Re: [OSL | CCIE_Voice] CME SRST Mode

[Jeff Price (jeffpric)]

Your source address changed from the first post, was it supposed to?

I'm really reach out on a limb here trying to help you b/c your config looks 
correct.

I would maybe suggest starting from square one and keep it simple.  

Remove all of the commands and only include the srst mode auto all command.  
See if your phones register like that.  Then add your various other statements 
one by one.  See if one of them is screwing everything up?

Jeff

-Original Message-
From: CCIE Voice [mailto:ccievoiced...@gmail.com] 
Sent: Wednesday, July 14, 2010 4:32 PM
To: Jeff Price (jeffpric); 'ccieid1ot'
Cc: 'ccie_voice@onlinestudylist.com'; ccie_voice-boun...@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CME SRST Mode

Jeff is correct...whenever I have to make a change to CME SRST config,
especially the ephones, I copy all of my related config to notepad and issue
the no telephony-service command.  I then paste everything back in and it
works fine.  Don't know if this is a bug or what but it happens every single
time.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Wednesday, July 14, 2010 7:12 PM
To: ccieid1ot
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

I have definitely had this problem before, but never really figure out why
it happens.  

Try this:
Copy your telephony-service config to Notepad 
do a no telephony-service
Register the phones with CUCM again
Paste in your telephony-service config
Bring down the WAN connection.  

Eventually the phones will register with the proper DNs.

It's not ideal, but it works most of the time.

Hope this helps,
Jeff

-Original Message-
From: ccieid1ot [mailto:ccieid...@gmail.com] 
Sent: Wednesday, July 14, 2010 4:10 PM
To: Jeff Price (jeffpric)
Cc: Edwin Dotson; ccie_voice@onlinestudylist.com;
ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

Jeff,

same thing.  I had it registered back to CCM then failback to SRST.


ccievoice-sitec#show telephony-service ephone-dn

ephone-dn 4 octo-line
number B16A55 no-reg primary
preference 0 secondary 9
huntstop
huntstop channel 8
call-waiting beep
conference ad-hoc

ephone-dn 10 octo-line
number A100
preference 0 secondary 9
huntstop
huntstop channel 8
call-waiting beep
conference ad-hoc

ccievoice-sitec#show run | s telephony
telephony-service
 sdspfarm units 5
 sdspfarm tag 1 SC-CFB
 no privacy
 conference hardware
 srst mode auto-provision all
 srst ephone template 1
 srst dn template 1
 srst dn line-mode octo
 max-ephones 10
 max-dn 10
 ip source-address 142.102.66.254 port 2000
 system message Your phones are in fallback
 max-conferences 8 gain -6
 call-forward pattern .T
 transfer-system full-consult
 create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
ccievoice-sitec#


On Wed, Jul 14, 2010 at 5:57 PM, Jeff Price (jeffpric)
 wrote:
> You have the auto provision mode set to none.  You should set to all so
that CME will download the phone's config completely.  This is most likely
your problem.
>
> Try and change the command to srst mode auto all and let me know what
happens.
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
> Sent: Wednesday, July 14, 2010 3:54 PM
> To: Edwin Dotson
> Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CME SRST Mode
>
> Done that, and phones get nothing.
>
> On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson  wrote:
>> Have you tried removing the static ephone configuration? They may be
picking that profile which don't have buttons specified.
>>
>> Thanks
>> Edwin
>> Sent from my Verizon Wireless BlackBerry
>>
>> -Original Message-
>> From: ccieid1ot 
>> Sender: "ccie_voice-boun...@onlinestudylist.com"
>>        
>> Date: Wed, 14 Jul 2010 15:20:27
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] CME SRST Mode
>>
>> Hi gangs,
>>
>> I can not seem to get the ephones to register with their assigned
>> DN's.  Tried all types of srst mode auto-pro commands.  Someone please
>> give me a sanity check.
>>
>>
>> !
>> telephony-service
>>  sdspfarm units 5
>>  sdspfarm tag 1 SC-CFB
>>  no privacy
>>  conference hardware
>>  srst mode auto-provision none
>>  srst ephone template 1
>>  srst dn template 1
>>  srst dn line-mode octo
>>  max-ephones 10
>>  max-dn 10
>>  ip source-address 192.102.66.254 port 2000
>>  system message Your phones are in fallback
>>  max-c

Re: [OSL | CCIE_Voice] CME SRST Mode

[Jeff Price (jeffpric)]

Good point.  

-Original Message-
From: Randall Saborio [mailto:ill2...@gmail.com] 
Sent: Wednesday, July 14, 2010 4:10 PM
To: Jeff Price (jeffpric)
Cc: ccieid1ot; Edwin Dotson; ccie_voice@onlinestudylist.com; 
ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

srst mode auto-provision none is not a problem if you intend it this
way. However, you must add the buttons to the ephone configuration:
ephone 1
button 1:4


On Wed, Jul 14, 2010 at 4:57 PM, Jeff Price (jeffpric)
 wrote:
> You have the auto provision mode set to none.  You should set to all so that 
> CME will download the phone's config completely.  This is most likely your 
> problem.
>
> Try and change the command to srst mode auto all and let me know what happens.
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
> Sent: Wednesday, July 14, 2010 3:54 PM
> To: Edwin Dotson
> Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CME SRST Mode
>
> Done that, and phones get nothing.
>
> On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson  wrote:
>> Have you tried removing the static ephone configuration? They may be picking 
>> that profile which don't have buttons specified.
>>
>> Thanks
>> Edwin
>> Sent from my Verizon Wireless BlackBerry
>>
>> -Original Message-
>> From: ccieid1ot 
>> Sender: "ccie_voice-boun...@onlinestudylist.com"
>>        
>> Date: Wed, 14 Jul 2010 15:20:27
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] CME SRST Mode
>>
>> Hi gangs,
>>
>> I can not seem to get the ephones to register with their assigned
>> DN's.  Tried all types of srst mode auto-pro commands.  Someone please
>> give me a sanity check.
>>
>>
>> !
>> telephony-service
>>  sdspfarm units 5
>>  sdspfarm tag 1 SC-CFB
>>  no privacy
>>  conference hardware
>>  srst mode auto-provision none
>>  srst ephone template 1
>>  srst dn template 1
>>  srst dn line-mode octo
>>  max-ephones 10
>>  max-dn 10
>>  ip source-address 192.102.66.254 port 2000
>>  system message Your phones are in fallback
>>  max-conferences 8 gain -6
>>  call-forward pattern .T
>>  transfer-system full-consult
>>  create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
>> !
>> !
>> ephone-dn-template  1
>>  call-forward busy 91658200
>>  call-forward noan 91658200 timeout 20
>> !
>> !
>> ephone-template  1
>>  privacy off
>>  privacy-button
>>  softkeys remote-in-use  Newcall CBarge
>>  softkeys idle  Newcall Redial
>>  softkeys seized  Redial Endcall Cfwdall Pickup Gpickup Meetme
>>  softkeys connected  Hold Endcall Trnsfer Park Confrn ConfList
>> !
>> !
>> ephone-dn  4  octo-line
>>  number B16A55 no-reg primary
>>  conference ad-hoc
>> !
>> !
>> ephone-dn  10  octo-line
>>  number A100
>>  conference ad-hoc
>> !
>> !
>> ephone  1
>>  privacy-button
>>  device-security-mode none
>>  max-calls-per-button 4
>>  busy-trigger-per-button 1
>> !
>> !
>> !
>> ephone  2
>>  privacy-button
>>  device-security-mode none
>>  max-calls-per-button 4
>>  busy-trigger-per-button 1
>> ___
>> 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
>



-- 
Randall "da ill" Saborio
CCIE Voice Wannabe #10054675811
(Real number coming this July 2010)
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] CME SRST Mode

[Jeff Price (jeffpric)]

I have definitely had this problem before, but never really figure out why it 
happens.  

Try this:
Copy your telephony-service config to Notepad 
do a no telephony-service
Register the phones with CUCM again
Paste in your telephony-service config
Bring down the WAN connection.  

Eventually the phones will register with the proper DNs.

It's not ideal, but it works most of the time.

Hope this helps,
Jeff

-Original Message-
From: ccieid1ot [mailto:ccieid...@gmail.com] 
Sent: Wednesday, July 14, 2010 4:10 PM
To: Jeff Price (jeffpric)
Cc: Edwin Dotson; ccie_voice@onlinestudylist.com; 
ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

Jeff,

same thing.  I had it registered back to CCM then failback to SRST.


ccievoice-sitec#show telephony-service ephone-dn

ephone-dn 4 octo-line
number B16A55 no-reg primary
preference 0 secondary 9
huntstop
huntstop channel 8
call-waiting beep
conference ad-hoc

ephone-dn 10 octo-line
number A100
preference 0 secondary 9
huntstop
huntstop channel 8
call-waiting beep
conference ad-hoc

ccievoice-sitec#show run | s telephony
telephony-service
 sdspfarm units 5
 sdspfarm tag 1 SC-CFB
 no privacy
 conference hardware
 srst mode auto-provision all
 srst ephone template 1
 srst dn template 1
 srst dn line-mode octo
 max-ephones 10
 max-dn 10
 ip source-address 142.102.66.254 port 2000
 system message Your phones are in fallback
 max-conferences 8 gain -6
 call-forward pattern .T
 transfer-system full-consult
 create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
ccievoice-sitec#


On Wed, Jul 14, 2010 at 5:57 PM, Jeff Price (jeffpric)
 wrote:
> You have the auto provision mode set to none.  You should set to all so that 
> CME will download the phone's config completely.  This is most likely your 
> problem.
>
> Try and change the command to srst mode auto all and let me know what happens.
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
> Sent: Wednesday, July 14, 2010 3:54 PM
> To: Edwin Dotson
> Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CME SRST Mode
>
> Done that, and phones get nothing.
>
> On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson  wrote:
>> Have you tried removing the static ephone configuration? They may be picking 
>> that profile which don't have buttons specified.
>>
>> Thanks
>> Edwin
>> Sent from my Verizon Wireless BlackBerry
>>
>> -Original Message-
>> From: ccieid1ot 
>> Sender: "ccie_voice-boun...@onlinestudylist.com"
>>        
>> Date: Wed, 14 Jul 2010 15:20:27
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] CME SRST Mode
>>
>> Hi gangs,
>>
>> I can not seem to get the ephones to register with their assigned
>> DN's.  Tried all types of srst mode auto-pro commands.  Someone please
>> give me a sanity check.
>>
>>
>> !
>> telephony-service
>>  sdspfarm units 5
>>  sdspfarm tag 1 SC-CFB
>>  no privacy
>>  conference hardware
>>  srst mode auto-provision none
>>  srst ephone template 1
>>  srst dn template 1
>>  srst dn line-mode octo
>>  max-ephones 10
>>  max-dn 10
>>  ip source-address 192.102.66.254 port 2000
>>  system message Your phones are in fallback
>>  max-conferences 8 gain -6
>>  call-forward pattern .T
>>  transfer-system full-consult
>>  create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
>> !
>> !
>> ephone-dn-template  1
>>  call-forward busy 91658200
>>  call-forward noan 91658200 timeout 20
>> !
>> !
>> ephone-template  1
>>  privacy off
>>  privacy-button
>>  softkeys remote-in-use  Newcall CBarge
>>  softkeys idle  Newcall Redial
>>  softkeys seized  Redial Endcall Cfwdall Pickup Gpickup Meetme
>>  softkeys connected  Hold Endcall Trnsfer Park Confrn ConfList
>> !
>> !
>> ephone-dn  4  octo-line
>>  number B16A55 no-reg primary
>>  conference ad-hoc
>> !
>> !
>> ephone-dn  10  octo-line
>>  number A100
>>  conference ad-hoc
>> !
>> !
>> ephone  1
>>  privacy-button
>>  device-security-mode none
>>  max-calls-per-button 4
>>  busy-trigger-per-button 1
>> !
>> !
>> !
>> ephone  2
>>  privacy-button
>>  device-security-mode none
>>  max-calls-per-button 4
>>  busy-trigger-per-button 1
>> ___
>> 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] CME SRST Mode

[Jeff Price (jeffpric)]

You have the auto provision mode set to none.  You should set to all so that 
CME will download the phone's config completely.  This is most likely your 
problem.

Try and change the command to srst mode auto all and let me know what happens.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
Sent: Wednesday, July 14, 2010 3:54 PM
To: Edwin Dotson
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

Done that, and phones get nothing.

On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson  wrote:
> Have you tried removing the static ephone configuration? They may be picking 
> that profile which don't have buttons specified.
>
> Thanks
> Edwin
> Sent from my Verizon Wireless BlackBerry
>
> -Original Message-
> From: ccieid1ot 
> Sender: "ccie_voice-boun...@onlinestudylist.com"
>        
> Date: Wed, 14 Jul 2010 15:20:27
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] CME SRST Mode
>
> Hi gangs,
>
> I can not seem to get the ephones to register with their assigned
> DN's.  Tried all types of srst mode auto-pro commands.  Someone please
> give me a sanity check.
>
>
> !
> telephony-service
>  sdspfarm units 5
>  sdspfarm tag 1 SC-CFB
>  no privacy
>  conference hardware
>  srst mode auto-provision none
>  srst ephone template 1
>  srst dn template 1
>  srst dn line-mode octo
>  max-ephones 10
>  max-dn 10
>  ip source-address 192.102.66.254 port 2000
>  system message Your phones are in fallback
>  max-conferences 8 gain -6
>  call-forward pattern .T
>  transfer-system full-consult
>  create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
> !
> !
> ephone-dn-template  1
>  call-forward busy 91658200
>  call-forward noan 91658200 timeout 20
> !
> !
> ephone-template  1
>  privacy off
>  privacy-button
>  softkeys remote-in-use  Newcall CBarge
>  softkeys idle  Newcall Redial
>  softkeys seized  Redial Endcall Cfwdall Pickup Gpickup Meetme
>  softkeys connected  Hold Endcall Trnsfer Park Confrn ConfList
> !
> !
> ephone-dn  4  octo-line
>  number B16A55 no-reg primary
>  conference ad-hoc
> !
> !
> ephone-dn  10  octo-line
>  number A100
>  conference ad-hoc
> !
> !
> ephone  1
>  privacy-button
>  device-security-mode none
>  max-calls-per-button 4
>  busy-trigger-per-button 1
> !
> !
> !
> ephone  2
>  privacy-button
>  device-security-mode none
>  max-calls-per-button 4
>  busy-trigger-per-button 1
> ___
> 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] Vol2 Lab8 Call Forward to VM

[Jeff Price (jeffpric)]

Do you have a mask configured at both the Hunt Pilot and the VM Pilot?
That may be why CUC isn't recognizing the extension.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin
Damisch
Sent: Tuesday, July 13, 2010 11:54 AM
To: osl osl
Subject: [OSL | CCIE_Voice] Vol2 Lab8 Call Forward to VM

 

This is usually a no-brainer.  Working on the VM section of Vol2 Lab8.
Whenever a forward busy/no answer call to 5002 goes to VM, it plays the
opening greeting instead of going to the 5002 mailbox.  RTMT doesn't
show any info about 5002.  Caller is 5001, called is 5600, reason is
Direct, and Redir/Last Redir are empty.  These are SIP phones and using
the SIP trunk to Unity Connection and not sure what is different about
doing this compared to the old school VM port wizard method.  I can
access the mailbox on 5002, then choose the option to send a VM to
itself, lights up MWI, I can check it, and MWI goes off.  That part is
good, it's just the busy/no answer doesn't work properly.  I've never
seen this behavior in production either.  Any thoughts?

 



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disclosure under applicable law. If you are not the intended recipient,
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please notify Vital Support Systems at 515 334 5700 and delete or
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___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue

[Jeff Price (jeffpric)]

I still think the easiest way to do this is to have all dial-peers with
a 9.  As you are configuring, you may run into issues with TEHO, but if
you do a debug voip dialpeer you can see the incoming number and add any
"extra" dial-peers you may need.  I have noticed for some reason when I
configure TEHO, sometimes CUCM doesn't add the 9 even though it is in
the prefix box of the RL.  In this event, as I said, it would be easier
to do the debug and figure out what is being sent.

 

This simplifies adding the SRST functionality, which you can pretty much
guarantee you will have to configure in some fashion.  As you do these
configurations over and over in this method, you will start to just
think naturally about sending the 9.  It is easier to have the 1
dial-peer than have to create 2 for each type of dialing pattern.

 

Just my opinion,

 

Jeff 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark
Holloway
Sent: Monday, July 12, 2010 6:23 PM
To: Graham Hopkins
Cc: CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue

 

I proceeded to use the method where all my H323 dial-peers start with 9
in the destination-pattern.  I imagine it's more work to have UCM keep
the 9 on the dialed number because of TEHO to multiple gateways, it gets
very "busy" to know when to prepend and not prepend in UCM route lists.
Assuming 9 is stripped on UCM and the H323 gateway is adding 9 before
sending the call to a POTS dial peer, is a VoIP dial-peer being created
to match any incoming call and then it is sent through a
translation-profile so it can match a POTS dial peer?

 

 

 

On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote:





With the two sets of dial-peers you do need to take care that
overlapping patterns don't cause problems in SRST for example I hit
issues with 

 

[2-9]..

 

and 

 

91[2-9]..[2-9]..

 

I decided to go with the translation pattern to put the 9 back on to the
digits sent by CUCM, but this 9 will still show up on the phone unless
you use

 

voice service voip

no supplementary-service h225-notify cid-update

 

Regards

 

Graham Hopkins

 

 

 

 

On 9 Jul 2010, at 19:21, Mark Holloway wrote:





Sounds like you have the PSTN to CUCM part working ok.  

 

This is what I have been doing.

 

On the H323 router create the following dial-peer 

 

dial-peer voice 10 pots

destination-pattern [2-9]..$

port 0/0/0:23

 

On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls
originated by BR1 phones and strip the predot. This way you can assign
the call type as Subscriber within the Route Pattern and if local calls
are supposed to send a 7 digit calling number you can set the calling
party transformation mask within the Route Pattern to XXX.

 

 

You could have a second dial-peer on your H323 router for SRST 

 

dial-peer voice 910 pots

destination-pattern 9[2-9]..$

port 0/0/0:23

translation-profile outgoing LOCAL

 

 

There are really two different ways to handle H323 gateway dial-peers.
You can strip the 9 in CUCM then add it back on the H323 gateway through
a translation-profile and only have one set of dial-peers.  Or, build
your dial-peers for local, LD, international, and 911 without the 9,
copy/paste in notepad and put a 9 in front of the dial-peer number and
the destination-pattern then paste it into your router. You will have
two sets of dial-peers for SRST and normal operation.

 

 

 

 

On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote:





HI Team,

 

I have an issue with this question.

 

Question

===

when pstn number 414363 call phones at site b they should display 7
digits on the phone display. 
For example when pstn calling ph 1 or ph 2 at branch B it should display
363 on the screen.

 

 

My solution

=

 

I have added +1 in Device pool of Branch B to make it globalised when
the call comes in the H323 Branch B router.

 

I have created \+1414.363 calling party transformation mask.

 

I have created \+1414.363 route pattern with Branch B as the
gateway. (branch b is the H323 gateway).

 

So on the Route pattern i have just done predot and in the branch b
route list i have done NANP-Predot and prefix 9. I have done vice versa
as well but things doesnt work.

 

IN the branch B router i have a dial-peer for the local calls.

 

dial-peer voice 1 pots

destination-pattern 9[2-9]..

port 0/0/0:23

translation-profile outgoing local

 

translation-rule 1 

rule 1 /^8.../ /363\0/

 

translation-rule 2 

rule 1 // // type any sub plan any isdn

 

translation-profile lcoal

translate called 2

translate calling 1

 

Note: If i make a dial-peer without 9 i.e (...)

Then the display is perfect. but i dont feel this would be the solution.

 

because in srst this would be an issue.

 

 

Issue

=

 

The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the
caller id is 363 a

Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue

[Jeff Price (jeffpric)]

The easiest way to accomplish this is to create a dial-peer for the
number w/o the 9.  Also, to effect the display on the phone, I've also
seen that you would have to strip predot on the Route Pattern.

 

So you will need the following:

-   A route pattern \+1414.XXX with DDI Predot - To route the
call and display correctly on the phone

-   A calling party transformation incoming \+1414.XXX DDI
Predot- to strip the number down on the display when calling from the
PSTN

-   A called party transformation outgoing \+1414.XXX DDI
Predot- to strip the number down when the call is going out

-   A dial-peer for the number 363 - to route the call from the
GW

 

I have done this plenty of times without issue with SRST.  Your users
would still use 9 to dial outside, therefore they technically wouldn't
be aware of that dial-peer and wouldn't dial the number 363.  Also
if you do it right, you can still add redundancy at the RL to allow
failover to the backup gateways.

 

Also, it's worth noting that you should probably apply the incoming
calling number prefix on the gateway instead of the Device Pool.
Although I suppose it could be personal preference rather than any real
issue.

 

Hope this helps,

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joaquim
Fernandes
Sent: Friday, July 09, 2010 10:28 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Globalisation/Localisation Issue

 

HI Team,

 

I have an issue with this question.

 

Question

===

when pstn number 414363 call phones at site b they should display 7
digits on the phone display. 
For example when pstn calling ph 1 or ph 2 at branch B it should display
363 on the screen.

 

 

My solution

=

 

I have added +1 in Device pool of Branch B to make it globalised when
the call comes in the H323 Branch B router.

 

I have created \+1414.363 calling party transformation mask.

 

I have created \+1414.363 route pattern with Branch B as the
gateway. (branch b is the H323 gateway).

 

So on the Route pattern i have just done predot and in the branch b
route list i have done NANP-Predot and prefix 9. I have done vice versa
as well but things doesnt work.

 

IN the branch B router i have a dial-peer for the local calls.

 

dial-peer voice 1 pots

destination-pattern 9[2-9]..

port 0/0/0:23

translation-profile outgoing local

 

translation-rule 1 

rule 1 /^8.../ /363\0/

 

translation-rule 2 

rule 1 // // type any sub plan any isdn

 

translation-profile lcoal

translate called 2

translate calling 1

 

Note: If i make a dial-peer without 9 i.e (...)

Then the display is perfect. but i dont feel this would be the solution.

 

because in srst this would be an issue.

 

 

Issue

=

 

The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the
caller id is 363 and in the missed call its globalized number
+1414363 

as per the question. 

 

But when i do redial using missed calls from Branch B ph1 or ph2 the
calling number on the ip phones is displayed as 9363 (9 is the
secondary dial tone) and the call goes through. Evrything works fine
except for the display on ph1 or ph2, there is 9.

 

How do i get rid of it 9.

 

I hope i have made my point very clear of what issue i am facing. The
question state the display on the phone should be only 363 and not
9363.


Regards, 
JF

 

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

Re: [OSL | CCIE_Voice] PSTN Gatekeeper

[Jeff Price (jeffpric)]

The incoming called number would actually be this:

 

dial-peer voice 200 voip

incoming called 1#

translation-profile in FROM_GK

 

voice translation-rule 1

rule 1 /1#\(.*\)/ /\1/

 

voice translation-profile FROM_GK

translate called 1

 

The called number will come in from the GK with the tech prefix that you
configured.  You will need to strip this in order to route the call
correctly.  Right now you are receiving an unallocated number b/c the
number would come in 1#01191674563892 and the GW doesn't have this
number for any endpoints.

 

If you aren't even getting any output from the debug voip dialpeer
command, that would mean that you have a GK routing issue.  However, I
suspect that b/c your dialpeer sent the call back to RAS, you can
probably fix with what I am telling you.  Keep me posted on your
progress.

 

Hope this helps,

Jeff


 

From: Mouhammad Nasser [mailto:engnasse...@hotmail.com] 
Sent: Wednesday, July 07, 2010 11:36 AM
To: Jeff Price (jeffpric)
Cc: ipexpert
Subject: RE: [OSL | CCIE_Voice] PSTN Gatekeeper

 

Hi Jeff,
 
Thank you for your reply
 
In order to make things simplist, I altered the extension number
configured at CME to be exactly like the arrived dialed string
"011916745738932", so no translation would be required
 
Anyway, I wasn't having an incoming dial-peer as you mentioned, so I
added the below:
 
dial-peer voice 200 voip
 incoming called-number 01191T
 session target ras
!
 
But also with no hope, the call does never reach the gateway (gateway
part of the router), because I receive no output at all for both debug
 
!
debug voip dialpeer all
 
debug voip ccapi inout
!
 
Sir, I have been trying for few days on it with no hope :(
 
I shall appreciate any help
 
 
BR,

 



Subject: RE: [OSL | CCIE_Voice] PSTN Gatekeeper
Date: Wed, 7 Jul 2010 13:26:29 -0500
From: jeffp...@cisco.com
To: engnasse...@hotmail.com; ccie_voice@onlinestudylist.com

My apologies, I didn't read the whole thing to see the H225 messages.  I
would still make sure that you have the incoming dial-peer configured.
If the location request/confirm is sent, that would mean the GK routing
is there, but the actual call routing isn't.  Mainly you need to strip
the tech-prefix that will be in front of the dialed number.  Depending
on how you have the DN configured on the PSTN router, you may need to
strip the string down further, but this would be based upon your
specific implementation.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mouhammad
Nasser
Sent: Wednesday, July 07, 2010 11:09 AM
To: ipexpert
Subject: [OSL | CCIE_Voice] PSTN Gatekeeper

 

Hi Experts,
 
I am trying to configure PSTN gatekeeper to test Gatekeeper VIA routing,
so I decided to start with a simple scenario where no VIA zone is
introduced, but I am not able to get it working, the dialed string
"011916745738932" arrives to PSTN gatekeeper but with no hope to match,
and the phone never rings
 
 
I hope anyone can help
 
My configuration and debug are below
 
HQ router
 
gatekeeper
 zone local GK ccie.com 142.1.64.254
 zone remote BBGK cisco.com 10.1.5.5 1719
 zone prefix BBGK 011*
 no shutdown
!
++
on PSTN router it is:
 
interface GigabitEthernet0/0.5
 encapsulation dot1Q 5 native
 ip address 10.1.5.5 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id BBGK ipaddr 10.1.5.5 1719
 h323-gateway voip h323-id PSTN
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 10.1.5.5
!
!
gatekeeper
 zone local BBGK cisco.com 10.1.5.5
 zone remote GK ccie.com 142.1.64.254 1719
 zone prefix BBGK 011*
 gw-type-prefix 1#* default-technology
 no shutdown
!
!
the dialed string is registered to gatekeeper:
 
PSTN-RTR#sh gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags 
--- - --- - -   &nb! sp;
 - 
10.1.5.51720  10.1.5.556645 BBGK  VOIP-GW 
E164-ID: 011916745738932
H323-ID: PSTN
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 1
 
 
the output of debug h225 asn1 is below
 
Jul  7 18:08:51.891: 
Jul  7 18:08:51.891: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest : 
{
  requestSeqNum 2054
  destinationInfo 
  {
dialedDigits : "011916745738932"
  }
  nonStandardData 
  {
nonStandardIdentifier h221NonStandard : 
{
  t35CountryCode 181
  t35Extension 0
  manufacturerCode 18
}
data '828B903011EDB5EBEBA941C3070007018ECA...'H
  }
  replyAddress ipAddress : 
  {
ip '8E0140FE'H
port 1719
  }
  sourceInfo 

Re: [OSL | CCIE_Voice] "Request pending..." when trying to access IPPAservice

[Jeff Price (jeffpric)]

What port number are you using?  The URL provide in the CAD installation
guide uses 8080, but the real port should be 6293.  Try that if you have
not.

 

Jeff

 

From: jeremy co [mailto:jeremy.coo...@gmail.com] 
Sent: Wednesday, July 07, 2010 11:29 AM
To: Jeff Price (jeffpric); avholloway+cisco-v...@gmail.com
Cc: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net
Subject: Re: [OSL | CCIE_Voice] "Request pending..." when trying to
access IPPAservice

 

Hi guys,

 

I reloaded both CUCM and UCCX servers and  I cannot get anything in
browser with this URL.

 

 

Cheers,

 

Jeremy

On Thu, Jul 8, 2010 at 4:22 AM, Jeff Price (jeffpric)
 wrote:

I've seen that a restart of the IPPA service and UCCX Engine under the
Control Center will fix most problems such as this, however I can't say
that I have ever seen that exact status.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co
Sent: Wednesday, July 07, 2010 11:04 AM
To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net
Subject: [OSL | CCIE_Voice] "Request pending..." when trying to access
IPPAservice

 

Hi Guys,

 

I get "Request pending..." when trying to access IPPA service

 

 

URL I used.:
http://100.0.0.11:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp

 

Also all of the names have been changed to IP addresses in Enterprise
param.

 

 

Any idea ?

 

Cheers,

 

Jeremy

 

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

Re: [OSL | CCIE_Voice] PSTN Gatekeeper

[Jeff Price (jeffpric)]

My apologies, I didn't read the whole thing to see the H225 messages.  I
would still make sure that you have the incoming dial-peer configured.
If the location request/confirm is sent, that would mean the GK routing
is there, but the actual call routing isn't.  Mainly you need to strip
the tech-prefix that will be in front of the dialed number.  Depending
on how you have the DN configured on the PSTN router, you may need to
strip the string down further, but this would be based upon your
specific implementation.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mouhammad
Nasser
Sent: Wednesday, July 07, 2010 11:09 AM
To: ipexpert
Subject: [OSL | CCIE_Voice] PSTN Gatekeeper

 

Hi Experts,
 
I am trying to configure PSTN gatekeeper to test Gatekeeper VIA routing,
so I decided to start with a simple scenario where no VIA zone is
introduced, but I am not able to get it working, the dialed string
"011916745738932" arrives to PSTN gatekeeper but with no hope to match,
and the phone never rings
 
 
I hope anyone can help
 
My configuration and debug are below
 
HQ router
 
gatekeeper
 zone local GK ccie.com 142.1.64.254
 zone remote BBGK cisco.com 10.1.5.5 1719
 zone prefix BBGK 011*
 no shutdown
!
++
on PSTN router it is:
 
interface GigabitEthernet0/0.5
 encapsulation dot1Q 5 native
 ip address 10.1.5.5 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id BBGK ipaddr 10.1.5.5 1719
 h323-gateway voip h323-id PSTN
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 10.1.5.5
!
!
gatekeeper
 zone local BBGK cisco.com 10.1.5.5
 zone remote GK ccie.com 142.1.64.254 1719
 zone prefix BBGK 011*
 gw-type-prefix 1#* default-technology
 no shutdown
!
!
the dialed string is registered to gatekeeper:
 
PSTN-RTR#sh gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags 
--- - --- - - 
- 
10.1.5.51720  10.1.5.556645 BBGK  VOIP-GW 
E164-ID: 011916745738932
H323-ID: PSTN
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 1
 
 
the output of debug h225 asn1 is below
 
Jul  7 18:08:51.891: 
Jul  7 18:08:51.891: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest : 
{
  requestSeqNum 2054
  destinationInfo 
  {
dialedDigits : "011916745738932"
  }
  nonStandardData 
  {
nonStandardIdentifier h221NonStandard : 
{
  t35CountryCode 181
  t35Extension 0
  manufacturerCode 18
}
data '828B903011EDB5EBEBA941C3070007018ECA...'H
  }
  replyAddress ipAddress : 
  {
ip '8E0140FE'H
port 1719
  }
  sourceInfo 
  {
h323-ID : {"GK"}
  }
  canMapAlias TRUE
  hopCount 6
}
 
Jul  7 18:08:51.891: H225 NONSTD INCOMING ENCODE BUFFER::=
828B903011EDB5EBEBA941C3070007018ECA401D05010180833407000A01C815820A
211E0034003800330044003800320044004300300030003000300030003000300032
Jul  7 18:08:51.891: 
Jul  7 18:08:51.891: H225 NONSTD INCOMING PDU ::=
value LRQnonStandardInfo ::= 
{
  ttl 6
  nonstd-callIdentifier 
  {
guid '00EDB5EBEBA941C3070007018ECA401D'H
  }
  gatewaySrcInfo 
  {
e164 : "5001"
  }
  h225NonStdSrcCallSignalAddress h225NonStdIpAddress : 
  {
ip '0A01C815'H
port 33290
  }
  h225NonStdSrcendpointIdentifier {"483D82DC0002"}
}
 
Jul  7 18:08:51.895: H225 NONSTD OUTGOING PDU ::=
value LCFnonStandardInfo ::= 
{
  termAlias 
  {
e164 : "011916745738932",
h323-ID : {"PSTN"}
  }
  gkID {"BBGK"}
  gateways 
  {
{
  gwType voip : NULL
  gwAlias 
  {
e164 : "011916745738932",
h323-ID : {"PSTN"}
  }
  sigAddress 
  {
ip '0A010505'H
port 1720
  }
  resources 
  {
maxDSPs 0
inUseDSPs 0
maxBChannels 0
inUseBChannels 0
activeCalls 0
bandwidth 0
inuseBandwidth 0
  }
}
  }
}
 
Jul  7 18:08:51.895: H225 NONSTD OUTGOING ENCODE BUFFER::=
00020700344C49A78A6BC65403005000530054004E06004200420047004B011002070034
4C49A78A6BC65403005000530054004E000A01050506B8
Jul  7 18:08:51.895: 
Jul  7 18:08:51.895: RAS OUTGOING PDU ::=
value RasMessage ::= locationConfirm : 
{
  requestSeqNum 2054
  callSignalAddress ipAddress : 
  {
ip '0A010505'H
port 1720
  }
  rasAddress ipAddress : 
  {
ip '0A010505'H
port 56645
  }
  nonStandardData 
  {
nonStandardIdentifie

Re: [OSL | CCIE_Voice] "Request pending..." when trying to access IPPAservice

[Jeff Price (jeffpric)]

I've seen that a restart of the IPPA service and UCCX Engine under the
Control Center will fix most problems such as this, however I can't say
that I have ever seen that exact status.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co
Sent: Wednesday, July 07, 2010 11:04 AM
To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net
Subject: [OSL | CCIE_Voice] "Request pending..." when trying to access
IPPAservice

 

Hi Guys,

 

I get "Request pending..." when trying to access IPPA service

 

 

URL I used.:
http://100.0.0.11:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp

 

Also all of the names have been changed to IP addresses in Enterprise
param.

 

 

Any idea ?

 

Cheers,

 

Jeremy

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

Re: [OSL | CCIE_Voice] PSTN Gatekeeper

[Jeff Price (jeffpric)]

On the PSTN gatekeeper, do you have an incoming dial-peer and a
translation-pattern to strip the number down?  

 

Also, do a debug gatekeeper main 10 on HQ router to make sure that the
GK is even routing the call.  Then move to the PSTN router and do a
debug h225 asn1 to see the H225 messages and then a debug dialpeer voip
to see whether you dial-peer is matching.

 

Hope this will help you get started in figuring this out,

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mouhammad
Nasser
Sent: Wednesday, July 07, 2010 11:09 AM
To: ipexpert
Subject: [OSL | CCIE_Voice] PSTN Gatekeeper

 

Hi Experts,
 
I am trying to configure PSTN gatekeeper to test Gatekeeper VIA routing,
so I decided to start with a simple scenario where no VIA zone is
introduced, but I am not able to get it working, the dialed string
"011916745738932" arrives to PSTN gatekeeper but with no hope to match,
and the phone never rings
 
 
I hope anyone can help
 
My configuration and debug are below
 
HQ router
 
gatekeeper
 zone local GK ccie.com 142.1.64.254
 zone remote BBGK cisco.com 10.1.5.5 1719
 zone prefix BBGK 011*
 no shutdown
!
++
on PSTN router it is:
 
interface GigabitEthernet0/0.5
 encapsulation dot1Q 5 native
 ip address 10.1.5.5 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id BBGK ipaddr 10.1.5.5 1719
 h323-gateway voip h323-id PSTN
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 10.1.5.5
!
!
gatekeeper
 zone local BBGK cisco.com 10.1.5.5
 zone remote GK ccie.com 142.1.64.254 1719
 zone prefix BBGK 011*
 gw-type-prefix 1#* default-technology
 no shutdown
!
!
the dialed string is registered to gatekeeper:
 
PSTN-RTR#sh gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags 
--- - --- - - 
- 
10.1.5.51720  10.1.5.556645 BBGK  VOIP-GW 
E164-ID: 011916745738932
H323-ID: PSTN
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 1
 
 
the output of debug h225 asn1 is below
 
Jul  7 18:08:51.891: 
Jul  7 18:08:51.891: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest : 
{
  requestSeqNum 2054
  destinationInfo 
  {
dialedDigits : "011916745738932"
  }
  nonStandardData 
  {
nonStandardIdentifier h221NonStandard : 
{
  t35CountryCode 181
  t35Extension 0
  manufacturerCode 18
}
data '828B903011EDB5EBEBA941C3070007018ECA...'H
  }
  replyAddress ipAddress : 
  {
ip '8E0140FE'H
port 1719
  }
  sourceInfo 
  {
h323-ID : {"GK"}
  }
  canMapAlias TRUE
  hopCount 6
}
 
Jul  7 18:08:51.891: H225 NONSTD INCOMING ENCODE BUFFER::=
828B903011EDB5EBEBA941C3070007018ECA401D05010180833407000A01C815820A
211E0034003800330044003800320044004300300030003000300030003000300032
Jul  7 18:08:51.891: 
Jul  7 18:08:51.891: H225 NONSTD INCOMING PDU ::=
value LRQnonStandardInfo ::= 
{
  ttl 6
  nonstd-callIdentifier 
  {
guid '00EDB5EBEBA941C3070007018ECA401D'H
  }
  gatewaySrcInfo 
  {
e164 : "5001"
  }
  h225NonStdSrcCallSignalAddress h225NonStdIpAddress : 
  {
ip '0A01C815'H
port 33290
  }
  h225NonStdSrcendpointIdentifier {"483D82DC0002"}
}
 
Jul  7 18:08:51.895: H225 NONSTD OUTGOING PDU ::=
value LCFnonStandardInfo ::= 
{
  termAlias 
  {
e164 : "011916745738932",
h323-ID : {"PSTN"}
  }
  gkID {"BBGK"}
  gateways 
  {
{
  gwType voip : NULL
  gwAlias 
  {
e164 : "011916745738932",
h323-ID : {"PSTN"}
  }
  sigAddress 
  {
ip '0A010505'H
port 1720
  }
  resources 
  {
maxDSPs 0
inUseDSPs 0
maxBChannels 0
inUseBChannels 0
activeCalls 0
bandwidth 0
inuseBandwidth 0
  }
}
  }
}
 
Jul  7 18:08:51.895: H225 NONSTD OUTGOING ENCODE BUFFER::=
00020700344C49A78A6BC65403005000530054004E06004200420047004B011002070034
4C49A78A6BC65403005000530054004E000A01050506B8
Jul  7 18:08:51.895: 
Jul  7 18:08:51.895: RAS OUTGOING PDU ::=
value RasMessage ::= locationConfirm : 
{
  requestSeqNum 2054
  callSignalAddress ipAddress : 
  {
ip '0A010505'H
port 1720
  }
  rasAddress ipAddress : 
  {
ip '0A010505'H
port 56645
  }
  nonStandardData 
  {
nonStandardIdentifier h221NonStandard : 
{
  t35CountryCode 181
  t35Extension 0
  

[OSL | CCIE_Voice] Intercom Label

[Jeff Price (jeffpric)]

Hey everyone,

 

I have noticed this behavior when configuration IPMA in proxy mode.  

 

When I first configure the Intercoms, they are labeled as I want as
"Manager Intercom" on the Assistant's phone, and "Assistant Intercom" on
the Manager's phone.  Seemingly at random times, the label will switch
to the name of the respective username that the Intercom destination is
associated with.  For instance, one my HQ Phone 2 (the Manager's phone)
will say "Assistant Intercom" and then randomly switch to "hquser1".  


Does anyone know why this behavior occurs?  Is there any way to fix it?
I have reset the phones and one has reverted back to the configured
label, whereas the other has stayed with the username.

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 

 

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

Re: [OSL | CCIE_Voice] Transcoder/Conf MRG

[Jeff Price (jeffpric)]

I would say better to put them in MRGs and then MRGLs.  Although both
would work, its better have control over who can access them.

 

For example - HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG

 

Then create separate MRGLs with the same MRGs in them:

HQ_MRGL - HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG

BR1_MRGL - HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG

 

However, for the exam purposes, it may just be easier to leave out for
time J

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao
Sent: Friday, June 25, 2010 6:06 PM
To: OSL
Subject: [OSL | CCIE_Voice] Transcoder/Conf MRG

 

If I want HQ, BR1, and BR2 all share one HD conference bridge and one HD
transcoder, will it be better if I just leave these resources in the
default null MRG, or assign them into the HQ_MRG, BR1_MRG, and BR2_MRG?

 

Thanks,

 


Bo

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

Re: [OSL | CCIE_Voice] Live Record

[Jeff Price (jeffpric)]

Do you have conferencing resources configured?  

Jeff

-Original Message-
From: ccieid1ot [mailto:ccieid...@gmail.com] 
Sent: Wednesday, June 23, 2010 7:10 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Live Record

Oh, and I have set the Live record pilot in CUE GUI, verified that
voicemail live-record pilot set to 4250.

On Wed, Jun 23, 2010 at 9:09 PM, ccieid1ot  wrote:
> Amy,
>
> SC Ph1 does have an active VM box since I can leave a VM and retrieved it.
>
> Jeff,
>
> Here's my config for Live Record
>
> Telephony-system
> live-record 4250
>
>
>
> ephone-dn 20 oct
> number 4250
> call-forward all 4220
>
>
> dial-peer voice 4220 voip
> destination-pattern 42..
>
>
> blah blah blah
>
> On Wed, Jun 23, 2010 at 2:55 PM, Jeff Price (jeffpric)
>  wrote:
>> I'm not sure for that specific problem.  But I would make sure you have
>> the following:
>> - correct pilot number configured under telephony-service
>> - a dial-peer that forwards all to cue
>> - a working dial-peer for cue
>> - conferencing configured
>> - the cue pilot number configured in cue
>>
>> I'm leaning towards the cue pilot number not being configured.  I
>> believe its on a page called VM Configuration
>>
>> Jeff
>>
>>
>> -Original Message-
>> From: ccie_voice-boun...@onlinestudylist.com
>> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
>> Sent: Wednesday, June 23, 2010 11:22 AM
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] Live Record
>>
>> What would be the problem when trying to use Live Record and I get a,
>> this mailbox is inactive?  Is this a license issue?
>> ___
>> For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Live Record

[Jeff Price (jeffpric)]

I'm not sure for that specific problem.  But I would make sure you have
the following:
- correct pilot number configured under telephony-service
- a dial-peer that forwards all to cue
- a working dial-peer for cue
- conferencing configured
- the cue pilot number configured in cue

I'm leaning towards the cue pilot number not being configured.  I
believe its on a page called VM Configuration

Jeff


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
Sent: Wednesday, June 23, 2010 11:22 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Live Record

What would be the problem when trying to use Live Record and I get a,
this mailbox is inactive?  Is this a license issue?
___
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please visit www.ipexpert.com
___
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Re: [OSL | CCIE_Voice] Streaming MoH from Router Flash

[Jeff Price (jeffpric)]

Hey Mark,

You will need under ccm-manager-fallback the following:

max-ephones 1
max-dn 1
ip source 

Also, under global config add the following command:
ccm-manager music-on-hold bind source 

Hope this helps,

Jeff 
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark
Sent: Wednesday, June 23, 2010 10:40 AM
To: Amy Ryan
Cc: OSL osl
Subject: Re: [OSL | CCIE_Voice] Streaming MoH from Router Flash

Thanks guys, in this case I am not using CME.  BR2 is part of UCM.  The
'problem' is when BR2 puts a call on hold there is silence.  BR1 puts a
call on hold and there is music.  I am essentially trying to accomplish
the same thing with both BR1 and BR2 by streaming from router flash yet
things appear to be the same but not working for BR2.

Amy, I created a dedicated Device Pool (DP_MoH) and Region (REG_MoH).
Hop count is 1 and I have 1 audio source. Since BR1 is working I think
my UCM configuration is good.  Should the call-manager-fallback config
on BR2 use 239.1.1.1 which is what I am using on BR1 or would I need to
increment the IP? 

BR1
call-manager-fallback
 max-conferences 8 gain -6
 transfer-system full-consult
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 10.20.10.1 10.220.10.254

interface Vlan302
 description VOICE
 ip pim dense-mode


BR2
call-manager-fallback
 max-conferences 8 gain -6
 transfer-system full-consult
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 10.30.10.1 10.230.10.254


.1 = Voice VLAN
.254 = Loopback


My Device Pools have the appropriate MRGL assigned. MRGL_BR1 and
MRGL_BR2 have the same MRG called MRG_PUB_MCAST_MoH.  That MRG contains
MOH_2(MOH)[Multicast]

Music on Hold Server belong to DP_MoH, Enable Multicast Audio Sources,
Multicast IP 239.1.1.1, port 16384, increment IP address, Max Hops = 1. 

For REG_MoH I have it set where it lists REG_BR1 G729, REG_BR2 G729.

As I mentioned before, BR1 does in fact work and I cannot isolate the
problem with BR2. 

Thanks for the assistance..

Mark






On Jun 23, 2010, at 10:15 AM, Amy Ryan wrote:

> Mark, 
> 
> A couple quick suggestions
> 
> -put CUCM PUB in a Device pool with a region that is set to g711-only
> -ensure you hop count is set to 1
> -ensure in CUCM you only have one audio source configured
> 
> HTH, 
> Amy
> 
> 
> ---
> Amy Ryan - CCIE #24677 (Voice)
> Technical Instructor - IPexpert, Inc.
> Mailto: ar...@ipexpert.com
> Telephone: +1.810.326.1444
> Live Assistance, Please visit: www.ipexpert.com/chat
> 
> eFax: +1.810.454.0130
> 
> IPexpert is a premier provider of Self-Study Workbooks, Video on
Demand,
> Audio Tools, Online Hardware Rental and Classroom Training for the
Cisco
> CCIE (R&S, Voice, Security & Service Provider) certification(s) with
> training locations throughout the United States, Europe, South Asia
and
> Australia. Be sure to visit our online communities at
> www.ipexpert.com/communities 
and our
> public website at www.ipexpert.com 
> 
> 
> 
>> From: Mark 
>> Date: Wed, 23 Jun 2010 10:04:29 -0700
>> To: OSL osl 
>> Subject: [OSL | CCIE_Voice] Streaming MoH from Router Flash
>> 
>> If PUB is configured for multicast MoH, 239.1.1.1, port 16384,
increment by
>> IP, and I need to stream MoH from router flash on both BR1 and BR2,
I'm having
>> difficulty getting it to work on BR2.  I have an MRG called
PUB_MCAST_MoH and
>> I've assigned it to both MRGL's MRGL_BR1 and MRGL_BR2 which are
assigned to
>> their respective Device Pools.
>> 
>> On the BR1 router, under call-manager-fallback, I have set 'moh
multicast
>> 239.1.1.1 port 16384 route  ' and it's
working. If
>> I repeat the same process on BR2 it doesn't work.  I've verified ip
>> multicasting is set and dense mode is set.  Any suggestions?
>> 
>> Thanks,
>> Mark
>> 
>> ___
>> For more information regarding industry leading CCIE Lab training,
please
>> visit www.ipexpert.com
> 
> 

___
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please visit www.ipexpert.com
___
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Re: [OSL | CCIE_Voice] sdspfarm

[Jeff Price (jeffpric)]

sdspfarm is how CME can register dsp resources.  For SRST if you had
configured the router's dsp resources to register with SUB then PUB, add
3rd statement that would then try the CME address.  Then add sdspfarm
statements under telephony-service to register the resources.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kalyan iyer
Sent: Monday, June 21, 2010 1:03 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] sdspfarm

 

Hi everyone,

 

I am not clear what sdspfarm is used for?

 

 is this different from dspfarm? 

 

Is this what is needed when the site goes into SRST? 

 

Does this dip into the same MIP count that is being used for the
dspfarm? 

 

 

Thanks

Kalyan

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Re: [OSL | CCIE_Voice] + for Calling Number

[Jeff Price (jeffpric)]

You could probably just create a Called Transformation Pattern with
"\+011" and not strip anything.  Because it is more specific it
would override the less-specific \+.!

Jeff

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark
Sent: Monday, June 21, 2010 11:43 AM
To: OSL osl
Subject: [OSL | CCIE_Voice] + for Calling Number

Ok, I'm stuck.. Could use some confirmation about my approach to + on
the Calling Number when dialing to the PSTN. :)

My phone displays the full E.164 in the upper right corner.  If calls to
the PSTN (conveniently) require + to be stripped from Calling Number in
all cases, this is easy because I can assign a CSS Transformation on the
gateway that strips the + from the Calling Number anytime a call
egresses that gateway.  However, if some call types don't require + in
the Calling Number (local, LD) and others do (international) I cannot
use the CSS on the gateway.

So, assuming I do not have the CSS on the gateway to make the
modification, would it be preferable to either create a Route Pattern or
Translation Pattern (however you're building your dialing plan) and use
the Calling Party Transformation within the RP or TP?

For example, lets say at HQ, Local requires a 7 digit calling party
number, LD requires 10 digits, and International requires +1 and 10
digits for Calling Party.  Therefore my Route Pattern (or Translation
Pattern) would look something like this.

Patterns for all 3 call types are 9.[2-9]XX, 9.1[2-9]XX[2-9]XX,
and 9.011!

Partition = PT_HQ

Local
Calling Partying Transformation Mask = XXX

LD
Calling Partying Transformation Mask = XX

International
Calling Partying Transformation Mask = 


I would simply not modify the International RP/TP because my external
phone mask is already +1XX.  If BR1 is has the same requirements
for Local and LD but also International does NOT require +1 in the
Calling Number..


9.[2-9]XX, 9.1[2-9]XX[2-9]XX, and 9.011!

Partition = PT_BR1 

Local
Calling Partying Transformation Mask = XXX

LD
Calling Partying Transformation Mask = 1XX

International
Calling Partying Transformation Mask = 1XX


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please visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] SD-BLF across WAN

[Jeff Price (jeffpric)]

Thank you Amy!  I am in the process of configuring a mock lab and will
attempt this later. 

 

From: Amy Ryan [mailto:ar...@ipexpert.com] 
Sent: Friday, June 18, 2010 12:24 PM
To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SD-BLF across WAN

 

Jeff, 

I am unsure if it is well documented anywhere, however in order to get
this to work below is an overview.

CUCM
-SIP Trunk Security Profile that has "Accept Presence Subscription"
enabled
-Assign that to a SIP trunk using BR2 IP as the Destination Address and
ensure proper SUBSCRIBE CSS is applied (will need to see RP to BR2
Phone)
-Add RP for BR2 phone using SIP Trunk

BR2-RTR

Enable Presence and add "allow watch" to the DN

HTH, 
Amy


---
Amy Ryan - CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipexpert.com/chat <
http://www.ipexpert.com/chat> 
eFax: +1.810.454.0130 

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




____

From: "Jeff Price (jeffpric)" 
Date: Fri, 18 Jun 2010 13:47:46 -0500
To: 
Subject: [OSL | CCIE_Voice] SD-BLF across WAN

Hi everyone,
 
I am having trouble finding any information regarding how to configure
this scenario (if it is even possible).
 
There is an HQ Phone that needs to have a SD-BLF for a phone that is
located at BR2 with CME.
 
I know that CME offers a way to configure external access to phones
using the "watcher all" command and a "server " under
presence.
 
Is anyone aware of any good documentation for how to accomplish such a
scenario?  Do you know how to configure CUCM to allow this?
 
Thanks in advance for your help,
 

   
   
  Jeff Price
 Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com <mailto:jeffp...@cisco.com> 
 Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
 San Jose, CA 95134-1706
 USA
 Cisco home page <http://www.cisco.com/>  
  

   

 

 
  
 Think before you print.
 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 
Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 
recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 
delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html <
http://www.cisco.com/web/about/doing_business/legal/cri/index.html> 





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please visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] CCIE Voice #26244

[Jeff Price (jeffpric)]

Congrats Ashar!

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar
Siddiqui
Sent: Friday, June 18, 2010 11:46 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CCIE Voice #26244

 

Hello all,

I went to Brussels yesterday and just an hour before learned that I am
now officially CCIE Voice. It was my 2nd attempt but it was worth it.
I learned a lot from my first attempt and it helped me build a better
strategy for the 2nd.

I am thankful to this wonderful list and IPExpert material which I used.
Special thanks to Amy Ryan for her help whenever I needed.
I am also grateful to my Study Partner Iwan Hoogendoorn, a triple CCIE
and I was so lucky to have him as Study partner. I will never forget the
way he use to make daily schedules and strictly made me follow those
otherwise I am a lazy man..this number is for you Iwan!

Few take home points for all those who will be making an attempt in
coming days:

 1 - Read the lab CAREFULLY (I made it Caps for a reason)..every word in
a question is there for a reason!
 2 - Do not rush! the mistakes you will make in first one hour will
haunt you in the entire lab (unless you are lucky to figure out what
went wrong)
 3 - Do not spend too much time if something is not working - you can
always come back to it.
 4 - Note down sections and task which you are working and cross them as
soon as you have completed it
 5 - Call routing - This is how I did it, not necessarily helpful for
you, I did call routing on a page first as what I am going to do at RL
level, Pattern level etc..I configured everything first and then tested
it one by one..took me 30 minutes to finish call routing
 6 - Test everything you have done at least twice and as if it was
configured by someone else and you are the proctor..I found one mistake
while doing my 2nd check
 7 - Save your config often, make sure before you leave that all
gateways are up and registered to CUCM.

I joined this list for my CCIE studies when I started my CCIE journey
back in December 2009 but now I have decided to stick with it as I won't
find such a nice bunch of people anywhere..

N.B: Above all, I loved my number..Digit '4' is my lucky number and
Cisco made sure that I have enough of them..  :)

Thank you all. It's party time now ;)

Ashar Siddiqui
CCIE#26244 (Voice)

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[OSL | CCIE_Voice] SD-BLF across WAN

[Jeff Price (jeffpric)]

Hi everyone,

 

I am having trouble finding any information regarding how to configure
this scenario (if it is even possible).

 

There is an HQ Phone that needs to have a SD-BLF for a phone that is
located at BR2 with CME.

 

I know that CME offers a way to configure external access to phones
using the "watcher all" command and a "server " under
presence.

 

Is anyone aware of any good documentation for how to accomplish such a
scenario?  Do you know how to configure CUCM to allow this?

 

Thanks in advance for your help,

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 

 

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Re: [OSL | CCIE_Voice] Better Voice Lab Locations

[Jeff Price (jeffpric)]

I believe it depends on your location, but normally they walk you to a
local Cisco cafeteria with a voucher for your lunch (up to a certain
price).

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992
Sent: Friday, June 11, 2010 4:10 AM
To: Amp; ccie voice
Cc: ccie_voice@onlinestudylist.com; Mouhammad Nasser
Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations

During lunch are we stuck in the lab area or can we go and buy?

--
From: "Amp" 
Sent: Thursday, June 10, 2010 11:01 PM
To: "ccie voice" 
Cc: ; "Mouhammad Nasser" 

Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations

> No not based on lunch. With the longer lunch time I will be able to
have 
> some time to think about what I have completed, what I need to
complete, 
> and if I need to change anything that I have done.
>
> Quoting ccie voice :
>
>> @Amp
>>
>> So you choose a lab location based on lunch?
>>
>> On Thu, Jun 10, 2010 at 1:14 PM, Amp  wrote:
>>
>>> I live here in the RTP area but have decided to take the lab in San 
>>> Jose.
>>> Here are my reasons:
>>>
>>> 1. Later Start Time
>>> 2. Longer Lunch
>>> 3. Better Weather
>>> 4. Just have a gut feeling about SJC
>>>
>>> Amp
>>>
>>>
>>> Quoting Jeff Garvas :
>>>
>>>  I heard that the West coast facility starts later, so someone east
of 
>>> that
 location would gain the time zone benefits as well as the late
start.
 RTP
 supposedly starts first thing in the morning bright and early.

 2010/6/9 Mouhammad Nasser 

   Hi,
>
> I think it is better to take one that is closest to one's
timezone! 
> this
> will eliminate the factor of travel sickness, and one may go to
exam
> awake
> enough!
>
>
>
> Regards,
>
> --
> Hotmail: Trusted email with powerful SPAM protection. Sign up
now.<
> https://signup.live.com/signup.aspx?id=60969>
>
>
> ___
> 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
> 
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please visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Hi everyone,

Does anyone have any other ideas?  Have you seen this before?  It's
something with the Tech prefix and/or gateway selection in GK, but I'm
not figuring this one out.  The correct digits are sent, CUCM (according
to DNA) should process the call right, and the right codecs should be
negotiated.  I'm lost.  Google isn't helping much either.

Please help!  Thanks,

Jeff

-Original Message-----
From: Jeff Price (jeffpric) 
Sent: Tuesday, May 25, 2010 4:33 PM
To: Jeff Price (jeffpric); Beck, Ken; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

I figured out what I was doing wrong on CME at least.

The command should be:
number 3001 secondary 442321313001 no-reg both

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Tuesday, May 25, 2010 4:26 PM
To: Beck, Ken; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

Hi Ken,

I actually tried doing that on CME, but it didn't work.  After your
email I tried again and I got the same result.

I just set the Significant Digits to 4 and that didn't work either.  I
have the CSS on the trunk configured with a Translation Pattern that
will strip the incoming 1#1775201.1001 to 1001.  The CSS of the
Translation Pattern has a CSS that can reach the phones.  When I check
the trunk using DNA, the call should route correctly.  Therefore, I
believe it's not even reaching CUCM.  I think that something is going
wrong on the GK and the call is never even making it to the CUCM routing
logic.

Thanks for the response Ken.

Any other suggestions?

Jeff


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Beck, Ken
Sent: Tuesday, May 25, 2010 4:03 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

Jeff on your CME ephone-dn's you'll need to put no-reg both after the
number assignment

Did you set the DDI on the GK Trunk to 4 or is it set to all.  Try
setting it to 4.

Also please send a show gatek gw

Regards,
Ken 


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Tuesday, May 25, 2010 3:54 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 51, Issue 142

Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com

To subscribe or unsubscribe via the World Wide Web, visit
http://onlinestudylist.com/mailman/listinfo/ccie_voice
or, via email, send a message with subject or body 'help' to
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You can reach the person managing the list at
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."


Today's Topics:

   1. Re: I Passed CCIE#26088!!! (kerboute kerboute)
   2. Re: I Passed CCIE#26088!!! (kerboute kerboute)
   3. Re: I Passed CCIE#26088!!! (Ashar Siddiqui)
   4. Gatekeeper Issue (Jeff Price (jeffpric))


--

Message: 1
Date: Tue, 25 May 2010 23:21:09 +0100
From: kerboute kerboute 
Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!!
To: ccie_voice@onlinestudylist.com
Message-ID: <4bfc4d55.1060...@cbi.ma>
Content-Type: text/plain; charset="iso-8859-1"

Congratulations brother

On 05/25/2010 11:18 PM, Ehab Salem wrote:
>
> Dear Group,
>
> I Passed from the first shot Really thanks a lot for all your 
> help...I really learned a lot from this kind study list J
>
> All what I want to say about my experience: the exam is easier than 
> what we have in Volume 2...so it's all about Time Management, Strategy

> and Plan. I finished the lab in almost 6 hours. And spent the rest of 
> time revising my configuration.
>
> I spent the week before the exam practicing on time management and 
> putting a strategy and plan for each part in the exam that may 
> comeand before the exam you should sleep well to start the exam 
> with your full performance and energy.
>
> Anyway, it's over now for me...and wish u all the best J
>
> Thanks and best regards,
>
> * *
>
> *E**HAB **S**ALEM*
>
> Cisco Instructor | Sigma IT -- Egypt
>
>
> ___
> For more information regarding industry leading CCIE Lab training,
please visit www.ipexpert.com
>

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-

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

[Jeff Price (jeffpric)]

I figured out what I was doing wrong on CME at least.

The command should be:
number 3001 secondary 442321313001 no-reg both

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Tuesday, May 25, 2010 4:26 PM
To: Beck, Ken; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

Hi Ken,

I actually tried doing that on CME, but it didn't work.  After your
email I tried again and I got the same result.

I just set the Significant Digits to 4 and that didn't work either.  I
have the CSS on the trunk configured with a Translation Pattern that
will strip the incoming 1#1775201.1001 to 1001.  The CSS of the
Translation Pattern has a CSS that can reach the phones.  When I check
the trunk using DNA, the call should route correctly.  Therefore, I
believe it's not even reaching CUCM.  I think that something is going
wrong on the GK and the call is never even making it to the CUCM routing
logic.

Thanks for the response Ken.

Any other suggestions?

Jeff


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Beck, Ken
Sent: Tuesday, May 25, 2010 4:03 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

Jeff on your CME ephone-dn's you'll need to put no-reg both after the
number assignment

Did you set the DDI on the GK Trunk to 4 or is it set to all.  Try
setting it to 4.

Also please send a show gatek gw

Regards,
Ken 


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Tuesday, May 25, 2010 3:54 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 51, Issue 142

Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com

To subscribe or unsubscribe via the World Wide Web, visit
http://onlinestudylist.com/mailman/listinfo/ccie_voice
or, via email, send a message with subject or body 'help' to
ccie_voice-requ...@onlinestudylist.com

You can reach the person managing the list at
ccie_voice-ow...@onlinestudylist.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."


Today's Topics:

   1. Re: I Passed CCIE#26088!!! (kerboute kerboute)
   2. Re: I Passed CCIE#26088!!! (kerboute kerboute)
   3. Re: I Passed CCIE#26088!!! (Ashar Siddiqui)
   4. Gatekeeper Issue (Jeff Price (jeffpric))


--

Message: 1
Date: Tue, 25 May 2010 23:21:09 +0100
From: kerboute kerboute 
Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!!
To: ccie_voice@onlinestudylist.com
Message-ID: <4bfc4d55.1060...@cbi.ma>
Content-Type: text/plain; charset="iso-8859-1"

Congratulations brother

On 05/25/2010 11:18 PM, Ehab Salem wrote:
>
> Dear Group,
>
> I Passed from the first shot Really thanks a lot for all your 
> help...I really learned a lot from this kind study list J
>
> All what I want to say about my experience: the exam is easier than 
> what we have in Volume 2...so it's all about Time Management, Strategy

> and Plan. I finished the lab in almost 6 hours. And spent the rest of 
> time revising my configuration.
>
> I spent the week before the exam practicing on time management and 
> putting a strategy and plan for each part in the exam that may 
> comeand before the exam you should sleep well to start the exam 
> with your full performance and energy.
>
> Anyway, it's over now for me...and wish u all the best J
>
> Thanks and best regards,
>
> * *
>
> *E**HAB **S**ALEM*
>
> Cisco Instructor | Sigma IT -- Egypt
>
>
> ___
> For more information regarding industry leading CCIE Lab training,
please visit www.ipexpert.com
>

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--

Message: 2
Date: Tue, 25 May 2010 23:21:25 +0100
From: kerboute kerboute 
Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!!
To: esa...@sigma-it.net
Cc: ccie_voice@onlinestudylist.com
Message-ID: <4bfc4d65.8060...@cbi.ma>
Content-Type: text/plain; charset="iso-8859-1"

Congratulations brother

On 05/25/2010 11:18 PM, Ehab Salem wrote:
>
> Dear Group,
>
> I Passed from the first shot Really thanks a lot for all your 
> help...I really learned a lot from this kind study list J
>
> All what I want to say about my experience: the exam is easier than 
> 

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

[Jeff Price (jeffpric)]

Hi Ken,

I actually tried doing that on CME, but it didn't work.  After your
email I tried again and I got the same result.

I just set the Significant Digits to 4 and that didn't work either.  I
have the CSS on the trunk configured with a Translation Pattern that
will strip the incoming 1#1775201.1001 to 1001.  The CSS of the
Translation Pattern has a CSS that can reach the phones.  When I check
the trunk using DNA, the call should route correctly.  Therefore, I
believe it's not even reaching CUCM.  I think that something is going
wrong on the GK and the call is never even making it to the CUCM routing
logic.

Thanks for the response Ken.

Any other suggestions?

Jeff


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Beck, Ken
Sent: Tuesday, May 25, 2010 4:03 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142

Jeff on your CME ephone-dn's you'll need to put no-reg both after the
number assignment

Did you set the DDI on the GK Trunk to 4 or is it set to all.  Try
setting it to 4.

Also please send a show gatek gw

Regards,
Ken 


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Tuesday, May 25, 2010 3:54 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 51, Issue 142

Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com

To subscribe or unsubscribe via the World Wide Web, visit
http://onlinestudylist.com/mailman/listinfo/ccie_voice
or, via email, send a message with subject or body 'help' to
ccie_voice-requ...@onlinestudylist.com

You can reach the person managing the list at
ccie_voice-ow...@onlinestudylist.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."


Today's Topics:

   1. Re: I Passed CCIE#26088!!! (kerboute kerboute)
   2. Re: I Passed CCIE#26088!!! (kerboute kerboute)
   3. Re: I Passed CCIE#26088!!! (Ashar Siddiqui)
   4. Gatekeeper Issue (Jeff Price (jeffpric))


--

Message: 1
Date: Tue, 25 May 2010 23:21:09 +0100
From: kerboute kerboute 
Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!!
To: ccie_voice@onlinestudylist.com
Message-ID: <4bfc4d55.1060...@cbi.ma>
Content-Type: text/plain; charset="iso-8859-1"

Congratulations brother

On 05/25/2010 11:18 PM, Ehab Salem wrote:
>
> Dear Group,
>
> I Passed from the first shot Really thanks a lot for all your 
> help...I really learned a lot from this kind study list J
>
> All what I want to say about my experience: the exam is easier than 
> what we have in Volume 2...so it's all about Time Management, Strategy

> and Plan. I finished the lab in almost 6 hours. And spent the rest of 
> time revising my configuration.
>
> I spent the week before the exam practicing on time management and 
> putting a strategy and plan for each part in the exam that may 
> comeand before the exam you should sleep well to start the exam 
> with your full performance and energy.
>
> Anyway, it's over now for me...and wish u all the best J
>
> Thanks and best regards,
>
> * *
>
> *E**HAB **S**ALEM*
>
> Cisco Instructor | Sigma IT -- Egypt
>
>
> ___
> For more information regarding industry leading CCIE Lab training,
please visit www.ipexpert.com
>

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--

Message: 2
Date: Tue, 25 May 2010 23:21:25 +0100
From: kerboute kerboute 
Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!!
To: esa...@sigma-it.net
Cc: ccie_voice@onlinestudylist.com
Message-ID: <4bfc4d65.8060...@cbi.ma>
Content-Type: text/plain; charset="iso-8859-1"

Congratulations brother

On 05/25/2010 11:18 PM, Ehab Salem wrote:
>
> Dear Group,
>
> I Passed from the first shot Really thanks a lot for all your 
> help...I really learned a lot from this kind study list J
>
> All what I want to say about my experience: the exam is easier than 
> what we have in Volume 2...so it's all about Time Management, Strategy

> and Plan. I finished the lab in almost 6 hours. And spent the rest of 
> time revising my configuration.
>
> I spent the week before the exam practicing on time management and 
> putting a strategy and plan for each part in the exam that may 
> comeand before the exam you should sle

[OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Hi everyone,

 

I am having trouble with my GK.  I have made Bold what is the problem,
but I can't seem to understand why I'm having this issue.  I configured
a tech-prefix of 1# under the Trunk configuration page.

 

 

 

 

 

Here is the config -

gatekeeper

 zone local ZONE_1 asccie.com 10.5.200.1

 zone prefix ZONE_1 1* gw-priority 10 CUCM_GK_TRUNK_2

 zone prefix ZONE_1 1* gw-priority 9 CUCM_GK_TRUNK_1

 zone prefix ZONE_1 1* gw-priority 0 BR2_R3_GW BR1_R2_GW

 zone prefix ZONE_1 44* gw-priority 10 BR2_R3_GW

 zone prefix ZONE_1 44* gw-priority 0 BR1_R2_GW CUCM_GK_TRUNK_2
CUCM_GK_TRUNK_1

 gw-type-prefix 1#* default-technology

 no shutdown

 

 

 

 

Here is the debug gatekeeper main 10 output:

May 25 23:55:58.011: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup

May 25 23:55:58.187: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup

R1(config-gk)#

May 25 23:56:00.115: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup

May 25 23:56:00.115: ////GK/gk_rassrv_arq:
arqp=0x4AE0FB04,crv=0x19, answerCall=0

May 25 23:56:00.115: ////GK/gk_rassrv_sep_arq:
ARQ Didn't use GK_AAA_PROC

May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_dns_query: No
Name servers

May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo:
(1#17752011001) Matched tech-prefix 1#

May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo:
(1#17752011001) Matched zone prefix 1 and remainder 7752011001

May 25 23:56:00.115:
////GK/gk_rassrv_get_ingress_network: ARQ
non-std ingress network = 1

May 25 23:56:00.115:
//E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check
the source side, src_zonep=0x4AE06200

May 25 23:56:00.115:
//E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone
is ZONE_1, and z_invian

R1(config-gk)#amelen=0

May 25 23:56:00.115:
//E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check
the destination side, dst_zonep=0x4AE06200

May 25 23:56:00.115:
//E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone
is ZONE_1, and z_outvianamelen=0

May 25 23:56:00.115:
////GK/gk_rassrv_get_ingress_network: ARQ
non-std ingress network = 1

May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo:
(1#17752011001) tech-prefix gateway selection failed.

May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_rassrv_sep_arq:
rassrv_get_addrinfo() failed (return code = 0x103)

 

 

 

 

Here is the show gatekeeper call 10 output:

May 26 00:02:15.899: ////GK/gk_call_new:
src_endptp=0x4AE0F9F0, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160,
crv=31, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x4925F3F8

May 26 00:02:15.899: ////GK/gk_call_find_endpts:
NOT_FOUND

May 26 00:02:15.899: ////GK/gk_call_new:
checking for default (CLI) carrier for sep endpt 0x4AE0F9F0

May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete:
callp=4AB57F54

May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete:
c_callstate 0x0, c_resbw1 0, resbw2 0, c_reszp1 0x0, c_reszp2 0x0

 

 

 

 

Here is the show gatekeeper endpoints output:

R1(config-gk)#do show gatekeeper end

GATEKEEPER ENDPOINT REGISTRATION



CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags 

--- - --- - - 
- 

10.5.201.1  1720  10.5.201.1  61751 ZONE_1VOIP-GW 

H323-ID: BR1_R2_GW

Voice Capacity Max.=  Avail.=  Current.= 0

10.5.202.1  1720  10.5.202.1  52635 ZONE_1VOIP-GW 

H323-ID: BR2_R3_GW

E164-ID: 3001

E164-ID: 3002

Voice Capacity Max.=  Avail.=  Current.= 0

172.21.51.204   37257 172.21.51.204   32858 ZONE_1TERM

H323-ID: CUCM_GK_TRUNK_1

172.21.51.205   34279 172.21.51.205   32814 ZONE_1TERM

H323-ID: CUCM_GK_TRUNK_2

Total number of active registrations = 4

 

(The reason why 3001 and 3002 are registering with GK is the fact that I
am using the secondary command on CME.  For some reason that is still
letting 3001/3002 register with the GK).

 

Thanks in advance for your help!

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for 

Re: [OSL | CCIE_Voice] Unable to Bind L3 to CCM

[Jeff Price (jeffpric)]

Thanks guys!  (Slaps forehead)

 

From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] 
Sent: Friday, May 21, 2010 4:01 PM
To: Jeff Price (jeffpric); CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Unable to Bind L3 to CCM

 

You forgot "service mgcp" at the end of your pri-group command in controller 
config. 

Sent from my Verizon Wireless Phone

- Reply message -----
From: "Jeff Price (jeffpric)" 
Date: Fri, May 21, 2010 6:55 pm
Subject: [OSL | CCIE_Voice] Unable to Bind L3 to CCM
To: "CCIE Voice Maillist" 

Hey everyone,



Have you ever seen a situation where you can register a MGCP GW to CUCM
but you are unable to bind L3 to CCM in IOS?



Here's what I see:



R1(config-if)#isdn bind-l3 ?

 q931  Select IOS Q.931



R1(config-if)#   



Here is my config:



R1#show run

Building configuration...





Current configuration : 3127 bytes

!

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

service password-encryption

!

hostname R1

!

boot-start-marker

boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-24.T.bin

boot-end-marker

!

logging message-counter syslog

enable password 7 110A1A0C12

!

no aaa new-model

network-clock-participate wic 0 

!

dot11 syslog

ip source-route

!

! 

ip cef

ip dhcp excluded-address 10.5.200.1

!

ip dhcp pool HQ_PHONES

  network 10.5.200.0 255.255.255.0

  option 150 ip 172.21.51.204 

  default-router 10.5.200.1 

!

!

no ipv6 cef

!

multilink bundle-name authenticated

!

!

!

!

isdn switch-type primary-ni

!

!

!

!

!

! 

!

!

!

!

!

!

!

!

!

!

!

!

!

voice-card 0

!

!

!

!

!

archive

log config

 hidekeys

! 

!

!

!

!

controller T1 0/0/0

pri-group timeslots 1-3,24

!

controller T1 0/0/1

!

!

!

!

!

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

no ip address

duplex auto

speed auto

!

interface GigabitEthernet0/0.102

encapsulation dot1Q 102 native

ip address 172.21.51.196 255.255.255.224

! 

interface GigabitEthernet0/0.150

encapsulation dot1Q 150

ip address 10.5.100.1 255.255.255.0

!

interface GigabitEthernet0/0.250

encapsulation dot1Q 250

ip address 10.5.200.1 255.255.255.0

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface Serial0/0/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-ni

isdn incoming-voice voice

isdn outgoing display-ie

no cdp enable

!

interface Serial0/1/0

no ip address

encapsulation frame-relay IETF

tx-ring-limit 128

tx-queue-limit 128

serial restart-delay 0

frame-relay lmi-type ansi

!

interface Serial0/1/0.1 point-to-point

ip address 162.5.101.1 255.255.255.0

ip ospf mtu-ignore

frame-relay interface-dlci 201   

!

interface Serial0/1/0.2 point-to-point

ip address 162.5.102.1 255.255.255.0

ip ospf mtu-ignore

frame-relay interface-dlci 202   

!

router ospf 1

log-adjacency-changes

network 10.5.100.0 0.0.0.255 area 0

network 10.5.200.0 0.0.0.255 area 0

network 162.5.101.0 0.0.0.255 area 0

network 162.5.102.0 0.0.0.255 area 0

network 172.5.100.0 0.0.0.255 area 0

network 172.21.51.0 0.0.0.255 area 0

!

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 172.21.51.193

ip http server

no ip http secure-server

!

!

!

nls resp-timeout 1

cpd cr-id 1

!

!

!

!

!

!

control-plane

!

!

!

voice-port 0/0/0:23

!

ccm-manager redundant-host 172.21.51.204

ccm-manager mgcp

!

mgcp

mgcp call-agent 172.21.51.205 service-type mgcp version 0.1

mgcp fax t38 ecm

mgcp bind control source-interface GigabitEthernet0/0.250

mgcp bind media source-interface GigabitEthernet0/0.250

!

mgcp profile default

!

!

!

!

!

!

gatekeeper

shutdown

!

line con 0

exec-timeout 0 0

logging synchronous

terminal-type mon

line aux 0

line vty 0 4

exec-timeout 0 0

password 7 110A1A0C12

logging synchronous

login

terminal-type mon

!

scheduler allocate 2 1000

end 





Here is the status of CCM registration:

R1(config)#do show ccm

MGCP Domain Name: R1

PriorityStatus   Host



Primary Registered   172.21.51.205

First BackupBackup Ready 172.21.51.204

Second Backup   None 



Current active Call Manager:172.21.51.205

Backhaul/Redundant link port:   2428

Failover Interval:  30 seconds

Keepalive Interval: 15 seconds

Last keepalive sent:00:03:46 UTC May 22 2010 (elapsed time:
00:00:06)

Last MGCP traffic time: 00:03:46 UTC May 22 2010 (elapsed time:
00:00:06)

Last failover time: 23:59:51 UTC May 21 2010 from
(172.21.51.205)

Last switchback time:   00:00:21 UTC May 22 2010 from
(172.21.51.204)

Switchback mode:Graceful

MGCP Fallback mode: Not Selected

Last MGCP Fallback start time:  None

Last

Re: [OSL | CCIE_Voice] Problem ISDN Lab 5C Vol1

[Jeff Price (jeffpric)]

Hi Naoufal,

Ensure that you have enough DSP resources to complete the call.  Try
decreasing the number of pri timeslots if you are unsure.

Jeff

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kerboute
kerboute
Sent: Tuesday, May 11, 2010 3:51 PM
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] Problem ISDN Lab 5C Vol1

Hi,

I'm working on lab 5C Vol1 and I'm having a troube with the HQ T1, when 
I make a call to 911 I've got the message below:

May 12 02:46:26.355: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid 
type/plan 0x0 0x0 may be overriden; sw-type 13
May 12 02:46:26.355: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type

0xD is 0x2 0x1, Calling num 2123945003
May 12 02:46:26.359: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type

0xD is 0x0 0x0, Called num 911
May 12 02:46:26.359: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref =

0x0089
 Bearer Capability i = 0x8090A2
 Standard = CCITT
 Transfer Capability = Speech
 Transfer Mode = Circuit
 Transfer Rate = 64 kbit/s
 Channel ID i = 0xA98381
 Exclusive, Channel 1
 Calling Party Number i = 0x2181, '2123945003'
 Plan:ISDN, Type:National
 Called Party Number i = 0x80, '911'
 Plan:Unknown, Type:Unknown
May 12 02:46:26.391: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8  
callref = 0x8089
 Channel ID i = 0xA98381
 Exclusive, Channel 1
May 12 02:46:26.403: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8  
callref = 0x8089
 Progress
HQ-RTR#Ind i = 0x8188 - In-band info or appropriate now available
May 12 02:46:26.443: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8  
callref = 0x0089
 Cause i = 0x80AC - Requested circuit/channel not available
May 12 02:46:26.451: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref

= 0x8089
May 12 02:46:26.455: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  
callref = 0x0089

However I can see the all 3 channel are up and T1 are 
multiframe-established.

Any Idea?

Thank you
Naoufal
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Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

[Jeff Price (jeffpric)]

This is true.  I forgot about this.  Normally it has to do with
oversubscribing the DSP resources.  Meaning, make sure you aren't using
all of them and the PSTN has enough to use.  Try decreasing the amount
of channels you create under the pri-group timeslots command.  Good
point.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010
Sent: Thursday, May 06, 2010 12:39 PM
To: Steve Denney (stdenney)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

 

cause 0x22 - generaly means no channel availableplease checj you
have right number of slots defined in "pri-group timeslots" statement
and double check your PSTN router for same.

On Thu, May 6, 2010 at 12:26 PM, Steve Denney (stdenney)
 wrote:

Hi,

 

Seeing some errors today that I haven't encountered before in any other
lab...wh! :)

 

I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from
the PSTN working into HQ. 

Pretty straightforward stuff, except the calls never seem to get out of
the PSTN router. 

 

When dialing the HQ phone from the PSTN phone (regardless of line
selected), I get the following debug isdn q931 errors from the PSTN
router:

May  6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Calling num 911

May  6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails
with cause 0x22

 

And every 30 seconds, I see the same batch of 4 ISDN Restart messages,
like this (also from the PSTN router):

May  6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

May  6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

Show isdn status on the PSTN router looks normal for this interface:

ISDN Serial0/3/0:23 interface

*** Network side configuration *** 

dsl 1, interface ISDN Switchtype = primary-ni

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

0 Active Layer 3 Call(s)

Active dsl 1 CCBs = 0

The Free Channel Mask:  0x8000

Number of L2 Discards = 0, L2 Session ID = 0

 

 

Attaching show run and show isdn status as well for the HQ router (the
other end) just for troubleshooting completeness, but there's no
indication of anything amiss, nor any debug messages at all, on the HQ
router. The call never gets that far.

 

I started this morning on Voice Pod 11 and hit this. Ryan was kind
enough to move me over to Voice Pod 16, but I'm hitting the same issue
here. 

OSL archive and Google search turned up nothing concrete, other than a
general theme of "it sounds like your telco / carrier has issues."  :)

 

Any ideas?

 

Cheers and TIA, sd

 


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please visit www.ipexpert.com  

 

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Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

[Jeff Price (jeffpric)]

To fix this problem in the past, I simply restarted the PSTN router and
it works.  However, I do realize that this isn't ideal in the real world
or the exam and I was never able to figure out the "proper" way to fix
this.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Thursday, May 06, 2010 12:27 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

 

Hi,

 

Seeing some errors today that I haven't encountered before in any other
lab...wh! :)

 

I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from
the PSTN working into HQ. 

Pretty straightforward stuff, except the calls never seem to get out of
the PSTN router. 

 

When dialing the HQ phone from the PSTN phone (regardless of line
selected), I get the following debug isdn q931 errors from the PSTN
router:

May  6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Calling num 911

May  6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails
with cause 0x22

 

And every 30 seconds, I see the same batch of 4 ISDN Restart messages,
like this (also from the PSTN router):

May  6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

May  6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX -> RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

Show isdn status on the PSTN router looks normal for this interface:

ISDN Serial0/3/0:23 interface

*** Network side configuration *** 

dsl 1, interface ISDN Switchtype = primary-ni

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

0 Active Layer 3 Call(s)

Active dsl 1 CCBs = 0

The Free Channel Mask:  0x8000

Number of L2 Discards = 0, L2 Session ID = 0

 

 

Attaching show run and show isdn status as well for the HQ router (the
other end) just for troubleshooting completeness, but there's no
indication of anything amiss, nor any debug messages at all, on the HQ
router. The call never gets that far.

 

I started this morning on Voice Pod 11 and hit this. Ryan was kind
enough to move me over to Voice Pod 16, but I'm hitting the same issue
here. 

OSL archive and Google search turned up nothing concrete, other than a
general theme of "it sounds like your telco / carrier has issues."  :)

 

Any ideas?

 

Cheers and TIA, sd

 

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Re: [OSL | CCIE_Voice] VLAN port speed and duplex

[Jeff Price (jeffpric)]

For the exam, no.  All routing is pre-configured and if there is any
issues you need to get the proctor involved. 

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao
Sent: Thursday, April 22, 2010 10:51 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] VLAN port speed and duplex

 

Hi guys,

 

I am just starting my lab prep, and I am at lab 1A.

 

When a port is combined with both voice and data, do we need to manually
set port speed and duplex(i.e., 10/half) for the exam?

 

Thank you,

 

 

Bo

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Re: [OSL | CCIE_Voice] cBarge on CME not working - says " messageFailed to setup barge"

[Jeff Price (jeffpric)]

You also need to configure some ad-hoc conference ephone-dn(s).  Then
this should work.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010
Sent: Thursday, April 22, 2010 8:29 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] cBarge on CME not working - says "
messageFailed to setup barge"

 

cBarge on CME -

- eph-dn octoline

- conf hardware in telephony service

- privacy off on ephone to be bacrged into

- ephon-temp "remote-in-use" to have cBArge key

- apply eph-temp to eph

 

Error message says " message Failed to setup barge"

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Re: [OSL | CCIE_Voice] OSL CUCM DHCP Server

[Jeff Price (jeffpric)]

This may seem silly, but it slips the mind sometimes.  Did you remember
to configure the switchports with the proper vlans?

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
wilson.sam...@usc-bt.com
Sent: Tuesday, April 13, 2010 12:12 PM
To: bkvalent...@gmail.com; ccie_voice-boun...@onlinestudylist.com;
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] OSL CUCM DHCP Server

Thanks for your response.

Yes, the ip helper address is correct, infact on the router I do see
that phone is looking for a DHCP Server, however the Server seems to be
silent.

I'm really stuck on this and don't seem to get any clue to get out of
this.

It must be something really silly, that I'm missing it

Regards

-Original Message-
From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] 
Sent: Tuesday, April 13, 2010 2:58 PM
To: Samuel, Wilson; ccie_voice-boun...@onlinestudylist.com;
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] OSL CUCM DHCP Server

Have IP helper address pointing to the CUCM Pub IP?

Make sure the default gateway offered is correct.  I won't tell you how
I learned that one.

Brian


Sent via BlackBerry from T-Mobile

-Original Message-
From: 
Date: Tue, 13 Apr 2010 13:55:51 
To: 
Subject: [OSL | CCIE_Voice] OSL CUCM DHCP Server

Hi All,

I'm trying to setup DHCP Server on CUCM (ver 7.0 ), however for some
reason its not offering any ACK or DHCPOFFR for any DHCP Request on the
network.

I have already disabled CSA on the CUCM, anything else I missed?

Appreciate any help on the aspect.

Kind Regards
Wilson Samuel
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Re: [OSL | CCIE_Voice] Inter-site calls Not routing over GK trunk

[Jeff Price (jeffpric)]

Ashar -

 

How is the number coming into the gateway?  Are you stripping the
tech-prefix so that the CUCME  will recognize the number?   You can use
the debug h225 asn1 command (a lot of output) to figure this one how,
however this assumes the GK config is correct.  Otherwise the h225 debug
will not show anything besides registration, because the GK will never
attempt to route the call to CUCME.

 

 For example, the number would come in to CUCME as 1#3001 and you would
need a translation rule to strip the 1# so that CUCME will recognize.
Also, if you are supporting abbreviated dialing from BR2 to HQ and BR1
you will need to prepend the tech-prefix prior to sending and then strip
when CUCM receives it using a translation pattern. 

 

Hope this helps,

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar
Siddiqui
Sent: Monday, April 05, 2010 9:37 AM
To: CCIE Voice OSL (ccie_voice@onlinestudylist.com)
Subject: [OSL | CCIE_Voice] Inter-site calls Not routing over GK trunk
Importance: High

 

Hello guys,

I am trying to route calls between HQ and BR1 to a CME site BR2 using
4-digit extension over the GK trunk.
Gatekeeper is on HQ and its a MGCP gateway.

The solution below is working fine as expected if we don't use "no-reg
primary" at ephone-dn. This means phones are registering with their
E.164 id to gatekeeper. I don't think this should happen. Phones should
not register with Gatekeeper isn't it? I checked show gatekeeper
endpoints and it was showing me E164 id 3001 and E164 id 3002 of BR2.
Everything was working fine, calls can be routed over the GK trunk to
CUCME site when a user dial 3001/3002 from HQ or BR1. 

But as soon as I entered no-reg primary command to the ephone-dn and did
a reset, the GK trunk is not working and calls from HQ and BR1 are
routed thru backup local GW with full E164 number. 
I am confiused a bit because CUCME is regsitered with 1# and GK knows 3*
numbers sits at CUCME site. I also changed 3* to 3... on Gatekeeper but
it isn't working.

My question is what is an expected behaviour? do we need that no-reg
primary command on ephone-dn? if not then the solution is working fine
but if yes we need it on ephone-dn then please let me know where I am
wrong as calls are not routed thru the GK trunk. Do I need  to prefix
anything at CUCM on Route List details-  GK trunk like 1# or something? 

Calls from SC to HQ and SB are working fine as normal over the GK trunk.
I have made a Region GK-729 with 729 codec within all sites. That region
is in the DP of GK-TRUNK.

Here is HQ config:

(My subscriber is down at the moment so you will only see one VOIP-GW in
outputs)

gatekeeper
 zone local GK ccievoice.com 10.10.110.1
 zone local CUCME ccievoice.com
 zone prefix GK 5... gw-priority 10 gk-trunk_2
 zone prefix GK 5... gw-priority 9 gk-trunk_1
 zone prefix GK 1... gw-priority 10 gk-trunk_2
 zone prefix GK 1... gw-priority 9 gk-trunk_1
 zone prefix CUCME 3*
 gw-type-prefix 1#* default-technology
 no shutdown
!

HQ#sh gatek endpoints 
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags 
--- - --- - - 
- 
10.10.110.3 1720  10.10.110.3 60441 GKH323-GW 
H323-ID: CUCME
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.101720  10.10.210.1044782 GKVOIP-GW 
H323-ID: gk-trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 2

HQ#
HQ#sh gatek zone prefix 
  ZONE PREFIX TABLE
  =
GK-NAME   E164-PREFIX
---   ---
GK5...
GK1...
CUCME3*
HQ#




BR2#interface Loopback0
 ip address 10.10.110.3 255.255.255.255
 ip ospf network point-to-point
 h323-gateway voip interface
 h323-gateway voip id GK ipaddr 10.10.110.1 1719
 h323-gateway voip h323-id CUCME
 h323-gateway voip tech-prefix 1#

BR2#sh ephone

ephone-1[0] Mac:0017.9497.1F89 TCP socket:[1] activeLine:0 REGISTERED in
SCCP ver 12/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
caps:8 privacy:1
IP:192.168.10.55 50512 7961  keepalive 42 max_line 6
button 1: dn 1  number 3001 CH1   IDLE CH2   IDLE CH3
IDLE CH4   IDLE CH5   IDLE CH6   IDLE
CH7   IDLE CH8   IDLE 
button 2: dn 3  number 3003 CH1   IDLE CH2   IDLE CH3
IDLE CH4   IDLE CH5   IDLE CH6   IDLE
CH7   IDLE CH8   IDLE shared 
privacy button is enabled
Preferred Codec: g711ulaw 
Username: scph1 


ephone-2[1] Mac:0017.E089.7382 TCP socket:[2] activeLine:0 REGISTERED in
SCCP ver 12/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
caps:8 privacy:1
IP:192.168.10.56 51267 7961  keepalive 179 

Re: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM

[Jeff Price (jeffpric)]

To my knowledge, it is that simple.  Use this command on CUE software
install clean url ftp:///license-filename.pkg user user pass
pass and have the license file available on an FTP server.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter
Sent: Monday, March 22, 2010 4:12 PM
To: osl osl
Subject: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM

 

Is the only requirement to go from CME integration to UCM to load the
proper license file?  This is my companies equipment not proctor labs. I
would like to be able to move back and forth similar to proctor labs but
am unsure it is as easy as just loading the proper license file.  

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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jeff Price (jeffpric)]

Are you by any chance running a VPN from routers to PSTN?  I've noticed
this causes distortion some times.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason
Granat
Sent: Thursday, March 11, 2010 9:46 AM
To: 
Subject: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

 

Perhaps this is something simple that I am overlooking but I have the
generic setup running in my home lab with 3 gateways and one PSTN
router. 2 of the gateways are T1 and one is E1. The PSTN router is also
running CME with a 7960 to simulate PSTN destinations. Calls from any
site to the PSTN phone are fine. Calls between T1 sites are fine. Calls
between T1 and E1 sites are distorted, like the gain is way too high. I
tried playing with the gain on the voice-port but no luck. I'm not
finding much online or in Cisco docs. Any suggestions?

 

Thanks,

 

Jason

 





http://slash128.com

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Re: [OSL | CCIE_Voice] Reset CUCM 7.x to initial config on VM Ware

[Jeff Price (jeffpric)]

Wael,

 

After you do a fresh install, take a snapshot in VMWare.  Then you can
always revert back to the snapshot without having to "un-configure"
everything you have done.

 

In VMWare you can normally only have 1 snapshot, unless you have one of
the more advanced versions, so for all other backups your would have to
use tftp, ftp, and DRS on CUCM in order to save configs and states.

 

Hope this helps,

Jeff 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wael Agina
Sent: Wednesday, March 10, 2010 10:35 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] Reset CUCM 7.x to initial config on VM Ware

 

Dear All,

 

   I have CUCM 7, CU7, CUCCX 7 running on VM ware.

My qeustions is how to reset all its setting to initial setup without
having any config ?

Also if possible how to make a backup from a certain config on each
server.

 

Sorry for simple qeustions but i am new to vmware world and a friend
give me the images.

-- 

Thanks and Best Regards,
Wael Agina

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Re: [OSL | CCIE_Voice] CM to GK BRQ behavior

[Jeff Price (jeffpric)]

Hi Scott,

 

Go to Enterprise parameters and change the "Advertise G722" to false.
Hope this helps.\

 

Jeff

 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo
Sent: Monday, March 08, 2010 12:37 PM
To: Scott ODonnell
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] CM to GK BRQ behavior

 

Hello Scott,

 

What is BRQ??

Am having siimilar issue

how did you combat it.

thanks

On Wed, Jul 8, 2009 at 9:03 PM, Scott ODonnell
 wrote:

I'm seeing something strange in making calls from CM to CME via GK. 

 

I've enabled the BRQ service parameter in CM.

I've included "bandwidth total default 16" in my gk config and did a
shut/no shut

 

When I make calls from CM to CME the deb h225 asn1 shows (I think) that
128k is being requested.

 

Am I missing something obivous here ?

Currently all my calls get rejected from the GK and go via the HQ GW.

If I remove the bandwidth command from the Gatekeeper config, the call
works using g729.

 

 

- Scott

 

 


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Re: [OSL | CCIE_Voice] Phone boot problem

[Jeff Price (jeffpric)]

Hi Chris,

 

SCCP45.8-4-1S is the phone load for the 7945.

 

I am unable to get to the settings page as the phone doesn't boot up far
enough.

 

Thanks,

 

Jeff

 

From: Christopher Clouse [mailto:christopherc_56...@hotmail.com] 
Sent: Tuesday, February 23, 2010 11:33 AM
To: Jeff Price (jeffpric)
Subject: Re: [OSL | CCIE_Voice] Phone boot problem

 

Check in CM under the device defaults for the phone loads and see what
the load it is attempting to grab.  If you can get as far as the
setttings page on the phone you should be able to see it under device
settings.

 

From: Jeff Price (jeffpric) <mailto:jeffp...@cisco.com>  

Sent: Tuesday, February 23, 2010 1:31 PM

To: Christopher Clouse <mailto:christopherc_56...@hotmail.com>  

Subject: RE: [OSL | CCIE_Voice] Phone boot problem

 

CUCM is 7.  How can I find out the software version of the phone?   I
don't believe the boot-up process gets far enough for me to access this
information.   It's worth noting that I factory reset the phones in an
attempt to get them to work.

 

Jeff

 

From: Christopher Clouse [mailto:christopherc_56...@hotmail.com] 
Sent: Tuesday, February 23, 2010 11:29 AM
To: Jeff Price (jeffpric)
Subject: Re: [OSL | CCIE_Voice] Phone boot problem

 

What version of software is on the phone and what is on the CM?

 

~Chris

 

From: Jeff Price (jeffpric) <mailto:jeffp...@cisco.com>  

Sent: Tuesday, February 23, 2010 12:41 PM

To: Jeff Price (jeffpric) <mailto:jeffp...@cisco.com>  ; OSL Group
<mailto:ccie_voice@onlinestudylist.com>  

Subject: Re: [OSL | CCIE_Voice] Phone boot problem

 

Hi everyone,

 

I hooked up a spare 7960 to this port and it has registered
successfully.So what would be up with the 7945s that it can't
register?

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Tuesday, February 23, 2010 10:32 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] Phone boot problem

 

Hi everyone,

 

I have a pair of 7945 phones that will not startup for me.  I have been
using these phones for the past 3 weeks or so and had no problems.  I
have configured DHCP and their switchports as follows:

 

On the 3750 switch that the phones are directly connected to:

interface GigabitEthernet1/0/13

 description HQ PHONE 1

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

!

interface GigabitEthernet1/0/14

 description HQ PHONE 2

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

!

interface GigabitEthernet1/0/15

description HQ PHONE 3

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

 

On the router that supplies the DHCP:

ip dhcp excluded-address 10.5.200.1

ip dhcp pool HQ_PHONES

   network 10.5.200.0 255.255.255.0

   option 150 ip 172.21.51.204 

   default-router 10.5.200.1

 

On port g1/0/13 of the switch I have a 7960 that has successfully
registered with CUCM, but the two 7945 keep loading up to a "Upgrading"
screen and then re-cycling.  The CUCM PUB is running the TFTP service
and the CM Group (for registration time only) is configured to use PUB
then SUB so that auto-registration will work correctly.  Any ideas why
this may be happening?  

 

Thanks,

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page <http://www.cisco.com/> 

 

 

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http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 



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

<><><>___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Phone boot problem

[Jeff Price (jeffpric)]

Amy,

 

I actually have already factory reset the phones and manually added to
the CUCM database.  I will try again though, because I may have made
some sort of mistake while doing so.  Thanks for your help.

 

Jeff

 

From: Amy Ryan [mailto:ar...@ipexpert.com] 
Sent: Tuesday, February 23, 2010 12:06 PM
To: Jeff Price (jeffpric); OSL Group
Subject: Re: [OSL | CCIE_Voice] Phone boot problem

 

Jeff, 

Can you try to manually add a 7945 to the UCM database with whatever the
existing protocol the phone was using in the previous weeks?  It sounds
like the phone is trying to switch firmware and is having problems doing
so.

In the event that this fails,  you could try to restore factory defaults
by using the following steps:

1.  unplug the Ethernet connection (an external power source if
required) 
2.  hold down the # key and while holding, plug in the Ethernet
connection (and power as needed)
3.  When the Line Buttons to the right of the LCD display begin to flash
amber, release the # key and press the numbers on the dialpad in the
following sequence:  123456789*0#

The phone will then initiate the factory reset and should download
intended firmware, etc.  This process can take 4-8 minutes roughly.

HTH, 
Amy

---
Amy Ryan - CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipexpert.com/chat <
http://www.ipexpert.com/chat> 
eFax: +1.810.454.0130 

IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA
(R&S, Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, Voice, Security
& Service Provider) Certification Training with locations throughout the
United States, Europe and Australia. Be sure to check out our online
communities at www.ipexpert.com/communities <
http://www.ipexpert.com/communities>  and our public website at
www.ipexpert.com <http://www.ipexpert.com> .




________

From: "Jeff Price (jeffpric)" 
Date: Tue, 23 Feb 2010 12:41:29 -0600
To: "Jeff Price (jeffpric)" , OSL Group <
ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] Phone boot problem

Hi everyone,
 
I hooked up a spare 7960 to this port and it has registered
successfully.So what would be up with the 7945s that it can't
register?
 
Jeff
 

From: ccie_voice-boun...@onlinestudylist.com [
mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Tuesday, February 23, 2010 10:32 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] Phone boot problem

Hi everyone,
 
I have a pair of 7945 phones that will not startup for me.  I have been
using these phones for the past 3 weeks or so and had no problems.  I
have configured DHCP and their switchports as follows:
 
On the 3750 switch that the phones are directly connected to:
interface GigabitEthernet1/0/13
 description HQ PHONE 1
 switchport access vlan 150
 switchport mode access
 switchport voice vlan 250
 spanning-tree portfast
!
interface GigabitEthernet1/0/14
 description HQ PHONE 2
 switchport access vlan 150
 switchport mode access
 switchport voice vlan 250
 spanning-tree portfast
!
interface GigabitEthernet1/0/15
description HQ PHONE 3
 switchport access vlan 150
 switchport mode access
 switchport voice vlan 250
 spanning-tree portfast
 
On the router that supplies the DHCP:
ip dhcp excluded-address 10.5.200.1
ip dhcp pool HQ_PHONES
   network 10.5.200.0 255.255.255.0
   option 150 ip 172.21.51.204 
   default-router 10.5.200.1
 
On port g1/0/13 of the switch I have a 7960 that has successfully
registered with CUCM, but the two 7945 keep loading up to a "Upgrading"
screen and then re-cycling.  The CUCM PUB is running the TFTP service
and the CM Group (for registration time only) is configured to use PUB
then SUB so that auto-registration will work correctly.  Any ideas why
this may be happening?  
 
Thanks,
 
 
   
   
  Jeff Price
 Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
 Phone: 408-525-8293
Mobile: 408-204-4510   Cisco Systems, Inc.
170 West Tasman Drive,
 San Jose, CA 95134-1706
 USA
 Cisco home page <http://www.cisco.com/>  
  

   

 

 
  
 Think before you print.
 
 
This email may contain confidential and privileged material for the sole
use of the intended recipient. 
Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 
recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 
delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html





___
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] Phone boot problem

[Jeff Price (jeffpric)]

Hi everyone,

 

I hooked up a spare 7960 to this port and it has registered
successfully.So what would be up with the 7945s that it can't
register?

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Tuesday, February 23, 2010 10:32 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] Phone boot problem

 

Hi everyone,

 

I have a pair of 7945 phones that will not startup for me.  I have been
using these phones for the past 3 weeks or so and had no problems.  I
have configured DHCP and their switchports as follows:

 

On the 3750 switch that the phones are directly connected to:

interface GigabitEthernet1/0/13

 description HQ PHONE 1

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

!

interface GigabitEthernet1/0/14

 description HQ PHONE 2

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

!

interface GigabitEthernet1/0/15

description HQ PHONE 3

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

 

On the router that supplies the DHCP:

ip dhcp excluded-address 10.5.200.1

ip dhcp pool HQ_PHONES

   network 10.5.200.0 255.255.255.0

   option 150 ip 172.21.51.204 

   default-router 10.5.200.1

 

On port g1/0/13 of the switch I have a 7960 that has successfully
registered with CUCM, but the two 7945 keep loading up to a "Upgrading"
screen and then re-cycling.  The CUCM PUB is running the TFTP service
and the CM Group (for registration time only) is configured to use PUB
then SUB so that auto-registration will work correctly.  Any ideas why
this may be happening?  

 

Thanks,

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page <http://www.cisco.com/> 

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 

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

[OSL | CCIE_Voice] Phone boot problem

[Jeff Price (jeffpric)]

Hi everyone,

 

I have a pair of 7945 phones that will not startup for me.  I have been
using these phones for the past 3 weeks or so and had no problems.  I
have configured DHCP and their switchports as follows:

 

On the 3750 switch that the phones are directly connected to:

interface GigabitEthernet1/0/13

 description HQ PHONE 1

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

!

interface GigabitEthernet1/0/14

 description HQ PHONE 2

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

!

interface GigabitEthernet1/0/15

description HQ PHONE 3

 switchport access vlan 150

 switchport mode access

 switchport voice vlan 250

 spanning-tree portfast

 

On the router that supplies the DHCP:

ip dhcp excluded-address 10.5.200.1

ip dhcp pool HQ_PHONES

   network 10.5.200.0 255.255.255.0

   option 150 ip 172.21.51.204 

   default-router 10.5.200.1

 

On port g1/0/13 of the switch I have a 7960 that has successfully
registered with CUCM, but the two 7945 keep loading up to a "Upgrading"
screen and then re-cycling.  The CUCM PUB is running the TFTP service
and the CM Group (for registration time only) is configured to use PUB
then SUB so that auto-registration will work correctly.  Any ideas why
this may be happening?  

 

Thanks,

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 

 

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

Re: [OSL | CCIE_Voice] Voice Vol 1 Lab 5c task 5.8

[Jeff Price (jeffpric)]

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
(9011916745738932) Tech-prefix match failed.

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
(9011916745738932) unresolved zone prefix, using source zone US

 

You need to strip the 9011 before sending this to the gatekeeper.  

 zone prefix SPAIN 34*

 zone prefix PSTN-WAN 91*

 

Hope this helps.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
CCIETalk.com
Sent: Tuesday, February 23, 2010 10:10 AM
To: Vik Malhi
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Voice Vol 1 Lab 5c task 5.8

 

Thanks Vik. Here is the debug output

 

Calling India

 

HQ-RTR#

Feb 23 17:45:35.916: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup

Feb 23 17:45:35.916: ////GK/gk_rassrv_arq:
arqp=0x67399F7C,crv=0x4, answerCall=0

Feb 23 17:45:35.920: ////GK/gk_rassrv_sep_arq:
ARQ Didn't use GK_AAA_PROC

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/gk_dns_query: No
Name servers

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
(9011916745738932) Tech-prefix match failed.

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
(9011916745738932) unresolved zone prefix, using source zone US

Feb 23 17:45:35.920:
////GK/gk_rassrv_get_ingress_network: returning
default ingress network = 1

Feb 23 17:45:35.920:
//004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: about to check
the source side, src_zonep=0x67A6DFFC

Feb 23 17:45:35.920:
//004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: matched zone
is US, and 

HQ-RTR#z_invianamelen=0

Feb 23 17:45:35.920:
//004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: about to check
the destination side, dst_zonep=0x67A6DFFC

Feb 23 17:45:35.920:
//004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: matched zone
is US, and z_outvianamelen=0

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
No tech prefix

 

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
Alias not found

 

Feb 23 17:45:35.920:
////GK/gk_rassrv_get_ingress_network: returning
default ingress network = 1

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
(9011916745738932) default-tech gateway selection failed, status = 0x805

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo:
(9011916745738932) unknown address and no default technology defined.

Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/gk_rassrv_sep_arq:
rassrv_get_addrinfo() failed (return code = 0x107)

 

Calling Spain

 

 

Feb 23 17:45:08.956: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup

Feb 23 17:45:08.956: ////GK/gk_rassrv_arq:
arqp=0x6816755C,crv=0x3, answerCall=0

Feb 23 17:45:08.956: ////GK/gk_rassrv_sep_arq:
ARQ Didn't use GK_AAA_PROC

Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/gk_dns_query: No
Name servers

Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo:
(90113432141891) Tech-prefix match failed.

Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo:
(90113432141891) unresolved zone prefix, using source zone US

Feb 23 17:45:08.956:
////GK/gk_rassrv_get_ingress_network: returning
default ingress network = 1

Feb 23 17:45:08.956:
//806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: about to check
the source side, src_zonep=0x67A6DFFC

Feb 23 17:45:08.956:
//806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: matched zone
is US, and z_in

HQ-RTR#vianamelen=0

Feb 23 17:45:08.956:
//806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: about to check
the destination side, dst_zonep=0x67A6DFFC

Feb 23 17:45:08.960:
//806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: matched zone
is US, and z_outvianamelen=0

Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo:
No tech prefix

 

Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo:
Alias not found

 

Feb 23 17:45:08.960:
////GK/gk_rassrv_get_ingress_network: returning
default ingress network = 1

Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo:
(90113432141891) default-tech gateway selection failed, status = 0x805

Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo:
(90113432141891) unknown address and no default technology defined.

Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/gk_rassrv_sep_arq:
rassrv_get_addrinfo() failed (return code = 0x107)

 

Gatekeeper config

 

HQ-RTR#sh run | begin gatekeeper

gatekeeper

 zone local SPAIN ipexpert.com 10.10.110.1

 zone local US ipexpert.com

 zone remote PSTN-WAN ipexpert.com 10.10.100.2 1719

 zone prefix SPAIN 34*

 zone prefix PSTN-

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Hi,

 

I have restored and begun another lab already so I am unable to supply this 
debug.  However, I realized I had the H225 Trunk was assigned to the HQ_DP, 
which used the HQ_REG, which requested G711.  Eventually G729 was negotiated, 
because that's all that the BR2 would allow.  I assigned the trunk to the 
correct DP, and the correct bandwidth was then requested.

 

Jeff

 

From: Angel Perez [mailto:gorr...@hotmail.com] 
Sent: Monday, February 22, 2010 2:17 AM
To: afat...@verizon.net; Jeff Price (jeffpric)
Cc: osl osl
Subject: RE: [OSL | CCIE_Voice] Gatekeeper Issue

 

Hello:
 
If the gk trunk has inbound fast start checked, the gk will ask for 128k of 
bandwith then with a BRQ message it will reduce the bw to 16k, a debug h225 
asn1 would be clarifaing.
 
Jeff: Can you provide the following debug:  deb h225 asn1
 
thx
 
> Date: Fri, 19 Feb 2010 18:48:47 -0800
> From: afat...@verizon.net
> To: jeffp...@cisco.com
> CC: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue
> 
> whats the direction of call? Previous debug has bandwidth 160, which is 
> 16k for g729 and this one is showing bandwidth 1280 which is 128k for 
> g711. So this call leg debug is showing is g711 is being requested somwhere.
> 
> -- Mustafa
> 
> 
> Jeff Price (jeffpric) wrote:
> > Here is the output of command debug gatekeeper call 10:
> >
> > *Feb 20 03:38:14.913: ////GK/gk_call_new:
> > src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160,
> > crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78
> > *Feb 20 03:38:14.913:
> > ////GK/gk_call_find_endpts: NOT_FOUND
> > *Feb 20 03:38:14.913: ////GK/gk_call_new:
> > checking for default (CLI) carrier for sep endpt 0x4ABCA7A8
> > *Feb 20 03:38:14.937: ////GK/gk_call_find_crv:
> > endptp=0x4A6F20B0, crv=32829: 
> > *Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
> > crv is DEP
> > R1#
> > *Feb 20 03:38:15.005: ////GK/gk_call_find_crv:
> > endptp=0x4A6F20B0, crv=32829: 
> > *Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
> > crv is DEP
> > R1#
> > *Feb 20 03:38:24.397: ////GK/gk_call_find_crv:
> > endptp=0x4ABCA7A8, crv=61: 
> > *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
> > crv is SEP
> > *Feb 20 03:38:24.397: ////GK/gk_call_clear_crv:
> > endptp=0x4A6F20B0, crv=32829: 
> > *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> > crv is DEP
> > *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> > c_callstate 0x801, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
> > 0x4A696CA4
> > *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> > callp=4A35B91C
> > R1#
> > *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> > c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
> > 0x4A696CA4
> > *Feb 20 03:38:24.425:
> > ////GK/gk_call_find_dstendpt: NOT_FOUND
> > *Feb 20 03:38:24.425: ////GK/gk_call_new:
> > src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0,
> > bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0,
> > ret_callpp=0x492D8D78
> > *Feb 20 03:38:24.425:
> > ////GK/gk_call_find_endpts: NOT_FOUND
> > *Feb 20 03:38:24.429: ////GK/gk_call_new:
> > checking for default (CLI) carrier for dep endpt 0x4ABFA848
> > R1#
> > *Feb 20 03:38:29.449: ////GK/gk_call_clear_crv:
> > endptp=0x4ABFA848, crv=32830: 
> > *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> > crv is DEP
> > *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> > c_callstate 0x800, c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4,
> > c_reszp2 0x4A696CA4
> > *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> > callp=4A57164C
> > *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> > c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2
> > 0x4A696CA4
> > *Feb 20 03:38:29.453: ////GK/gk_call_clear_crv:
> > endptp=0x4ABCA7A8, crv=61: 
> > R1#
> > *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> > crv is SEP
> > *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> > c_callst

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Hi Kavi,

 

Thanks.  I already had this configured.  I appreciate your help.  Now
the GK functionality is working like a charm.  I just forgot the Predot
DDI in the CUCM Translation Pattern.  

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kavi ten
Sent: Friday, February 19, 2010 8:12 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Gatekeeper Issue

 

Hi Jeff,

 

Please chk the Tech prefix on the CUCM , 

If its set to 1# , then u need to send 1 # along with the no from CME &
remove it at the GW with incoming allowed nos or use TP.

 

 

 

 

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

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Hi Mustafa,

The direction is from CME to CUCM.  Both are configured for G.729 and
I've confirmed a call goes through and uses G729 both ways. That's
definitely odd for the bandwidth to say that.  I don't really have an
answer, but the call is using G729 as configured.

Thanks again.

Jeff

-Original Message-
From: Mustafa [mailto:afat...@verizon.net] 
Sent: Friday, February 19, 2010 6:49 PM
To: Jeff Price (jeffpric)
Cc: CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue

whats the direction of call? Previous debug has bandwidth 160, which is 
16k for g729 and this one is showing bandwidth 1280 which is 128k for 
g711. So this call leg debug is showing is g711 is being requested
somwhere.

-- Mustafa


Jeff Price (jeffpric) wrote:
> Here is the output of command debug gatekeeper call 10:
>
> *Feb 20 03:38:14.913: ////GK/gk_call_new:
> src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0,
bw=160,
> crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78
> *Feb 20 03:38:14.913:
> ////GK/gk_call_find_endpts: NOT_FOUND
> *Feb 20 03:38:14.913: ////GK/gk_call_new:
> checking for default (CLI) carrier for sep endpt 0x4ABCA7A8
> *Feb 20 03:38:14.937: ////GK/gk_call_find_crv:
> endptp=0x4A6F20B0, crv=32829: 
> *Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
> crv is DEP
> R1#
> *Feb 20 03:38:15.005: ////GK/gk_call_find_crv:
> endptp=0x4A6F20B0, crv=32829: 
> *Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
> crv is DEP
> R1#
> *Feb 20 03:38:24.397: ////GK/gk_call_find_crv:
> endptp=0x4ABCA7A8, crv=61: 
> *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
> crv is SEP
> *Feb 20 03:38:24.397:
////GK/gk_call_clear_crv:
> endptp=0x4A6F20B0, crv=32829: 
> *Feb 20 03:38:24.397:
//620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> crv is DEP
> *Feb 20 03:38:24.397:
//620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> c_callstate 0x801,  c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4,
c_reszp2
> 0x4A696CA4
> *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> callp=4A35B91C
> R1#
> *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4,
c_reszp2
> 0x4A696CA4
> *Feb 20 03:38:24.425:
> ////GK/gk_call_find_dstendpt: NOT_FOUND
> *Feb 20 03:38:24.425: ////GK/gk_call_new:
> src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0,
dst_pxp=0x0,
> bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0,
> ret_callpp=0x492D8D78
> *Feb 20 03:38:24.425:
> ////GK/gk_call_find_endpts: NOT_FOUND
> *Feb 20 03:38:24.429: ////GK/gk_call_new:
> checking for default (CLI) carrier for dep endpt 0x4ABFA848
> R1#
> *Feb 20 03:38:29.449:
////GK/gk_call_clear_crv:
> endptp=0x4ABFA848, crv=32830: 
> *Feb 20 03:38:29.449:
//620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> crv is DEP
> *Feb 20 03:38:29.449:
//620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> c_callstate 0x800,  c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4,
> c_reszp2 0x4A696CA4
> *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> callp=4A57164C
> *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4,
c_reszp2
> 0x4A696CA4
> *Feb 20 03:38:29.453:
////GK/gk_call_clear_crv:
> endptp=0x4ABCA7A8, crv=61: 
> R1#
> *Feb 20 03:38:29.453:
//620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> crv is SEP
> *Feb 20 03:38:29.453:
//620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
> c_callstate 0x7C00, c_resbw1 0,  resbw2 0, c_reszp1 0x4A696CA4,
c_reszp2
> 0x4A696CA4
> *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> callp=4A35B91C
> *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
> c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4,
c_reszp2
> 0x4A696CA4
>
> The gk_call_find_endpts: NOT_FOUND line is leading me to believe
> something that I was noticed on the debug gatekeeper main 10 command:
>   
>> //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
>> Matched zone prefix 1 and remainder 7752011001
>> 
>
> The pattern I've configured on CUCM is expecting to receive in the
form
> 17752011001.  This output almost makes it seem like only 7752011001 is
> being sent, is this correct?  I'm going to try to add another pattern
> without

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Mustafa,

So I figured out why.  CUCM was unable to find the number and that's why
it couldn't route.  The reason why CUCM could find the number is...a
stupid mistake.  I simply forgot to set the DDI to Predot.  Now this is
working.  It always the little things...

Thanks Mustafa for your help.

Jeff

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Friday, February 19, 2010 6:38 PM
To: CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue

Here is the output of command debug gatekeeper call 10:

*Feb 20 03:38:14.913: ////GK/gk_call_new:
src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160,
crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78
*Feb 20 03:38:14.913:
////GK/gk_call_find_endpts: NOT_FOUND
*Feb 20 03:38:14.913: ////GK/gk_call_new:
checking for default (CLI) carrier for sep endpt 0x4ABCA7A8
*Feb 20 03:38:14.937: ////GK/gk_call_find_crv:
endptp=0x4A6F20B0, crv=32829: 
*Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
crv is DEP
R1#
*Feb 20 03:38:15.005: ////GK/gk_call_find_crv:
endptp=0x4A6F20B0, crv=32829: 
*Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
crv is DEP
R1#
*Feb 20 03:38:24.397: ////GK/gk_call_find_crv:
endptp=0x4ABCA7A8, crv=61: 
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
crv is SEP
*Feb 20 03:38:24.397: ////GK/gk_call_clear_crv:
endptp=0x4A6F20B0, crv=32829: 
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
crv is DEP
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
c_callstate 0x801,  c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
callp=4A35B91C
R1#
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4
*Feb 20 03:38:24.425:
////GK/gk_call_find_dstendpt: NOT_FOUND
*Feb 20 03:38:24.425: ////GK/gk_call_new:
src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0,
bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0,
ret_callpp=0x492D8D78
*Feb 20 03:38:24.425:
////GK/gk_call_find_endpts: NOT_FOUND
*Feb 20 03:38:24.429: ////GK/gk_call_new:
checking for default (CLI) carrier for dep endpt 0x4ABFA848
R1#
*Feb 20 03:38:29.449: ////GK/gk_call_clear_crv:
endptp=0x4ABFA848, crv=32830: 
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
crv is DEP
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
c_callstate 0x800,  c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4,
c_reszp2 0x4A696CA4
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
callp=4A57164C
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2
0x4A696CA4
*Feb 20 03:38:29.453: ////GK/gk_call_clear_crv:
endptp=0x4ABCA7A8, crv=61: 
R1#
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
crv is SEP
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
c_callstate 0x7C00, c_resbw1 0,  resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
callp=4A35B91C
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4

The gk_call_find_endpts: NOT_FOUND line is leading me to believe
something that I was noticed on the debug gatekeeper main 10 command:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
> Matched zone prefix 1 and remainder 7752011001

The pattern I've configured on CUCM is expecting to receive in the form
17752011001.  This output almost makes it seem like only 7752011001 is
being sent, is this correct?  I'm going to try to add another pattern
without the 1 so that I can test this.

Jeff

-Original Message-
From: Mustafa [mailto:afat...@verizon.net] 
Sent: Friday, February 19, 2010 6:28 PM
To: Jeff Price (jeffpric); CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue

GK seems to be routing the call. Do you get a busy tone when the call is

unsuccessful? Have you configured an xcoder anyway on CME?

-- Mustafa

Jeff Price (jeffpric) wrote:
> I'm looking through this as I'm sending.  I also wanted to note that I
> set the Service Parameter "BRQ Enable" to True, because I noticed that
> the DRQs were coming shortly 

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Here is the output of command debug gatekeeper call 10:

*Feb 20 03:38:14.913: ////GK/gk_call_new:
src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160,
crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78
*Feb 20 03:38:14.913:
////GK/gk_call_find_endpts: NOT_FOUND
*Feb 20 03:38:14.913: ////GK/gk_call_new:
checking for default (CLI) carrier for sep endpt 0x4ABCA7A8
*Feb 20 03:38:14.937: ////GK/gk_call_find_crv:
endptp=0x4A6F20B0, crv=32829: 
*Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
crv is DEP
R1#
*Feb 20 03:38:15.005: ////GK/gk_call_find_crv:
endptp=0x4A6F20B0, crv=32829: 
*Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
crv is DEP
R1#
*Feb 20 03:38:24.397: ////GK/gk_call_find_crv:
endptp=0x4ABCA7A8, crv=61: 
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv:
crv is SEP
*Feb 20 03:38:24.397: ////GK/gk_call_clear_crv:
endptp=0x4A6F20B0, crv=32829: 
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
crv is DEP
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
c_callstate 0x801,  c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
callp=4A35B91C
R1#
*Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4
*Feb 20 03:38:24.425:
////GK/gk_call_find_dstendpt: NOT_FOUND
*Feb 20 03:38:24.425: ////GK/gk_call_new:
src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0,
bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0,
ret_callpp=0x492D8D78
*Feb 20 03:38:24.425:
////GK/gk_call_find_endpts: NOT_FOUND
*Feb 20 03:38:24.429: ////GK/gk_call_new:
checking for default (CLI) carrier for dep endpt 0x4ABFA848
R1#
*Feb 20 03:38:29.449: ////GK/gk_call_clear_crv:
endptp=0x4ABFA848, crv=32830: 
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
crv is DEP
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
c_callstate 0x800,  c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4,
c_reszp2 0x4A696CA4
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
callp=4A57164C
*Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2
0x4A696CA4
*Feb 20 03:38:29.453: ////GK/gk_call_clear_crv:
endptp=0x4ABCA7A8, crv=61: 
R1#
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
crv is SEP
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv:
c_callstate 0x7C00, c_resbw1 0,  resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
callp=4A35B91C
*Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete:
c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2
0x4A696CA4

The gk_call_find_endpts: NOT_FOUND line is leading me to believe
something that I was noticed on the debug gatekeeper main 10 command:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
> Matched zone prefix 1 and remainder 7752011001

The pattern I've configured on CUCM is expecting to receive in the form
17752011001.  This output almost makes it seem like only 7752011001 is
being sent, is this correct?  I'm going to try to add another pattern
without the 1 so that I can test this.

Jeff

-Original Message-
From: Mustafa [mailto:afat...@verizon.net] 
Sent: Friday, February 19, 2010 6:28 PM
To: Jeff Price (jeffpric); CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue

GK seems to be routing the call. Do you get a busy tone when the call is

unsuccessful? Have you configured an xcoder anyway on CME?

-- Mustafa

Jeff Price (jeffpric) wrote:
> I'm looking through this as I'm sending.  I also wanted to note that I
> set the Service Parameter "BRQ Enable" to True, because I noticed that
> the DRQs were coming shortly after the BRQ were sent by the CME.  
>
> *Feb 20 03:18:10.457: ////GK/gk_process: got a
> TIMER event
>
> *Feb 20 03:18:10.457: ////GK/gk_handle_timers
>
> *Feb 20 03:18:10.457: ////GK/gk_handle_timers:
> managed timer expired 0x47620C08 
>
> *Feb 20 03:18:10.733: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> R1#
> *Feb 20 03:18:13.393: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:13.3

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

Mustafa,

Do I need a XCODER on CME?  Region is configured to use G729 and so is
the dial-peer on CME.  The call goes through successfully when calling
from CUCM to CME site, so that leads me to believe the XCODER is
necessary?  

The call never goes through.  No ringing, no busy, nothing.  As far as I
can tell, the call goes through the GK logic of trying both TRUNKS (SUB
then PUB) and then falls back to the PSTN dial-peer I've configured and
the call is completed successfully through the PSTN.

Jeff

-Original Message-
From: Mustafa [mailto:afat...@verizon.net] 
Sent: Friday, February 19, 2010 6:28 PM
To: Jeff Price (jeffpric); CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue

GK seems to be routing the call. Do you get a busy tone when the call is

unsuccessful? Have you configured an xcoder anyway on CME?

-- Mustafa

Jeff Price (jeffpric) wrote:
> I'm looking through this as I'm sending.  I also wanted to note that I
> set the Service Parameter "BRQ Enable" to True, because I noticed that
> the DRQs were coming shortly after the BRQ were sent by the CME.  
>
> *Feb 20 03:18:10.457: ////GK/gk_process: got a
> TIMER event
>
> *Feb 20 03:18:10.457: ////GK/gk_handle_timers
>
> *Feb 20 03:18:10.457: ////GK/gk_handle_timers:
> managed timer expired 0x47620C08 
>
> *Feb 20 03:18:10.733: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> R1#
> *Feb 20 03:18:13.393: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:13.393: ////GK/gk_rassrv_arq:
> arqp=0x4A5F7A4C,crv=0x37, answerCall=0
> *Feb 20 03:18:13.393:
////GK/gk_rassrv_sep_arq:
> ARQ Didn't use GK_AAA_PROC
> *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/gk_dns_query: No
> Name servers
> *Feb 20 03:18:13.393:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
> Matched tech-prefix 1#
> *Feb 20 03:18:13.393:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
> Matched zone prefix 1 and remainder 7752011001
> *Feb 20 03:18:13.393:
> ////GK/gk_rassrv_get_ingress_network: ARQ
> non-std ingress network = 1
> *Feb 20 03:18:13.393:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to
check
> the source side, src_zonep=0x4A696CA4
> *Feb 20 03:18:13.393:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone
> is ZONE_01, an
> R1#d z_invianamelen=0
> *Feb 20 03:18:13.393:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to
check
> the destination side, dst_zonep=0x4A696CA4
> *Feb 20 03:18:13.393:
> //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone
> is ZONE_01, and z_outvianamelen=0
> *Feb 20 03:18:13.397:
> ////GK/gk_rassrv_get_ingress_network: ARQ
> non-std ingress network = 1
> *Feb 20 03:18:13.421: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:13.421: ////GK/gk_rassrv_arq:
> arqp=0x4AD8C8B8,crv=0x8037, answerCall=1
> *Feb 20 03:18:13.421:
//95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq:
> ARQ Didn't use GK_AAA_PROC
> *Feb 20 03:18:13.465: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:13.469: ////GK/gk_rassrv_brq:
> state = 0xF
> *Feb 20 03:18:13.469: ////GK/gk_rassrv_brq:
> brqp=0x4A6F94A8, crv=0x8037, bandWidth=160
> R1#
> R1#
> *Feb 20 03:18:22.781: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:22.781: ////GK/gk_rassrv_brq:
> state = 0xF
> *Feb 20 03:18:22.781: ////GK/gk_rassrv_brq:
> brqp=0x4A6F94A8, crv=0x37, bandWidth=0
> *Feb 20 03:18:22.785: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:22.805: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:22.805: ////GK/gk_rassrv_arq:
> arqp=0x4AB97164,crv=0x8038, answerCall=1
> *Feb 20 03:18:22.805:
//95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq:
> ARQ Didn't use GK_AAA_PROC
> R1#
> *Feb 20 03:18:25.457: ////GK/gk_process: got a
> TIMER event
>
> *Feb 20 03:18:25.457: ////GK/gk_handle_timers
>
> *Feb 20 03:18:25.457: ////GK/gk_handle_timers:
> managed timer expired 0x47620C08 
>
> R1#
> *Feb 20 03:18:27.825: ////GK/gk_process:
> QUEUE_EVENT (minor 0) wakeup
> *Feb 20 03:18:27.82

Re: [OSL | CCIE_Voice] Gatekeeper Issue

[Jeff Price (jeffpric)]

I'm looking through this as I'm sending.  I also wanted to note that I
set the Service Parameter "BRQ Enable" to True, because I noticed that
the DRQs were coming shortly after the BRQ were sent by the CME.  

*Feb 20 03:18:10.457: ////GK/gk_process: got a
TIMER event

*Feb 20 03:18:10.457: ////GK/gk_handle_timers

*Feb 20 03:18:10.457: ////GK/gk_handle_timers:
managed timer expired 0x47620C08 

*Feb 20 03:18:10.733: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
R1#
*Feb 20 03:18:13.393: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:13.393: ////GK/gk_rassrv_arq:
arqp=0x4A5F7A4C,crv=0x37, answerCall=0
*Feb 20 03:18:13.393: ////GK/gk_rassrv_sep_arq:
ARQ Didn't use GK_AAA_PROC
*Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/gk_dns_query: No
Name servers
*Feb 20 03:18:13.393:
//95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
Matched tech-prefix 1#
*Feb 20 03:18:13.393:
//95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001)
Matched zone prefix 1 and remainder 7752011001
*Feb 20 03:18:13.393:
////GK/gk_rassrv_get_ingress_network: ARQ
non-std ingress network = 1
*Feb 20 03:18:13.393:
//95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to check
the source side, src_zonep=0x4A696CA4
*Feb 20 03:18:13.393:
//95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone
is ZONE_01, an
R1#d z_invianamelen=0
*Feb 20 03:18:13.393:
//95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to check
the destination side, dst_zonep=0x4A696CA4
*Feb 20 03:18:13.393:
//95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone
is ZONE_01, and z_outvianamelen=0
*Feb 20 03:18:13.397:
////GK/gk_rassrv_get_ingress_network: ARQ
non-std ingress network = 1
*Feb 20 03:18:13.421: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:13.421: ////GK/gk_rassrv_arq:
arqp=0x4AD8C8B8,crv=0x8037, answerCall=1
*Feb 20 03:18:13.421: //95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq:
ARQ Didn't use GK_AAA_PROC
*Feb 20 03:18:13.465: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:13.469: ////GK/gk_rassrv_brq:
state = 0xF
*Feb 20 03:18:13.469: ////GK/gk_rassrv_brq:
brqp=0x4A6F94A8, crv=0x8037, bandWidth=160
R1#
R1#
*Feb 20 03:18:22.781: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:22.781: ////GK/gk_rassrv_brq:
state = 0xF
*Feb 20 03:18:22.781: ////GK/gk_rassrv_brq:
brqp=0x4A6F94A8, crv=0x37, bandWidth=0
*Feb 20 03:18:22.785: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:22.805: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:22.805: ////GK/gk_rassrv_arq:
arqp=0x4AB97164,crv=0x8038, answerCall=1
*Feb 20 03:18:22.805: //95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq:
ARQ Didn't use GK_AAA_PROC
R1#
*Feb 20 03:18:25.457: ////GK/gk_process: got a
TIMER event

*Feb 20 03:18:25.457: ////GK/gk_handle_timers

*Feb 20 03:18:25.457: ////GK/gk_handle_timers:
managed timer expired 0x47620C08 

R1#
*Feb 20 03:18:27.825: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:27.829: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
R1#
*Feb 20 03:18:33.965: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 20 03:18:34.381: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
R1#
*Feb 20 03:18:40.457: ////GK/gk_process: got a
TIMER event

*Feb 20 03:18:40.457: ////GK/gk_handle_timers

*Feb 20 03:18:40.457: ////GK/gk_handle_timers:
managed timer expired 0x47620C08

Thanks,

Jeff


-Original Message-
From: Mustafa [mailto:afat...@verizon.net] 
Sent: Friday, February 19, 2010 6:09 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue

Jeff,

Do a "debug gatekeeper main 10" on the gatekeeper and check/send the 
output when you make a call. It will tell you whats happening and if 
there is a routing failure at the gatekeeper.

Also, take a look at this documentation 
http://www.cisco.com/en/US/docs/ios/voice/cubegk/configuration/guide/ve_
book/ve_book.html 
, it details how to configure gatekeeper with CUBE and its a good read.

-- Mustafa

Jeff Price (jeffpric) wrote:
>
> Hi everyone,
>
> I'm having an issue with my gatekeeper. When calling from CUCM to GK 
> to CME, th

[OSL | CCIE_Voice] PSTN/ISDN issue

[Jeff Price (jeffpric)]

Hi everyone,

 

I am having an issue with my PSTN router.  The PSTN router keeps sending
a Restart message on channel 1 every 30 seconds or so to my HQ's router
(R1).  Does anyone know what could cause this behavior?  I have tried
reconfiguring my isdn on both sides, but I am still seeing the same
issue even after reconfiguring this.  

 

Here is the error as seen from R1:

 

Feb 13 06:29:00.402: ISDN Se0/0/0:23 Q931: RX <- RESTART pd = 8  callref
= 0x 

Channel ID i = 0xA98381 

Exclusive, Channel 1 

Restart Indicator i = 0x80

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 

 

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

Re: [OSL | CCIE_Voice] Dialing problem

[Jeff Price (jeffpric)]

Hey Bas,

 

I got everything working for the most part.  It seems like some things
just take a while to start working for this lab.  I did create some SIP
dial rules, because I know its best practice to offload that processing.
I have since restored all of my lab back to a clean config to run
through the lab from the beginning, so if I run into any problems I will
let you all know.  Thanks for getting back to me.

 

Jeff

 

From: Bas Janssen [mailto:basmj...@msn.com] 
Sent: Friday, February 12, 2010 4:15 AM
To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Dialing problem

 

Hi Jeff,

Any luck so far? 

One last remark, if you use SIP phones, don't forget to make dial rules,
even if your SIP phones supports KPML Otherwise calling back from the
missed call list might not work because SIP phones use overlap sending
from the missed call list without SIP dial rules.

regards,

Bas








Subject: RE: [OSL | CCIE_Voice] Dialing problem
Date: Tue, 9 Feb 2010 12:32:46 -0600
From: jeffp...@cisco.com
To: basmj...@msn.com; ccie_voice@onlinestudylist.com

Hey Bas,

 

Thanks for the input.  I will look into all of this later today.  

 

Jeff

 

From: Bas Janssen [mailto:basmj...@msn.com] 
Sent: Tuesday, February 09, 2010 6:03 AM
To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Dialing problem

 

Hi Jeff,

I run into the same issue today, it was introduced on configuring +
dialing. So, call flow is for example 93942123--->trans. pattern--->
+12123942123>route pattern \+!->Route List --->GW--->called
party transform->3942123 type subscriber
Objective is to use only one route pattern to PSTN.
Had "play second dial tone" ON on TP and RP.  You have to check box
urgent priority on the \+! dial pattern, otherwise you have the second
dial tone while waiting for inter digit timeout.  You can remove the
check for playing second dial tone, but than you stiil have an inter
digit timeout. (without urgent priority) 
Watch out for international numbers, these will fail with urgent
priority on the \+! 
so if you have a trans pattern with \+!, remove the urgent priority here
to get international calls to work.


regards,

Bas







Date: Mon, 8 Feb 2010 20:10:08 -0600
From: jeffp...@cisco.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Dialing problem

Jonathan,

 

CUCM and SCCP.

 

Jeff

 

From: Jonathan Charles [mailto:jonv...@gmail.com] 
Sent: Monday, February 08, 2010 4:32 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Dialing problem

 

CME or CUCM? SIP or SCCP? 

If SIP, did you create a SIP Dial Plan?



Jonathan

On Mon, Feb 8, 2010 at 5:28 PM, Jeff Price (jeffpric)
 wrote:

Hi everyone,

 

I am having an issue with the dialing for my phones.  All of my phones
are able to call across the PSTN, however it only works when I dial the
digits and then press the "Dial" softkey.  If I pick up the headset and
dial that way I get a steady tone.  Anyone familiar with this issue?
Thanks.

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 


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

 

 



Hotmail: Free, trusted and rich email service. Get it now.
<https://signup.live.com/signup.aspx?id=60969> 

 



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<><><>___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Dialing problem

[Jeff Price (jeffpric)]

Hey Bas,

 

Thanks for the input.  I will look into all of this later today.  

 

Jeff

 

From: Bas Janssen [mailto:basmj...@msn.com] 
Sent: Tuesday, February 09, 2010 6:03 AM
To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Dialing problem

 

Hi Jeff,

I run into the same issue today, it was introduced on configuring +
dialing. So, call flow is for example 93942123--->trans. pattern--->
+12123942123>route pattern \+!->Route List --->GW--->called
party transform->3942123 type subscriber
Objective is to use only one route pattern to PSTN.
Had "play second dial tone" ON on TP and RP.  You have to check box
urgent priority on the \+! dial pattern, otherwise you have the second
dial tone while waiting for inter digit timeout.  You can remove the
check for playing second dial tone, but than you stiil have an inter
digit timeout. (without urgent priority) 
Watch out for international numbers, these will fail with urgent
priority on the \+! 
so if you have a trans pattern with \+!, remove the urgent priority here
to get international calls to work.


regards,

Bas








Date: Mon, 8 Feb 2010 20:10:08 -0600
From: jeffp...@cisco.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Dialing problem

Jonathan,

 

CUCM and SCCP.

 

Jeff

 

From: Jonathan Charles [mailto:jonv...@gmail.com] 
Sent: Monday, February 08, 2010 4:32 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Dialing problem

 

CME or CUCM? SIP or SCCP? 

If SIP, did you create a SIP Dial Plan?



Jonathan

On Mon, Feb 8, 2010 at 5:28 PM, Jeff Price (jeffpric)
 wrote:

Hi everyone,

 

I am having an issue with the dialing for my phones.  All of my phones
are able to call across the PSTN, however it only works when I dial the
digits and then press the "Dial" softkey.  If I pick up the headset and
dial that way I get a steady tone.  Anyone familiar with this issue?
Thanks.

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 


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

 

 



Hotmail: Free, trusted and rich email service. Get it now.
<https://signup.live.com/signup.aspx?id=60969> 

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

Re: [OSL | CCIE_Voice] Dialing problem

[Jeff Price (jeffpric)]

Jonathan,

 

CUCM and SCCP.

 

Jeff

 

From: Jonathan Charles [mailto:jonv...@gmail.com] 
Sent: Monday, February 08, 2010 4:32 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Dialing problem

 

CME or CUCM? SIP or SCCP? 

If SIP, did you create a SIP Dial Plan?



Jonathan

On Mon, Feb 8, 2010 at 5:28 PM, Jeff Price (jeffpric) <
jeffp...@cisco.com> wrote:

Hi everyone,

 

I am having an issue with the dialing for my phones.  All of my phones
are able to call across the PSTN, however it only works when I dial the
digits and then press the "Dial" softkey.  If I pick up the headset and
dial that way I get a steady tone.  Anyone familiar with this issue?
Thanks.

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page <http://www.cisco.com/> 

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 


___
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] Dialing problem

[Jeff Price (jeffpric)]

Hi everyone,

 

I am having an issue with the dialing for my phones.  All of my phones
are able to call across the PSTN, however it only works when I dial the
digits and then press the "Dial" softkey.  If I pick up the headset and
dial that way I get a steady tone.  Anyone familiar with this issue?
Thanks.

 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 

 

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

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

Now I am able to.  I ran the utils dbreplication status command as well, which 
didn't report any issues.  However, maybe this command had an issue as well.  
I'm just glad it's working now.

 

From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas
Sent: Monday, February 08, 2010 3:11 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

 

Jeff - you ARE able to make the calls?

Sounds like you may have had some kind of replication issue going on between 
the pub/sub.

-jeff

On Mon, Feb 8, 2010 at 5:47 PM, Jeff Price (jeffpric)  
wrote:

Hi everyone,

 

So after trying all of your suggestions, I guess the old Windows philosophy 
worked.  After I restarted the PUB/SUB I am not able to make the calls.  Thanks 
for all of your help.

 

Jeff

 

From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas
Sent: Monday, February 08, 2010 2:09 PM


To: Jeff Price (jeffpric)
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

 

I'm sorry Jeff.  I -always- do that but if I go to the CUCM interface I do it 
right.

The Route List can be reset -- try that and see if you can suddenly start 
dialing.   If not, try resetting the gateway too.We've had cases in our 
production environment where a location can't dial anywhere, but if you reset 
the route list its fixes it.

-jeff

On Mon, Feb 8, 2010 at 4:58 PM, Jeff Price (jeffpric)  
wrote:

Jeff,

 

I am not sure what you mean by reset the route group.  I don't see a reset 
option under Route Group.  However, I have reset the GW, and Route List 
numerous times.  Is this what you mean?  If not, could you provide me some more 
information?  Thanks for your help.

 

Jeff 

 

From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas
Sent: Monday, February 08, 2010 12:46 PM
To: Jeff Price (jeffpric)


Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

 

Jeff - did you try to reset the route group?  

If not, navigate to your route group, reset it and see if your call attempt 
shows up in debug isdn q931.   If you're not using the console don't forget to 
turn on console logging.  About 99% of the time in our production environment 
when I get a fast busy signal immediately after dialing (and no presence of the 
call attempting to exit the gateway)  its because the route group needs to be 
reset. 

-jeff

On Mon, Feb 8, 2010 at 2:50 PM, Jeff Price (jeffpric)  
wrote:

Hi all,

I'm fairly new to the RTMT tool.  I'm looking around trying to find the correct 
area to be and what to select to monitor.  Can someone point me in the right 
direction for this problem?  Thanks.

Jeff


-Original Message-
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Monday, February 08, 2010 4:08 AM
To: Scott Totaro (stotaro); Jeff Price (jeffpric); afatsum
Cc: ccie_voice@onlinestudylist.com

Subject: SV: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Scott,
Even if you use MGCP, ie backhauled D-channel, you can see the output of "deb 
isdn q931" on the gateway. It will be locally echoed.

Brgds,
Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: Scott Totaro (stotaro) [stot...@cisco.com]
Skickat: den 8 februari 2010 01:15
Till: Jeff Price (jeffpric); afatsum
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Because the D-channel is backhauled to CUCM for MGCP gateways, you will
not see Q931 output on the router. Instead, you'll need to look at the
detailed trace files on the subscriber that the MGCP gateway is
registered too. In my experience, you won't be able to use any IOS
commands that depend on the router terminating the D-channel (e.g. show
isdn history.)

Hope this helps,
Scott

-Original Message-
From: Jeff Price (jeffpric)
Sent: Sunday, February 07, 2010 3:41 PM
To: afatsum
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

It appears that my transformation is working.  When dialing
91408425, the display on the phone says "To 408425".  And the
DNA analysis shows what the CUCM is going through process-wise.  Yet I
am still not receiving any ISDN Q931 debug output on R1 and the phones
still receive a fast busy.  As I had said in a previous email, the PSTN
router that the phones are calling to is already pre-configured and I
don't have access to it. Even if it was the PSTN router causing the
problem, wouldn't I still see the Q931 output on R1?

Thanks for the help.

Jeff

-Original Message-
From: Jeff Price (jeffpric)
Sent: Sunday, February 07, 2010 12:35 PM
To: 'afatsum'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem

I have deactivated all of the serv

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

Hi everyone,

 

So after trying all of your suggestions, I guess the old Windows philosophy 
worked.  After I restarted the PUB/SUB I am not able to make the calls.  Thanks 
for all of your help.

 

Jeff

 

From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas
Sent: Monday, February 08, 2010 2:09 PM
To: Jeff Price (jeffpric)
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

 

I'm sorry Jeff.  I -always- do that but if I go to the CUCM interface I do it 
right.

The Route List can be reset -- try that and see if you can suddenly start 
dialing.   If not, try resetting the gateway too.We've had cases in our 
production environment where a location can't dial anywhere, but if you reset 
the route list its fixes it.

-jeff



On Mon, Feb 8, 2010 at 4:58 PM, Jeff Price (jeffpric)  
wrote:

Jeff,

 

I am not sure what you mean by reset the route group.  I don't see a reset 
option under Route Group.  However, I have reset the GW, and Route List 
numerous times.  Is this what you mean?  If not, could you provide me some more 
information?  Thanks for your help.

 

Jeff 

 

From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas
Sent: Monday, February 08, 2010 12:46 PM
To: Jeff Price (jeffpric)


Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

 

Jeff - did you try to reset the route group?  

If not, navigate to your route group, reset it and see if your call attempt 
shows up in debug isdn q931.   If you're not using the console don't forget to 
turn on console logging.  About 99% of the time in our production environment 
when I get a fast busy signal immediately after dialing (and no presence of the 
call attempting to exit the gateway)  its because the route group needs to be 
reset. 

-jeff

On Mon, Feb 8, 2010 at 2:50 PM, Jeff Price (jeffpric)  
wrote:

Hi all,

I'm fairly new to the RTMT tool.  I'm looking around trying to find the correct 
area to be and what to select to monitor.  Can someone point me in the right 
direction for this problem?  Thanks.

Jeff


-Original Message-
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Monday, February 08, 2010 4:08 AM
To: Scott Totaro (stotaro); Jeff Price (jeffpric); afatsum
Cc: ccie_voice@onlinestudylist.com

Subject: SV: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Scott,
Even if you use MGCP, ie backhauled D-channel, you can see the output of "deb 
isdn q931" on the gateway. It will be locally echoed.

Brgds,
Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: Scott Totaro (stotaro) [stot...@cisco.com]
Skickat: den 8 februari 2010 01:15
Till: Jeff Price (jeffpric); afatsum
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Because the D-channel is backhauled to CUCM for MGCP gateways, you will
not see Q931 output on the router. Instead, you'll need to look at the
detailed trace files on the subscriber that the MGCP gateway is
registered too. In my experience, you won't be able to use any IOS
commands that depend on the router terminating the D-channel (e.g. show
isdn history.)

Hope this helps,
Scott

-Original Message-
From: Jeff Price (jeffpric)
Sent: Sunday, February 07, 2010 3:41 PM
To: afatsum
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

It appears that my transformation is working.  When dialing
91408425, the display on the phone says "To 408425".  And the
DNA analysis shows what the CUCM is going through process-wise.  Yet I
am still not receiving any ISDN Q931 debug output on R1 and the phones
still receive a fast busy.  As I had said in a previous email, the PSTN
router that the phones are calling to is already pre-configured and I
don't have access to it. Even if it was the PSTN router causing the
problem, wouldn't I still see the Q931 output on R1?

Thanks for the help.

Jeff

-Original Message-
From: Jeff Price (jeffpric)
Sent: Sunday, February 07, 2010 12:35 PM
To: 'afatsum'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem

I have deactivated all of the services on the SUB and let everything
register with the PUB.

Jeff

-Original Message-
From: afatsum [mailto:afat...@verizon.net]
Sent: Sunday, February 07, 2010 12:05 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Jeff,

Can you shutdown the sub and let everything register to pub and then do
the testing? This way atleast we can eliminate the sub for
troubleshooting purposes.

-- Mustafa

Jeff Price (jeffpric) wrote:
>
> Hi again,
>
> I was able to access the DNA on SUB, but not the PUB even though both
> servers are running the service.
>
> 

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

Hi all,

I'm fairly new to the RTMT tool.  I'm looking around trying to find the correct 
area to be and what to select to monitor.  Can someone point me in the right 
direction for this problem?  Thanks.

Jeff

-Original Message-
From: Roger Källberg [mailto:roger.kallb...@cygate.se] 
Sent: Monday, February 08, 2010 4:08 AM
To: Scott Totaro (stotaro); Jeff Price (jeffpric); afatsum
Cc: ccie_voice@onlinestudylist.com
Subject: SV: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Scott,
Even if you use MGCP, ie backhauled D-channel, you can see the output of "deb 
isdn q931" on the gateway. It will be locally echoed.

Brgds,
Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: Scott Totaro (stotaro) [stot...@cisco.com]
Skickat: den 8 februari 2010 01:15
Till: Jeff Price (jeffpric); afatsum
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Because the D-channel is backhauled to CUCM for MGCP gateways, you will
not see Q931 output on the router. Instead, you'll need to look at the
detailed trace files on the subscriber that the MGCP gateway is
registered too. In my experience, you won't be able to use any IOS
commands that depend on the router terminating the D-channel (e.g. show
isdn history.)

Hope this helps,
Scott

-----Original Message-----
From: Jeff Price (jeffpric)
Sent: Sunday, February 07, 2010 3:41 PM
To: afatsum
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

It appears that my transformation is working.  When dialing
91408425, the display on the phone says "To 408425".  And the
DNA analysis shows what the CUCM is going through process-wise.  Yet I
am still not receiving any ISDN Q931 debug output on R1 and the phones
still receive a fast busy.  As I had said in a previous email, the PSTN
router that the phones are calling to is already pre-configured and I
don't have access to it. Even if it was the PSTN router causing the
problem, wouldn't I still see the Q931 output on R1?

Thanks for the help.

Jeff

-----Original Message-
From: Jeff Price (jeffpric)
Sent: Sunday, February 07, 2010 12:35 PM
To: 'afatsum'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem

I have deactivated all of the services on the SUB and let everything
register with the PUB.

Jeff

-Original Message-
From: afatsum [mailto:afat...@verizon.net]
Sent: Sunday, February 07, 2010 12:05 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Jeff,

Can you shutdown the sub and let everything register to pub and then do
the testing? This way atleast we can eliminate the sub for
troubleshooting purposes.

-- Mustafa

Jeff Price (jeffpric) wrote:
>
> Hi again,
>
> I was able to access the DNA on SUB, but not the PUB even though both
> servers are running the service.
>
> Here is the output of DNA. Everything seems to be okay.
>
> *Cisco Unified Communications Manager Dialed Number Analyzer Results *
>
>
>
>
> * *Results Summary*
>   o *Calling Party Information*
> + *Calling Party* = +14085252001
> + *Partition* = PT_HQ_DEVICES
> + *Device CSS* = CSS_HQ_DEVICES
> + *Line CSS* =
> + *AAR Group Name* =
> + *AAR CSS* =
>   o *Dialed Digits* = 91408425
>   o *Match Result* = RouteThisPattern
>   o *Matched Pattern Information*
> + *Pattern* = \+!
> + *Partition* = PT_GLOBAL
> + *Time Schedule* =
>   o *Called Party Number* = +1408425
>   o *Time Zone* = Pacific Standard/Daylight Time
>   o *End Device* = RL_LOCAL
>   o *Call Classification* = OffNet
>   o *InterDigit Timeout* = NO
>   o *Device Override* = Disabled
>   o *Outside Dial Tone* = NO
> * *Call Flow*
>   o *TranslationPattern* :*Pattern*= 9.1[2-9]XX[2-9]XX
> + *Positional Match List* = +1408425
> + *Calling Party Number* = +14085252001
> + *PreTransform Calling Party Number* = 2001
> + *PreTransform Called Party Number* = 91408425
> + *Calling Party Transformations*
>   # *External Phone Number Mask* = YES
>   # *Calling Party Mask* =
>   # *Prefix* =
>   # *CallingLineId Presentation* = Default
>   # *CallingName Presentation* = Default
>   # *Calling Party Number

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

Mustafa - 

I have configured Locations, however all of the HQ devices (GW and
phones) should all be in the HQ_LOC location.  However, I will verify
this tomorrow when I am back in the lab.  

Scott -

Thanks for the advice.  I plan on trying the RTMT tomorrow while back in
the lab.  Hopefully, this will help reveal the problem.

Thanks everyone for the help and I will provide more information
tomorrow as to my findings.

Jeff

-Original Message-
From: Scott Totaro (stotaro) 
Sent: Sunday, February 07, 2010 8:41 PM
To: afatsum
Cc: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem

I stand corrected...sorry for the misdirection. I guess it's the show
commands that haven't worked whereas the debugs are still useful. I'd
still suggest investigating the detailed trace files for the source of
the failure as Mustafa points out that the lack of debug likely
indicates that the call is never making it to the gateway.

Scott

-Original Message-
From: afatsum [mailto:afat...@verizon.net] 
Sent: Sunday, February 07, 2010 7:46 PM
To: Scott Totaro (stotaro)
Cc: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

If you do a "Debug isdn q931", you will see the layer 3 activity debug 
on the gateway. The layer 3 backhaul makes all the processing happen on 
the CUCM, thats why no cli commands, other than mgcp and ccm-manager are

useful on the gateway.

The debug isdn q931 output should show if the call is ever sent over the

isdn interface. If you don't see the output, then its never making it to

the gateway.

Another issue that can cause fast busy is if the gateway and phones are 
in two different locations and CAC has been implemented. Do you have 
configured CAC? Hub_none is still a different location.

-- Mustafa


Scott Totaro (stotaro) wrote:
> Because the D-channel is backhauled to CUCM for MGCP gateways, you
will
> not see Q931 output on the router. Instead, you'll need to look at the
> detailed trace files on the subscriber that the MGCP gateway is
> registered too. In my experience, you won't be able to use any IOS
> commands that depend on the router terminating the D-channel (e.g.
show
> isdn history.)
>
> Hope this helps,
> Scott
>
> -Original Message-
> From: Jeff Price (jeffpric) 
> Sent: Sunday, February 07, 2010 3:41 PM
> To: afatsum
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem
>
> It appears that my transformation is working.  When dialing
> 91408425, the display on the phone says "To 408425".  And the
> DNA analysis shows what the CUCM is going through process-wise.  Yet I
> am still not receiving any ISDN Q931 debug output on R1 and the phones
> still receive a fast busy.  As I had said in a previous email, the
PSTN
> router that the phones are calling to is already pre-configured and I
> don't have access to it. Even if it was the PSTN router causing the
> problem, wouldn't I still see the Q931 output on R1?
>
> Thanks for the help.
>
> Jeff
>
> -Original Message-
> From: Jeff Price (jeffpric) 
> Sent: Sunday, February 07, 2010 12:35 PM
> To: 'afatsum'
> Cc: ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem
>
> I have deactivated all of the services on the SUB and let everything
> register with the PUB.  
>
> Jeff
>
> -Original Message-
> From: afatsum [mailto:afat...@verizon.net] 
> Sent: Sunday, February 07, 2010 12:05 PM
> To: Jeff Price (jeffpric)
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem
>
> Hi Jeff,
>
> Can you shutdown the sub and let everything register to pub and then
do 
> the testing? This way atleast we can eliminate the sub for 
> troubleshooting purposes.
>
> -- Mustafa
>
> Jeff Price (jeffpric) wrote:
>   
>> Hi again,
>>
>> I was able to access the DNA on SUB, but not the PUB even though both

>> servers are running the service.
>>
>> Here is the output of DNA. Everything seems to be okay.
>>
>> *Cisco Unified Communications Manager Dialed Number Analyzer Results
*
>>
>>
>>
>>
>> * *Results Summary*
>>   o *Calling Party Information*
>> + *Calling Party* = +14085252001
>> + *Partition* = PT_HQ_DEVICES
>> + *Device CSS* = CSS_HQ_DEVICES
>> + *Line CSS* =
>> + *AAR Group Name* =
>> + *AAR CSS* =
>>   o *Dialed Digits* = 91408425
>>   o *Match Result* = RouteThisPattern
>>   o *Ma

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

It appears that my transformation is working.  When dialing
91408425, the display on the phone says "To 408425".  And the
DNA analysis shows what the CUCM is going through process-wise.  Yet I
am still not receiving any ISDN Q931 debug output on R1 and the phones
still receive a fast busy.  As I had said in a previous email, the PSTN
router that the phones are calling to is already pre-configured and I
don't have access to it. Even if it was the PSTN router causing the
problem, wouldn't I still see the Q931 output on R1?

Thanks for the help.

Jeff

-Original Message-----
From: Jeff Price (jeffpric) 
Sent: Sunday, February 07, 2010 12:35 PM
To: 'afatsum'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem

I have deactivated all of the services on the SUB and let everything
register with the PUB.  

Jeff

-Original Message-
From: afatsum [mailto:afat...@verizon.net] 
Sent: Sunday, February 07, 2010 12:05 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Jeff,

Can you shutdown the sub and let everything register to pub and then do 
the testing? This way atleast we can eliminate the sub for 
troubleshooting purposes.

-- Mustafa

Jeff Price (jeffpric) wrote:
>
> Hi again,
>
> I was able to access the DNA on SUB, but not the PUB even though both 
> servers are running the service.
>
> Here is the output of DNA. Everything seems to be okay.
>
> *Cisco Unified Communications Manager Dialed Number Analyzer Results *
>
>
>
>
> * *Results Summary*
>   o *Calling Party Information*
> + *Calling Party* = +14085252001
> + *Partition* = PT_HQ_DEVICES
> + *Device CSS* = CSS_HQ_DEVICES
> + *Line CSS* =
> + *AAR Group Name* =
> + *AAR CSS* =
>   o *Dialed Digits* = 91408425
>   o *Match Result* = RouteThisPattern
>   o *Matched Pattern Information*
> + *Pattern* = \+!
> + *Partition* = PT_GLOBAL
> + *Time Schedule* =
>   o *Called Party Number* = +1408425
>   o *Time Zone* = Pacific Standard/Daylight Time
>   o *End Device* = RL_LOCAL
>   o *Call Classification* = OffNet
>   o *InterDigit Timeout* = NO
>   o *Device Override* = Disabled
>   o *Outside Dial Tone* = NO
> * *Call Flow*
>   o *TranslationPattern* :*Pattern*= 9.1[2-9]XX[2-9]XX
> + *Positional Match List* = +1408425
> + *Calling Party Number* = +14085252001
> + *PreTransform Calling Party Number* = 2001
> + *PreTransform Called Party Number* = 91408425
> + *Calling Party Transformations*
>   # *External Phone Number Mask* = YES
>   # *Calling Party Mask* =
>   # *Prefix* =
>   # *CallingLineId Presentation* = Default
>   # *CallingName Presentation* = Default
>   # *Calling Party Number* = +14085252001
> + *ConnectedParty Transformations*
>   # *ConnectedLineId Presentation* = Default
>   # *ConnectedName Presentation* = Default
> + *Called Party Transformations*
>   # *Called Party Mask* =
>   # *Discard Digits Instruction* = PreDot
>   # *Prefix* = +
>   # *Called Number* = +1408425
>   o *Route Pattern* :*Pattern*= \+!
> + *Positional Match List* = +1408425
> + *DialPlan* =
> + *Route Filter*
>   # *Filter Name* =
>   # *Filter Clause* =
> + *Require Forced Authorization Code* = No
> + *Authorization Level* = 0
> + *Require Client Matter Code* = No
> + *Call Classification* =
> + *PreTransform Calling Party Number* = +14085252001
> + *PreTransform Called Party Number* = +1408425
> + *Calling Party Transformations*
>   # *External Phone Number Mask* = YES
>   # *Calling Party Mask* =
>   # *Prefix* =
>   # *CallingLineId Presentation* = Default
>   # *CallingName Presentation* = Default
>   # *Calling Party Number* = +14085252001
> + *ConnectedParty Transformations*
>   # *Connect

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

I have deactivated all of the services on the SUB and let everything
register with the PUB.  

Jeff

-Original Message-
From: afatsum [mailto:afat...@verizon.net] 
Sent: Sunday, February 07, 2010 12:05 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

Hi Jeff,

Can you shutdown the sub and let everything register to pub and then do 
the testing? This way atleast we can eliminate the sub for 
troubleshooting purposes.

-- Mustafa

Jeff Price (jeffpric) wrote:
>
> Hi again,
>
> I was able to access the DNA on SUB, but not the PUB even though both 
> servers are running the service.
>
> Here is the output of DNA. Everything seems to be okay.
>
> *Cisco Unified Communications Manager Dialed Number Analyzer Results *
>
>
>
>
> * *Results Summary*
>   o *Calling Party Information*
> + *Calling Party* = +14085252001
> + *Partition* = PT_HQ_DEVICES
> + *Device CSS* = CSS_HQ_DEVICES
> + *Line CSS* =
> + *AAR Group Name* =
> + *AAR CSS* =
>   o *Dialed Digits* = 91408425
>   o *Match Result* = RouteThisPattern
>   o *Matched Pattern Information*
> + *Pattern* = \+!
> + *Partition* = PT_GLOBAL
> + *Time Schedule* =
>   o *Called Party Number* = +1408425
>   o *Time Zone* = Pacific Standard/Daylight Time
>   o *End Device* = RL_LOCAL
>   o *Call Classification* = OffNet
>   o *InterDigit Timeout* = NO
>   o *Device Override* = Disabled
>   o *Outside Dial Tone* = NO
> * *Call Flow*
>   o *TranslationPattern* :*Pattern*= 9.1[2-9]XX[2-9]XX
> + *Positional Match List* = +1408425
> + *Calling Party Number* = +14085252001
> + *PreTransform Calling Party Number* = 2001
> + *PreTransform Called Party Number* = 91408425
> + *Calling Party Transformations*
>   # *External Phone Number Mask* = YES
>   # *Calling Party Mask* =
>   # *Prefix* =
>   # *CallingLineId Presentation* = Default
>   # *CallingName Presentation* = Default
>   # *Calling Party Number* = +14085252001
> + *ConnectedParty Transformations*
>   # *ConnectedLineId Presentation* = Default
>   # *ConnectedName Presentation* = Default
> + *Called Party Transformations*
>   # *Called Party Mask* =
>   # *Discard Digits Instruction* = PreDot
>   # *Prefix* = +
>   # *Called Number* = +1408425
>   o *Route Pattern* :*Pattern*= \+!
> + *Positional Match List* = +1408425
> + *DialPlan* =
> + *Route Filter*
>   # *Filter Name* =
>   # *Filter Clause* =
> + *Require Forced Authorization Code* = No
> + *Authorization Level* = 0
> + *Require Client Matter Code* = No
> + *Call Classification* =
> + *PreTransform Calling Party Number* = +14085252001
> + *PreTransform Called Party Number* = +1408425
> + *Calling Party Transformations*
>   # *External Phone Number Mask* = YES
>   # *Calling Party Mask* =
>   # *Prefix* =
>   # *CallingLineId Presentation* = Default
>   # *CallingName Presentation* = Default
>   # *Calling Party Number* = +14085252001
> + *ConnectedParty Transformations*
>   # *ConnectedLineId Presentation* = Default
>   # *ConnectedName Presentation* = Default
> + *Called Party Transformations*
>   # *Called Party Mask* =
>   # *Discard Digits Instruction* = None
>   # *Prefix* =
>   # *Called Number* = +1408425
>   o *Route List* :*Route List Name*= RL_LOCAL
> + *RouteGroup* :*RouteGroup Name*= Standard Local
>   Route Group
>   # *PreTransform Calling Party Number* =
+14085252001
>   # *PreTransform Called Party Number* =
+1408425
>   # *Calling Party Transformations*
>  

Re: [OSL | CCIE_Voice] MGCP Gateway Problem

[Jeff Price (jeffpric)]

Good morning everyone,

 

Thanks Jeff for the response.  I'm going to be looking into all of this.

 

However, my configuration is as follow:

HQ Phones - CSS_HQ_DEVICES (DEVICE, no LINE CSS as of yet)

Translation Patterns - PT_USA, CSS_GLOBAL

Route Pattern - PT_GLOBAL

 

CSSs

CSS_HQ_DEVICES: PT_HQ_DEVICES, PT_USA

CSS_GLOBAL: PT_GLOBAL

 

This brings me to the next question, why would DNA not allow me to use
it?  Every time I open up the DNA, I get a "DNA Service is still
initializing.  Please refresh after some time" or something like that.
Which if you look at the user guide from Cisco, that's normal.  But it
stays like that for over an hour.  I can't actually use the DNA.  I've
dis-enabled and then re-enabled the Database synchronization and then
restarted the service through the Control Center - Feature Services.
I've also gone to Service Activation and then Deactivated and then
Activated, and still the same message for long after it should have
stopped.

 

Has anyone seen this issue before?  Any suggestions as to how to get
this to work? 

 

Jeff

 

From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff
Garvas
Sent: Saturday, February 06, 2010 2:07 PM
To: Jeff Price (jeffpric)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem

 

Jeff,

If debug q931 isn't producing output and everything appears to be
registered right I'd assume that you have an issue with your phone/line
CSS, route patterns, route list, route group, etc.

Have you tried resetting the Route Group?   Have you tried the dialed
number analyzer to see where CUCM thinks the calls should be going for a
DN in the particular CSS in use, if anywhere?  DNA will tell you if you
have a failure in your design, but it won't tell you to go reset the
Route Group / Gateway etc.

 -Jeff




On Sat, Feb 6, 2010 at 3:44 PM, Jeff Price (jeffpric) <
jeffp...@cisco.com> wrote:

Hey everyone,

 

I wanted see if anyone had any ideas as to what may be wrong here.  I
have set up the MGCP gateway correctly as far as I can tell.  It is
registered with CUCM.  Attached is the GW's config as well as show
ccm-manager, show isdn status and the call routing elements.

 

Everything configuration wise seems to be working.  I have a registered
status in CUCM and on the GW.  Everything seems pretty standard with the
routing elements.

 

I have repeatedly done a no mgcp, mgcp to restart the mgcp process.  I
have physically unplugged the t1 cable to restart it.  I have deleted
and reconfigured the GW in CUCM as well as on the GW itself.  I changed
the "Stop Routing on Unallocated Number" Service Parameter to False
(which I know probably isn't effecting anything related to this).   I've
restarted and/or activated/deactivated all of the various related
feature services.   I'm sure there's more I've done as I've been trying
to figure this out for almost two days.  Also, the debug isdn q931 isn't
producing any output, which is odd to me.  I keep getting a fast busy
whenever I dial any numbers.  I don't have access to the PSTN router's
configuration, however I know it was working as I was able to make phone
calls to it when I first got here, but now that I've done my own
configuration it is not working anymore.  Also, one of the engineers who
set it up verified the configuration for me.  

 

Please help!  It is very much appreciated. J 

 

 



 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com
Phone: 408-525-8293
Mobile: 408-204-4510

Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page <http://www.cisco.com/> 

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 


___
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] MGCP Gateway Problem

[Jeff Price (jeffpric)]

Hey everyone,

 

I wanted see if anyone had any ideas as to what may be wrong here.  I
have set up the MGCP gateway correctly as far as I can tell.  It is
registered with CUCM.  Attached is the GW's config as well as show
ccm-manager, show isdn status and the call routing elements.

 

Everything configuration wise seems to be working.  I have a registered
status in CUCM and on the GW.  Everything seems pretty standard with the
routing elements.

 

I have repeatedly done a no mgcp, mgcp to restart the mgcp process.  I
have physically unplugged the t1 cable to restart it.  I have deleted
and reconfigured the GW in CUCM as well as on the GW itself.  I changed
the "Stop Routing on Unallocated Number" Service Parameter to False
(which I know probably isn't effecting anything related to this).   I've
restarted and/or activated/deactivated all of the various related
feature services.   I'm sure there's more I've done as I've been trying
to figure this out for almost two days.  Also, the debug isdn q931 isn't
producing any output, which is odd to me.  I keep getting a fast busy
whenever I dial any numbers.  I don't have access to the PSTN router's
configuration, however I know it was working as I was able to make phone
calls to it when I first got here, but now that I've done my own
configuration it is not working anymore.  Also, one of the engineers who
set it up verified the configuration for me.  

 

Please help!  It is very much appreciated. J 

 

 



 

 

Jeff Price
Network Consulting Engineer - Unified Communications Practice


jeffp...@cisco.com  
Phone: 408-525-8293
Mobile: 408-204-4510



Cisco Systems, Inc.
170 West Tasman Drive,
San Jose, CA 95134-1706
USA
Cisco home page  

 

 

 Think before you print.

 

This email may contain confidential and privileged material for the sole
use of the intended recipient. 

Any review, use, distribution or disclosure by others is strictly
prohibited. If you are not the intended 

recipient (or authorized to receive for the recipient), please contact
the sender by reply email and 

delete all copies of this message.

For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 

 

<><><>version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname R1
!
boot-start-marker
boot system flash:c2800nm-adventerprisek9-mz.124-24.T.bin
boot-end-marker
!
logging message-counter syslog
enable password 7 110A1A0C12
!
no aaa new-model
network-clock-participate wic 0
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
voice-card 0
!
!
!
!
!
archive
 log config
  hidekeys
!
!
!
!
!
controller T1 0/0/0
 pri-group timeslots 1-6,24 service mgcp
!
controller T1 0/0/1
!
!
!
!
!
interface GigabitEthernet0/0
 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
 no ip address
 duplex auto
 speed auto
!
interface GigabitEthernet0/0.102
 encapsulation dot1Q 102 native
 ip address 172.21.51.196 255.255.255.224
!
interface GigabitEthernet0/0.150
 encapsulation dot1Q 150
 ip address 10.5.100.1 255.255.255.0
!
interface GigabitEthernet0/0.250
 encapsulation dot1Q 250
 ip address 10.5.200.1 255.255.255.0
 ip helper-address 172.21.51.204
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 tx-ring-limit 128
 tx-queue-limit 128
 serial restart-delay 0
 frame-relay lmi-type ansi
!
interface Serial0/1/0.1 point-to-point
 ip address 162.5.101.1 255.255.255.0
 ip ospf mtu-ignore
 frame-relay interface-dlci 201
!
interface Serial0/1/0.2 point-to-point
 ip address 162.5.102.1 255.255.255.0
 ip ospf mtu-ignore
 frame-relay interface-dlci 202
!
router ospf 1
 log-adjacency-changes
 network 10.5.100.0 0.0.0.255 area 0
 network 10.5.200.0 0.0.0.255 area 0
 network 162.5.101.0 0.0.0.255 area 0
 network 162.5.102.0 0.0.0.255 area 0
 network 172.5.100.0 0.0.0.255 area 0
 network 172.21.51.193 0.0.0.0 area 0
 network 172.21.51.0 0.0.0.255 area 0
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 172.21.51.193
ip http server
no ip http secure-server
!
!
!
!
!
!
!
!
!
control-plane
!
!
!
voice-port 0/0/0:23
!
ccm-manager redundant-host 172.21.51.204
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 172.21.51.205
ccm-manager config
!
mgcp
mgcp call-agent 172.21.51.205 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capa

Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

[Jeff Price (jeffpric)]

Otto,

 

Thanks for getting back to me.  After spending a lot of time on this, I 
determined it was an underlying routing issue.  The lab I was using was 
mis-configured on the PUB/SUB with the incorrect Default GW.  Therefore, the 
DHCP server would work at Layer 2, but then when the phones would try to get 
their configs, they couldn't get a response from the PUB/SUB.  Also there may 
have been an inter-VLAN routing issue as well, but I had tried so many 
different things I can't recall if I had to fix that or not.  

 

Basically, if DB replication is working, it most likely will have to be an 
underlying routing problem for this error to occur is what I have found.

 

Jeff

 

From: Otto Sanchez [mailto:o...@ipexpert.com] 
Sent: Friday, February 05, 2010 5:43 AM
To: Jeff Price (jeffpric); kavi ten
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Hi Jeff, Kavi,

A cause for this error is that the auto registration number range has been 
exhausted (i.e. at some point in time the dn numbers have been assigned), 
please increase that range and try again, also, make sure your pub and sub have 
different ranges, 

If still with problems, I would think that the sub server is not getting the 
auto-registration settings from cisco unified cm configuration section, so as 
Roger mentioned this might be a replication issue,

Then try to force a database replication from the pub server's cli, if it still 
doesn't work, please send us the sub call manager service traces when the phone 
is auto registering,

Thanks,

On Thu, Feb 4, 2010 at 3:04 PM, Jeff Price (jeffpric)  
wrote:

Anyone have any ideas?  I'm still stuck here.  I can't find anything on google 
relating to this error, other than IP communicator stuff.  I don't believe its 
DB replication, because the Phones have replicated to the SUB when I am logged 
into the CM Administration page of the SUB.  However, I am in a lab that has 
restricted access to CLI, OS Admin Page, and DRS page, so I'm unable to verify 
other than logging into the SUB CM Admin page.

 

Thanks,

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price 
(jeffpric)
Sent: Thursday, February 04, 2010 9:55 AM
To: Roger Källberg; kavi ten; ccie_voice@onlinestudylist.com


Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Kavi,

 

I am having the same issue.  I will let you know if I have any success in 
finding a solution.  I'm asking if you don't mind doing the same.  Thank you 
all for your help.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sent: Thursday, February 04, 2010 9:39 AM
To: kavi ten; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Sound like a db replication issue or possibly, but less likely, the order of 
CPE in the call manager group.

 

Roger Källberg
Unified Communication Consultant
Cygate AB

 

From: kavi ten [mailto:kaucc...@gmail.com] 
Sent: den 4 februari 2010 14:23
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Hi Guys,

 

I have my DHCP server as Publisher

DHCP Server: PUB

 Primary  TFTP : 10.10.210.11

  Secondary TFTP : 10.10.210.10

 

 

Auto reguistration enabed for SUB with range specified.

 

Now the phone shown in the Devices--> Phones page but Status Rejected

On the Phone it shows Rejectration Rejected: Security Error

 

When I auto register with PUB it registers properly.

 

What could be the problem when Auto regiosteration is enabled in the SUB.

 

Thanks,

 


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




-- 
Regards,

Otto Sanchez 
CCIE #25592 (Voice) 
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com

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

Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

[Jeff Price (jeffpric)]

Anyone have any ideas?  I'm still stuck here.  I can't find anything on google 
relating to this error, other than IP communicator stuff.  I don't believe its 
DB replication, because the Phones have replicated to the SUB when I am logged 
into the CM Administration page of the SUB.  However, I am in a lab that has 
restricted access to CLI, OS Admin Page, and DRS page, so I'm unable to verify 
other than logging into the SUB CM Admin page.

 

Thanks,

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price 
(jeffpric)
Sent: Thursday, February 04, 2010 9:55 AM
To: Roger Källberg; kavi ten; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Kavi,

 

I am having the same issue.  I will let you know if I have any success in 
finding a solution.  I'm asking if you don't mind doing the same.  Thank you 
all for your help.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sent: Thursday, February 04, 2010 9:39 AM
To: kavi ten; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Sound like a db replication issue or possibly, but less likely, the order of 
CPE in the call manager group.

 

Roger Källberg
Unified Communication Consultant
Cygate AB

 

From: kavi ten [mailto:kaucc...@gmail.com] 
Sent: den 4 februari 2010 14:23
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Hi Guys,

 

I have my DHCP server as Publisher

DHCP Server: PUB

 Primary  TFTP : 10.10.210.11

  Secondary TFTP : 10.10.210.10

 

 

Auto reguistration enabed for SUB with range specified.

 

Now the phone shown in the Devices--> Phones page but Status Rejected

On the Phone it shows Rejectration Rejected: Security Error

 

When I auto register with PUB it registers properly.

 

What could be the problem when Auto regiosteration is enabled in the SUB.

 

Thanks,

 

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

Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

[Jeff Price (jeffpric)]

Kavi,

 

I am having the same issue.  I will let you know if I have any success in 
finding a solution.  I'm asking if you don't mind doing the same.  Thank you 
all for your help.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sent: Thursday, February 04, 2010 9:39 AM
To: kavi ten; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Sound like a db replication issue or possibly, but less likely, the order of 
CPE in the call manager group.

 

Roger Källberg
Unified Communication Consultant
Cygate AB

 

From: kavi ten [mailto:kaucc...@gmail.com] 
Sent: den 4 februari 2010 14:23
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected

 

Hi Guys,

 

I have my DHCP server as Publisher

DHCP Server: PUB

 Primary  TFTP : 10.10.210.11

  Secondary TFTP : 10.10.210.10

 

 

Auto reguistration enabed for SUB with range specified.

 

Now the phone shown in the Devices--> Phones page but Status Rejected

On the Phone it shows Rejectration Rejected: Security Error

 

When I auto register with PUB it registers properly.

 

What could be the problem when Auto regiosteration is enabled in the SUB.

 

Thanks,

 

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