dtmf-relay h245-alphanumeric*
 codec g711ulaw
 no vad

Since the B-ACD was complaining about DTMF-Relay, I thought it might be the
DTMF-Relay settings on different dial-peers (3000 and 222).  So I tried the
following:

1) I changed dial-peer 3000.  Set dtmf-relay = h245-alphanumeric.  This
doesn't work.  I guess for SIP, we'll have to use rtp-nte?  So I revert it
back to dtmf-relay = rtp-nte.

2) I changed dial-peer 222.  Set ftmf-relay - rtp-nte.  This fixed the
problem.

My questions are:

1) From "debug voip dialpeer", I don't see dial-peer 3000 being matched at
all.  So I guess dial-peer 3000 should be out of the picture?

2) For H323 dial-peers, what's the effect of using rtp-nte instead of
h245-alphanumeric?

3) How does B-ACD script detect dtmf issue immediately (before user got a
chance to actually press any key)?

Thanks!

Michael




On Tue, Jan 25, 2011 at 10:10 AM, Michael Luo  wrote:

> Update: if the caller was one of the CME phone, it works fine.  If the
> caller is external, it doesn't work.  So how do we troubleshoot this error?
>
>
> "%IVR-3-APP_ERR: TCL AA: +++ Enable DTMF Relay in dialpeer +++"
>
> Thanks!
> Michael
>
>
> On Tue, Jan 25, 2011 at 9:50 AM, Michael Luo  wrote:
>
>> I was testing B-ACD (IpExpert Vol1 lab 9.3).  This is the 2nd round I
>> practice this lab.  The first round works just fine but not the 2nd round.
>>
>> The symptom was: B-ACD didn't play welcome prompt and play music on hold
>> directly.  Debug messages says:
>>
>> Jan 25 16:36:33.151 CET: //22//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing
>> Welcome Prompt and options menu ++
>> Jan 25 16:36:33.371 CET: %IVR-3-APP_ERR: TCL AA: +++ Enable DTMF Relay in
>> dialpeer +++
>>
>> I do have DTMF Relay in dialpeer.  So I guess it's a misleading message
>> that actually indicate the digit-collection process didn't get any digit
>> input.
>>
>> I tried to wipe out the config and re-load my first-round config (which
>> worked before).  I got the same problem.
>>
>> Attached are the router config and debug output.
>>
>> Any hints would be appreciated!
>>
>> Michael
>>
>
>
___
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Re: [OSL | CCIE_Voice] B-ACD not playing welcome prompt

[Michael Luo]

Update: if the caller was one of the CME phone, it works fine.  If the
caller is external, it doesn't work.  So how do we troubleshoot this error?

"%IVR-3-APP_ERR: TCL AA: +++ Enable DTMF Relay in dialpeer +++"

Thanks!
Michael

On Tue, Jan 25, 2011 at 9:50 AM, Michael Luo  wrote:

> I was testing B-ACD (IpExpert Vol1 lab 9.3).  This is the 2nd round I
> practice this lab.  The first round works just fine but not the 2nd round.
>
> The symptom was: B-ACD didn't play welcome prompt and play music on hold
> directly.  Debug messages says:
>
> Jan 25 16:36:33.151 CET: //22//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing
> Welcome Prompt and options menu ++
> Jan 25 16:36:33.371 CET: %IVR-3-APP_ERR: TCL AA: +++ Enable DTMF Relay in
> dialpeer +++
>
> I do have DTMF Relay in dialpeer.  So I guess it's a misleading message
> that actually indicate the digit-collection process didn't get any digit
> input.
>
> I tried to wipe out the config and re-load my first-round config (which
> worked before).  I got the same problem.
>
> Attached are the router config and debug output.
>
> Any hints would be appreciated!
>
> Michael
>
___
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www.ipexpert.com

[OSL | CCIE_Voice] B-ACD not playing welcome prompt

[Michael Luo]

I was testing B-ACD (IpExpert Vol1 lab 9.3).  This is the 2nd round I
practice this lab.  The first round works just fine but not the 2nd round.

The symptom was: B-ACD didn't play welcome prompt and play music on hold
directly.  Debug messages says:

Jan 25 16:36:33.151 CET: //22//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing
Welcome Prompt and options menu ++
Jan 25 16:36:33.371 CET: %IVR-3-APP_ERR: TCL AA: +++ Enable DTMF Relay in
dialpeer +++

I do have DTMF Relay in dialpeer.  So I guess it's a misleading message that
actually indicate the digit-collection process didn't get any digit input.

I tried to wipe out the config and re-load my first-round config (which
worked before).  I got the same problem.

Attached are the router config and debug output.

Any hints would be appreciated!

Michael
BR2-RTR#sh run
Building configuration...


Current configuration : 9089 bytes
!
! Last configuration change at 16:34:40 CET Tue Jan 25 2011
! NVRAM config last updated at 16:34:35 CET Tue Jan 25 2011
!
version 12.4
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
!
hostname BR2-RTR
!
boot-start-marker
boot-end-marker
!
card type e1 0 0
logging message-counter syslog
logging buffered 512000
enable secret 5 $1$mwEV$3uqLkfGBScOF9BIOcFFLd1
!
no aaa new-model
clock timezone CET 1
clock summer-time CET recurring 1 Sun Apr 1:00 last Sun Oct 1:00
network-clock-participate wic 0 
!
!
!
dot11 syslog
ip source-route
ip cef
!
!
ip dhcp excluded-address 10.10.202.1 10.10.202.49
ip dhcp excluded-address 10.10.202.70 10.10.202.254
!
ip dhcp pool CME
   network 10.10.202.0 255.255.255.0
   option 150 ip 10.10.110.3 
   default-router 10.10.202.1 
!
!
no ip domain lookup
no ipv6 cef
!
multilink bundle-name authenticated
! 
!
!
!
isdn switch-type primary-net5
!
voice-card 0
!
!
!
voice service voip 
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol cisco 
 h323
  no call service stop
 sip
  bind control source-interface Vlan400
  bind media source-interface Vlan400
  registrar server expires max 600 min 60
!
!
! 
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
!
!
voice class h323 1
  h225 timeout tcp establish 3
!
!
!
!
!
!
!
!
!
!
voice register global
 mode cme
 source-address 10.10.202.1 port 5060
 max-dn 2 
 max-pool 2
 load 7960-7940 P0S3-8-12-00
 authenticate register
 time-format 24
 date-format D/M/Y
 tftp-path flash:
 create profile sync 349653424358
!
voice register dn  1
 number 3002
 name BR2 Phone2
 auto-answer
!
voice register dn  2
 number 3004
 name BR2 SIP
!
voice register template  1
 dialplan 1
 no conference enable
!
voice register dialplan  1
 type 7940-7960-others
 pattern 1 3...
 pattern 2 999
!
voice register pool  1
 id mac 001F.6C7F.9C7B
 type 7960
 number 1 dn 1
 template 1
 dtmf-relay rtp-nte
 username 3002 password cisco
 description 3214-3002
 codec g711ulaw
!
voice register pool  2
 id mac AABB.CCDD.EEFF
 type 7960
 number 1 dn 2
 template 1
 dtmf-relay rtp-nte
 username 3004 password cisco
 description 3214-3004
 codec g711ulaw
! 
voice hunt-group 1 parallel
 list 3001,3005
 pilot 3210 
!
!
voice logout-profile 1
 number 3002 type normal
!
voice user-profile 1
 max-idle-time 10
 pin 1234
 user br2ph3 password admin
 number 3102 type normal
 speed-dial 1 3005 label "3005" 
!
!
voice translation-rule 1
 rule 1 /^\(3...\)$/ /3214\1/
!
voice translation-rule 2
 rule 1 /^32143/ /3/
 rule 2 /^\+3432143/ /3/
! 
voice translation-rule 3
 rule 1 /^\(3...\)$/ /343214\1/
!
voice translation-rule 3500
 rule 1 /^5\(3...\)$/ /\1/
!
!
voice translation-profile 10digitANI
 translate calling 3
!
voice translation-profile 4digitDNIS
 translate called 2
!
voice translation-profile 8digitANI
 translate calling 1
!
voice translation-profile BACD
 translate redirect-called 3500
!
!
!
application
 service aa flash:bacdprompts/app-b-acd-aa-3.0.0.2.tcl
  paramspace english index 1
  param number-of-hunt-grps 2
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 3500
  paramspace english location flash://bacdprompts/
  param second-greeting-time 60
  param welcome-prompt _bacd_welcome.au
  param queue-manager-debugs 1
  param call-retry-timer 15
  param voice-mail 3002
  param max-time-call-retry 30
  param service-name queue
 !
 service aa-drop flash:bacdprompts/app-b-acd-aa-3.0.0.2.tcl
  paramspace english index 1
  param service-name queue
  param drop-through-option 2
  paramspace english language en
  param second-greeting-time 60
  param max-time-vm-retry 2
  param max-time-call-retry 30
  param voice-mail 3002
  paramspace english location flash://bacdprompts/
  param aa-pilot 3501
  param number-of-hunt-grps 1
  param handoff-string aa-drop
  param call-retry-timer 15
 !
 service queue flash:bacdprompts/app-b-acd-3.0.0.2.tcl
  param queue-len 

Re: [OSL | CCIE_Voice] SRST address not populated to SIP phone

[Michael Luo]

Never mind.

I was trying to save some typing and that was the problem.  In device pool,
I specified SRST Reference to "Use Default Gateway".  Obviously, this works
for SCCP but not SIP.

The "right" way is to create a SRST reference explicitly.

Michael

On Sun, Jan 23, 2011 at 6:34 PM, Michael Luo  wrote:

> I was testing SRST for MGCP gateway.  I configured SRST reference in device
> pool.  But it populate to SCCP phones only, not SIP phones (7960).
>
> If I review the settings from SIP phone itself (Settings > Network
> Configuration), I can only find CM Pub and Sub.  "CallManager 3" was blank.
>
> I reviewed the configuration file from CUCM TFTP (SIP.cnf).
> It's like below:
>
> proxy1_address : "USECALLMANAGER"
> call_manager1_addr : "10.10.210.11"
> call_manager1_sip_port : 5060
> proxy2_address : "USECALLMANAGER"
> call_manager2_addr : "10.10.210.10"
> call_manager2_sip_port : 5060
> call_manager3_addr : ""
>
> I guess the "call_manager3_addr" should be populated with the SRST router
> address?
>
> I disable TFTP cache, restarted TFTP service, reset the phone.  Still can't
> get SRST address populated to SIP phone.  Any suggestions?
>
> Thanks!
> Michael
>
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[OSL | CCIE_Voice] SRST address not populated to SIP phone

[Michael Luo]

I was testing SRST for MGCP gateway.  I configured SRST reference in device
pool.  But it populate to SCCP phones only, not SIP phones (7960).

If I review the settings from SIP phone itself (Settings > Network
Configuration), I can only find CM Pub and Sub.  "CallManager 3" was blank.

I reviewed the configuration file from CUCM TFTP (SIP.cnf).
It's like below:

proxy1_address : "USECALLMANAGER"
call_manager1_addr : "10.10.210.11"
call_manager1_sip_port : 5060
proxy2_address : "USECALLMANAGER"
call_manager2_addr : "10.10.210.10"
call_manager2_sip_port : 5060
call_manager3_addr : ""

I guess the "call_manager3_addr" should be populated with the SRST router
address?

I disable TFTP cache, restarted TFTP service, reset the phone.  Still can't
get SRST address populated to SIP phone.  Any suggestions?

Thanks!
Michael
___
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Re: [OSL | CCIE_Voice] IPExpert's PSTN configuration problem

[Michael Luo]

This is by purpose.  The goal is to block unwanted numbers.

For example, if you're at BR1, you should dial 7-digit local number
"863" instead of 10 or 11 digit numbers (like 617-863- or
1-617-863-).

This is to simulate some telco policies in real world and force you apply
different techniques to manipulate the numbers (like route pattern, route
list, route group, translation pattern, etc.).

Michael



On Sun, Jan 23, 2011 at 11:42 AM, George Goglidze wrote:

> Hi all,
>
> I'm doing right now an online session of IPExpert, and ran into this
> problem.
> on PSTN router, on incoming voice-port they have a translation pattern
> which translates all numbers to 1234!!! thus call failing.
>
> Is it me? Maybe I loaded wrong configuration of PSTN router,  because it's
> just not going to work like this.
>
> thanks,
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] AAR display

[Michael Luo]

I have ""isdn outgoing display-ie" on the D channel.  H323 gateway sent out
the caller ID.  But it doesn't seem to send the "redirect party number".
Not sure why.

2011/1/22 Roger Källberg 

>   Hi Michael,
>
> It most certainly also applies to H323 GW's, but you must also add "isdn
> outgoing display-ie" to the serial interface for the D channel.
>
> Sincerely
>
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
> Eric Perssons väg 21, SE-217 62 MALMÖ
>
>  --
> *Från:* Michael Luo [hout...@gmail.com]
> *Skickat:* den 22 januari 2011 01:57
> *Till:* ccie_voice@onlinestudylist.com
> *Ämne:* Re: [OSL | CCIE_Voice] AAR display
>
>  Never mind.  It's the "redirect IE" checkbox on outbound gateway.
> Interesting thing was, this only applies to MGCP but not H323 gateway.
>
> On Fri, Jan 21, 2011 at 5:56 PM, Michael Luo  wrote:
>
>> By the way, in CCM trace:
>>
>> 01/21/2011 16:48:03.289 CCM|StationD:(009) (2,100,9,20) CallInfo
>> callingPartyName='BR1 Phone1' callingParty=8631001 cgpnVoiceMailbox=
>> alternateCallingParty=  calledPartyName=''
>> calledParty=12123945001 cdpnVoiceMailbox=
>> originalCalledPartyName='' originalCalledParty=5001
>> originalCdpnVoiceMailbox= originalCdpnRedirectReason=1
>> lastRedirectingPartyName='' 
>> *lastRedirectingParty=5001*lastRedirectingVoiceMailbox=
>> *lastRedirectingReason=1* callType=2(OutBound) lineInstance=1
>> callReference=48742317. version:
>> 85720011|> Transition>
>>
>> I think RedirectingReason=1 means "Call Forward Busy".  But how?  I tried
>> to call different phones in HQ site.  They all display the same thing.
>>
>>
>> On Fri, Jan 21, 2011 at 5:54 PM, Michael Luo  wrote:
>>
>>> I was doing IPExpert volume 1 lab 6 task 6.1 - AAR (Automated Alternate
>>> Routing).
>>>
>>> Per the proctor guide, the configuration was pretty simple.  I got it
>>> worked but observer strange behaviors.
>>>
>>> When HQ Phone1(5001) calls BR1 Phone1(1001) via AAR, the display on BR1
>>> Phone1 was "From HQ Phone1", which was the expected behavior.
>>>
>>> But BR1 calls HQ, the display on HQ Phone1's screen was "Forward BR1
>>> Phone1 For HQ Phone1".  The call did go through.  But I was curious why the
>>> display says "Forward".
>>>
>>> Any recommendation where I shall check?
>>>
>>> Thanks!
>>> Michael
>>>
>>
>>
>
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Re: [OSL | CCIE_Voice] AAR display

[Michael Luo]

Never mind.  It's the "redirect IE" checkbox on outbound gateway.
Interesting thing was, this only applies to MGCP but not H323 gateway.

On Fri, Jan 21, 2011 at 5:56 PM, Michael Luo  wrote:

> By the way, in CCM trace:
>
> 01/21/2011 16:48:03.289 CCM|StationD:(009) (2,100,9,20) CallInfo
> callingPartyName='BR1 Phone1' callingParty=8631001 cgpnVoiceMailbox=
> alternateCallingParty=  calledPartyName=''
> calledParty=12123945001 cdpnVoiceMailbox=
> originalCalledPartyName='' originalCalledParty=5001
> originalCdpnVoiceMailbox= originalCdpnRedirectReason=1
> lastRedirectingPartyName='' 
> *lastRedirectingParty=5001*lastRedirectingVoiceMailbox=
> *lastRedirectingReason=1* callType=2(OutBound) lineInstance=1
> callReference=48742317. version:
> 85720011| Transition>
>
> I think RedirectingReason=1 means "Call Forward Busy".  But how?  I tried
> to call different phones in HQ site.  They all display the same thing.
>
>
> On Fri, Jan 21, 2011 at 5:54 PM, Michael Luo  wrote:
>
>> I was doing IPExpert volume 1 lab 6 task 6.1 - AAR (Automated Alternate
>> Routing).
>>
>> Per the proctor guide, the configuration was pretty simple.  I got it
>> worked but observer strange behaviors.
>>
>> When HQ Phone1(5001) calls BR1 Phone1(1001) via AAR, the display on BR1
>> Phone1 was "From HQ Phone1", which was the expected behavior.
>>
>> But BR1 calls HQ, the display on HQ Phone1's screen was "Forward BR1
>> Phone1 For HQ Phone1".  The call did go through.  But I was curious why the
>> display says "Forward".
>>
>> Any recommendation where I shall check?
>>
>> Thanks!
>> Michael
>>
>
>
___
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Re: [OSL | CCIE_Voice] AAR display

[Michael Luo]

By the way, in CCM trace:

01/21/2011 16:48:03.289 CCM|StationD:(009) (2,100,9,20) CallInfo
callingPartyName='BR1 Phone1' callingParty=8631001 cgpnVoiceMailbox=
alternateCallingParty=  calledPartyName=''
calledParty=12123945001 cdpnVoiceMailbox=
originalCalledPartyName='' originalCalledParty=5001
originalCdpnVoiceMailbox= originalCdpnRedirectReason=1
lastRedirectingPartyName=''
*lastRedirectingParty=5001*lastRedirectingVoiceMailbox=
*lastRedirectingReason=1* callType=2(OutBound) lineInstance=1
callReference=48742317. version:
85720011|

I think RedirectingReason=1 means "Call Forward Busy".  But how?  I tried to
call different phones in HQ site.  They all display the same thing.

On Fri, Jan 21, 2011 at 5:54 PM, Michael Luo  wrote:

> I was doing IPExpert volume 1 lab 6 task 6.1 - AAR (Automated Alternate
> Routing).
>
> Per the proctor guide, the configuration was pretty simple.  I got it
> worked but observer strange behaviors.
>
> When HQ Phone1(5001) calls BR1 Phone1(1001) via AAR, the display on BR1
> Phone1 was "From HQ Phone1", which was the expected behavior.
>
> But BR1 calls HQ, the display on HQ Phone1's screen was "Forward BR1 Phone1
> For HQ Phone1".  The call did go through.  But I was curious why the display
> says "Forward".
>
> Any recommendation where I shall check?
>
> Thanks!
> Michael
>
___
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[OSL | CCIE_Voice] AAR display

[Michael Luo]

I was doing IPExpert volume 1 lab 6 task 6.1 - AAR (Automated Alternate
Routing).

Per the proctor guide, the configuration was pretty simple.  I got it worked
but observer strange behaviors.

When HQ Phone1(5001) calls BR1 Phone1(1001) via AAR, the display on BR1
Phone1 was "From HQ Phone1", which was the expected behavior.

But BR1 calls HQ, the display on HQ Phone1's screen was "Forward BR1 Phone1
For HQ Phone1".  The call did go through.  But I was curious why the display
says "Forward".

Any recommendation where I shall check?

Thanks!
Michael
___
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Re: [OSL | CCIE_Voice] GK-Controlled Trunk in CUCM

[Michael Luo]

Hi Roger,

I was hoping the same as you.  But I couldn't figure out who to get it
worked without adding the H323 GW (CME) to CUCM.

>From my observation, adding H323 GW (CME) to CUCM is needed because:

1) For security reason, CUCM won't accept H323 connection from an "unknown"
source (ie. not configured in CUCM).
2) Without the GW configured in CUCM, there's no way to configure in-bound
call parameters, such as Calling Search Space, Significant Digits, etc.

So I guess without the GW being configured on CUCM, you can make outbound
calls (from CUCM to CME).  But you cannot make inbound calls (from CME to
CUCM).

Michael

On Mon, Jan 17, 2011 at 4:24 PM, Roger Carpio wrote:

> Michael,
>
> You do not need to add the GW to CUCM; otherwise this will not be scalable
> at all. I had the same confusion but then I got it to work without adding my
> GW (CME) to CUCM.
>
> Regards,
> Roger Carpio.
>
> On Mon, Jan 17, 2011 at 2:16 PM, Michael Luo  wrote:
>
>> I was trying to understand the "Gatekeeper-controlled Trunk" in CUCM.
>>
>> If I configured the "Gatekeeper-controlled Trunk", do I still have to
>> configured the H323 gateways?
>>
>> Consider this scenario:
>>
>> GW --- GK --- CUCM
>>
>> If there's a call from GW to CUCM, my understanding was:
>>
>> 1) GW sends ARQ to GK.
>> 2) GK resolves the called number and sends ACF to GW.
>> 3) GW will try to communicate with CUCM.
>>
>> In step 3, if GW was not configured in CUCM, the call will fail.
>>
>> Did I understand this wrong?
>>
>> Thanks!
>> Michael
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
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[OSL | CCIE_Voice] GK-Controlled Trunk in CUCM

[Michael Luo]

I was trying to understand the "Gatekeeper-controlled Trunk" in CUCM.

If I configured the "Gatekeeper-controlled Trunk", do I still have to
configured the H323 gateways?

Consider this scenario:

GW --- GK --- CUCM

If there's a call from GW to CUCM, my understanding was:

1) GW sends ARQ to GK.
2) GK resolves the called number and sends ACF to GW.
3) GW will try to communicate with CUCM.

In step 3, if GW was not configured in CUCM, the call will fail.

Did I understand this wrong?

Thanks!
Michael
___
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Re: [OSL | CCIE_Voice] CME: CFNA to CUE got fast busy

[Michael Luo]

I think I found the answer here:
http://uc500.com/external-access-voicemail-over-sip-trunk

On Sat, Jan 15, 2011 at 6:21 PM, Michael Luo  wrote:

> I'm running CME 7.1 and CUE 7.0.6 on a 3825 router with IOS 12.4(24)T3.
>
> I have SCCP phone and SIP phone reigstered to CME.
>
> I can call the VoiceMail pilot (3600) from both phones and get the welcome
> prompt.
>
> Call from SCCP phone (3001) to SIP phone (3002) and
> "Call-Forward-No-Answer" to voicemail works fine.
>
> However, Call from SIP phone (3002) to SCCP phone (3001) and
> "Call-Forward-No-Answer" to voicemail doesn't work.  Once the call was
> forwarded, I got fast busy.
>
> What commands I could use to troubleshoot this?
>
> Attached is the CCAPI INOUT debug if that helps.
>
> Thanks!
>
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Re: [OSL | CCIE_Voice] Why RSVP agent breaks xcoder?

[Michael Luo]

Nice catch!

That's what I had suspected and you confirmed it.

When I typed the codec command I saw there were multiple options for g729.
I just didn't know which g729 CUCM was using.

Thank you very much!

Michael

On Sat, Jan 15, 2011 at 12:55 PM, Randall Saborio  wrote:

> Michael,
>
> I think I got it figured it out. Though the solution is kind of strange or
> silly.
>
> I tried to configure both solutions given on that doc and first none were
> working but I got both resolved the same way.
>
> What happened to me is that CUCM would say the capabilites between the
> transcoder and the mtp were not matching.
>
> I had it like  (default transcoder codecs):
> dspfarm profile 6 transcode
>  codec g711ulaw
>  codec g711alaw
>  codec g729ar8
>  codec g729abr8
>  maximum sessions 6
>  associate application SCCP
>
>
> dspfarm profile 5 mtp
>  codec g729r8
>  codec pass-through
>  rsvp
>  maximum sessions software 50
>  associate application SCCP
>
> From the theory I know, g729r8 and g729ar8 should be compatible, but for
> some reason CUCM didn't think so. I went and added g729ar8 to the mtp
> profile, and issue got solved:
>
>
> dspfarm profile 5 mtp
>  codec g729r8
>  codec pass-through
>  codec g729ar8
>  rsvp
>  maximum sessions software 50
>  associate application SCCP
>
> Should have been the same to add g729r8 to the transcoder profile. Probably
> some people would not notice this problem as I know some people just add as
> much codecs they can to the transcoder profile but I usually leave it as
> defaults.
>
> This is nice practice. Thx for sharing!
>
>
>
> On Sat, Jan 15, 2011 at 9:33 AM, Michael Luo  wrote:
>
>> Hi Randall,
>>
>> Thank you for the info.
>>
>> I think my scenario is exactly like this:
>>
>> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a02rsvp.html#wp1034607
>>
>> In the lab, the caller was in region BR1 and the IVR (UCCX with G.711) was
>> in region HQ.  The inter-region codec was configured to use G.729.  Thus a
>> transcoder is needed.
>>
>> The call works fine without RSVP agent (transcoded as expected).
>>
>> When MTP RSVP agent was added to the MRGL, the call failed.  Per the link
>> above, it should work as my MTP was configured as required in the link:
>>
>>  dspfarm profile 2 mtp
>>
>>  codec g729r8
>>
>>  codec pass-through
>>
>>  rsvp
>>
>>  maximum session software 4
>>
>>  associate application sccp
>>
>> Thanks!
>> Michael
>>
>>
>> On Fri, Jan 14, 2011 at 11:08 PM, Randall Saborio wrote:
>>
>>> Michael, I can only contribute to your last question. In my opinion, MTP
>>> is more scalable because you can always have a lot more MTP software
>>> sessions than transcoder, which is always hardware. MTP can be software on
>>> IOS and use G.729 and can have way more sessions limited only by the CPU on
>>> the router.
>>>
>>> I haven't tried what you say from lab 10 but sounds interesting. Will try
>>> to do some labbing on that tomorrow to understand.
>>>
>>> Cheers.
>>>
>>> On Fri, Jan 14, 2011 at 8:00 PM, Michael Luo  wrote:
>>>
>>>> I was reading IP Expert CCIE Voice Volume 1 Proctor Guide page 531 of
>>>> 645 (Lab 10.1).
>>>>
>>>> Quoted from the page "At this stage we would have broken our
>>>> inter-cluster calls".  I was trying to understand why adding the MTP RSVP
>>>> agent would break the transcoding.
>>>>
>>>> I did multiple tests.  Even I had XCode and MTP in the same MRG, the
>>>> call won't invoke the XCode and thus a codec mismatch fails the call.  If I
>>>> remove MTP (RSVP agent) from MRG, it'll work again.
>>>>
>>>> But why is that?  Why MTP would break functionality of Transcoder?  Is
>>>> there a way to make them work together?
>>>>
>>>> Another thing I don't quite understand was - the proctor guide said we
>>>> could user transcoder as RSVP agent but MTP is more scalable.  Why was 
>>>> that?
>>>>
>>>> Thanks!
>>>> Michael
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
>>>> please visit www.ipexpert.com
>>>>
>>>>
>>>
>>>
>>> --
>>> Randall "da ill" Saborio
>>> CCIE Voice Wannabe #10054675811
>>>
>>>
>>
>
>
> --
> Randall "da ill" Saborio
> CCIE Voice Wannabe #10054675811
>
>
___
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Re: [OSL | CCIE_Voice] Why RSVP agent breaks xcoder?

[Michael Luo]

Hi Randall,

Thank you for the info.

I think my scenario is exactly like this:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a02rsvp.html#wp1034607

In the lab, the caller was in region BR1 and the IVR (UCCX with G.711) was
in region HQ.  The inter-region codec was configured to use G.729.  Thus a
transcoder is needed.

The call works fine without RSVP agent (transcoded as expected).

When MTP RSVP agent was added to the MRGL, the call failed.  Per the link
above, it should work as my MTP was configured as required in the link:

 dspfarm profile 2 mtp

 codec g729r8

 codec pass-through

 rsvp

 maximum session software 4

 associate application sccp

Thanks!
Michael

On Fri, Jan 14, 2011 at 11:08 PM, Randall Saborio  wrote:

> Michael, I can only contribute to your last question. In my opinion, MTP is
> more scalable because you can always have a lot more MTP software sessions
> than transcoder, which is always hardware. MTP can be software on IOS and
> use G.729 and can have way more sessions limited only by the CPU on the
> router.
>
> I haven't tried what you say from lab 10 but sounds interesting. Will try
> to do some labbing on that tomorrow to understand.
>
> Cheers.
>
> On Fri, Jan 14, 2011 at 8:00 PM, Michael Luo  wrote:
>
>> I was reading IP Expert CCIE Voice Volume 1 Proctor Guide page 531 of 645
>> (Lab 10.1).
>>
>> Quoted from the page "At this stage we would have broken our inter-cluster
>> calls".  I was trying to understand why adding the MTP RSVP agent would
>> break the transcoding.
>>
>> I did multiple tests.  Even I had XCode and MTP in the same MRG, the call
>> won't invoke the XCode and thus a codec mismatch fails the call.  If I
>> remove MTP (RSVP agent) from MRG, it'll work again.
>>
>> But why is that?  Why MTP would break functionality of Transcoder?  Is
>> there a way to make them work together?
>>
>> Another thing I don't quite understand was - the proctor guide said we
>> could user transcoder as RSVP agent but MTP is more scalable.  Why was that?
>>
>> Thanks!
>> Michael
>>
>>
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
>
> --
> Randall "da ill" Saborio
> CCIE Voice Wannabe #10054675811
>
>
___
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[OSL | CCIE_Voice] Why RSVP agent breaks xcoder?

[Michael Luo]

I was reading IP Expert CCIE Voice Volume 1 Proctor Guide page 531 of 645
(Lab 10.1).

Quoted from the page "At this stage we would have broken our inter-cluster
calls".  I was trying to understand why adding the MTP RSVP agent would
break the transcoding.

I did multiple tests.  Even I had XCode and MTP in the same MRG, the call
won't invoke the XCode and thus a codec mismatch fails the call.  If I
remove MTP (RSVP agent) from MRG, it'll work again.

But why is that?  Why MTP would break functionality of Transcoder?  Is there
a way to make them work together?

Another thing I don't quite understand was - the proctor guide said we could
user transcoder as RSVP agent but MTP is more scalable.  Why was that?

Thanks!
Michael
___
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Re: [OSL | CCIE_Voice] B-ACD "service aa" command syntax?

[Michael Luo]

Never mind.  I forgot the service aa has been defined.

On Sun, Jan 9, 2011 at 12:56 PM, Michael Luo  wrote:

> Hi,
>
> I'm using CME 7.1 on C3825-ADVENTERPRISEK9_IVS-M, Version 12.4(24)T3.
>
> It looks like I cannot specify the TCL script location in the "service aa"
> command.  IOS was expecting  after that.  Any idea?  Thanks!
>
> BR2-RTR(config)#application
> BR2-RTR(config-app)#service aa flash:bacdprompts/app-b-acd-aa-3.0.0.2.tcl
>^
> % Invalid input detected at '^' marker.
>
> BR2-RTR(config-app)#service aa ?
>   
>
> BR2-RTR(config-app)#service ?
>   ipsla-responder
>   CALLIndSs_SErviCe
>   RetrProxy
>   aa
>   session
>   clid_authen_npw
>   app-b-acd-aa
>   clid_authen_collect
>   app-b-acd
>   clid_col_npw_3
>   clid_authen
>   clid_col_npw_npw
>   aa-drop
>   clid_authen_col_npw
>   ipsla-testcall
>   AFW_THIRD_PARTY_CC
>   CTAPP
>   fax_hop_on
>   dsapp
>   queue
>   Default
>   WORD Name of the service/package
>
>
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[OSL | CCIE_Voice] B-ACD "service aa" command syntax?

[Michael Luo]

Hi,

I'm using CME 7.1 on C3825-ADVENTERPRISEK9_IVS-M, Version 12.4(24)T3.

It looks like I cannot specify the TCL script location in the "service aa"
command.  IOS was expecting  after that.  Any idea?  Thanks!

BR2-RTR(config)#application
BR2-RTR(config-app)#service aa flash:bacdprompts/app-b-acd-aa-3.0.0.2.tcl
   ^
% Invalid input detected at '^' marker.

BR2-RTR(config-app)#service aa ?
  

BR2-RTR(config-app)#service ?
  ipsla-responder
  CALLIndSs_SErviCe
  RetrProxy
  aa
  session
  clid_authen_npw
  app-b-acd-aa
  clid_authen_collect
  app-b-acd
  clid_col_npw_3
  clid_authen
  clid_col_npw_npw
  aa-drop
  clid_authen_col_npw
  ipsla-testcall
  AFW_THIRD_PARTY_CC
  CTAPP
  fax_hop_on
  dsapp
  queue
  Default
  WORD Name of the service/package
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[OSL | CCIE_Voice] How to test AAR?

[Michael Luo]

I was doing ipexpert's CCIE Voice Volume 1 Lab 6A Task 6.1 - regarding
Automated Alternate Route (AAR).

In the "Verification" section, it says "Test AAR by reducing the RSVP
BANDWIDTH" allocated to either the HQ or BR1 WAN interface down to 39kbps,
and then making a call from HQ Phone 2 to BR1 Phone 2".

How exactly do I do that?  I applied the command "ip rsvp bandwidth 39" to
HQ/BR1 serial (frame-relay) interface, but the call still goes through IP
network instead of PSTN.

How does CallManager know the bandwidth has been decreased?

Thanks!
Michael
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Re: [OSL | CCIE_Voice] How to test AAR?

[Michael Luo]

Never mind.  The bandwidth was configured on CUCM instead of the "WAN
interface".

On Tue, Jan 4, 2011 at 11:50 AM, Michael Luo  wrote:

> I was doing ipexpert's CCIE Voice Volume 1 Lab 6A Task 6.1 - regarding
> Automated Alternate Route (AAR).
>
> In the "Verification" section, it says "Test AAR by reducing the RSVP
> BANDWIDTH" allocated to either the HQ or BR1 WAN interface down to 39kbps,
> and then making a call from HQ Phone 2 to BR1 Phone 2".
>
> How exactly do I do that?  I applied the command "ip rsvp bandwidth 39" to
> HQ/BR1 serial (frame-relay) interface, but the call still goes through IP
> network instead of PSTN.
>
> How does CallManager know the bandwidth has been decreased?
>
> Thanks!
> Michael
>
>
>
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Re: [OSL | CCIE_Voice] Stop Routing on Unallocated Number Flag

[Michael Luo]

No.  I reviewed CCM trace.  CCM stopped routing once the 1st gateway (MGCP)
got "Unallocated Number" from PSTN.  CCM didn't attempt the 2nd gateway
(H323) at all.

If I shutdown the voice-port on the 1st gateway, CCM route calls to 2nd
one.  So I know route pattern, route list and route group were configured
properly.

It looks like CCM just didn't honor the service parameter "Stop Routing on
Unallocated Number Flag".  Some posts said it was for ICT (Inter-Cluster
Trunk) only.  Not sure if that's reason.

Any suggestion to troubleshoot?

Thanks!
Michael

On Wed, Dec 29, 2010 at 9:01 AM, Prashant Patel
wrote:

> Hi Michael,
>
> Do you see the call hitting the h323 gw in debug isdn q931 ?
>
> Thanks,
> Prashant
>
> On Wed, Dec 29, 2010 at 12:32 AM, Michael Luo  wrote:
>
>> Has anyone tested the "Stop Routing on Unallocated Number Flag" and made
>> it worked?
>>
>> I have two route groups:
>>
>> RG1 was pointing to a MGCP gateway.
>> RG2 was pointing to a H323 gateway.
>>
>> Then I have a route list (RL) pointing to these two route groups with RG1
>> on the top and RG2 at the bottom.
>>
>> The purpose was to test redundancy of the gateways.
>>
>> If I shut down the voice-port on MGCP gateway, test call went through H323
>> gateway, which was the desired behavior.
>>
>> However, if MGCP gateway got a "Unallocated Number" message from telco
>> side, test call didn't route through H323 gateway.
>>
>> I set "Stop Routing on Unallocated Number Flag" and "Stop Routing on User
>> Busy Flag" to "False".  I also restarted "Cisco CallManager" service.  It
>> didn't seem to take effect.  I'm still getting CCM announcement - "The call
>> cannot be completed as dialed.  Please check your number and try again".
>>
>> Any hints will be appreciated.
>>
>> Michael
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
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[OSL | CCIE_Voice] Stop Routing on Unallocated Number Flag

[Michael Luo]

Has anyone tested the "Stop Routing on Unallocated Number Flag" and made it
worked?

I have two route groups:

RG1 was pointing to a MGCP gateway.
RG2 was pointing to a H323 gateway.

Then I have a route list (RL) pointing to these two route groups with RG1 on
the top and RG2 at the bottom.

The purpose was to test redundancy of the gateways.

If I shut down the voice-port on MGCP gateway, test call went through H323
gateway, which was the desired behavior.

However, if MGCP gateway got a "Unallocated Number" message from telco side,
test call didn't route through H323 gateway.

I set "Stop Routing on Unallocated Number Flag" and "Stop Routing on User
Busy Flag" to "False".  I also restarted "Cisco CallManager" service.  It
didn't seem to take effect.  I'm still getting CCM announcement - "The call
cannot be completed as dialed.  Please check your number and try again".

Any hints will be appreciated.

Michael
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Re: [OSL | CCIE_Voice] Localization - Calling Party Transformation CSS

[Michael Luo]

Never mind.  On the phone configuration page, I had the "Use Device Pool
Calling Party Transformation CSS" checked.  Uncheck that fixed the problem.

On Mon, Dec 27, 2010 at 1:47 PM, Michael Luo  wrote:

> Hi,
>
> I was doing the "localization" with "Calling Party Transformation CSS" per
> the proctor guide.
>
> One thing that puzzled me was: if I set the "Calling Party Transformation
> CSS" in Device Pool, it works.  If I set the "Calling Party Transformation
> CSS" on the phone, it doesn't work.
>
> I reviewed CCM SDI logs for the non-working one:
>
> 12/27/2010 11:43:18.064 CCM| LocalizeCgpnAndSendOutpulsedNumber:
> StationCdpc on device HQSCCP , CSS = ,useDevicePoolCgpnCss =1
> AlternateCgpn(global)=+12123942123
> cgpn=+12123942123|
>
> Based on "CSS = ,useDevicePoolCgpnCss =1 AlternateCgpn(global)=+12123942123
> cgpn=+12123942123", I guess CCM was not able to see any "Calling Party
> Transformation CSS" configured on the phone and tried to use the one
> configured in device pool.
>
> Is there any way we can configure it on the phone instead of device pool?
> Just for curiosity.
>
> Thanks!
> Michael
>
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[OSL | CCIE_Voice] Localization - Calling Party Transformation CSS

[Michael Luo]

Hi,

I was doing the "localization" with "Calling Party Transformation CSS" per
the proctor guide.

One thing that puzzled me was: if I set the "Calling Party Transformation
CSS" in Device Pool, it works.  If I set the "Calling Party Transformation
CSS" on the phone, it doesn't work.

I reviewed CCM SDI logs for the non-working one:

12/27/2010 11:43:18.064 CCM| LocalizeCgpnAndSendOutpulsedNumber: StationCdpc
on device HQSCCP , CSS = ,useDevicePoolCgpnCss =1
AlternateCgpn(global)=+12123942123
cgpn=+12123942123|

Based on "CSS = ,useDevicePoolCgpnCss =1 AlternateCgpn(global)=+12123942123
cgpn=+12123942123", I guess CCM was not able to see any "Calling Party
Transformation CSS" configured on the phone and tried to use the one
configured in device pool.

Is there any way we can configure it on the phone instead of device pool?
Just for curiosity.

Thanks!
Michael
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