Re: [OSL | CCIE_Voice] H323 COR ?

[Balamurugan Singaram]

Hi,
 
 
How can we block international (or specific calling number) call ONLY in SRST 
mode? in H323 gateway.
 
Thanks,
Bala.


--- On Mon, 6/10/08, Mike Brooks <[EMAIL PROTECTED]> wrote:

From: Mike Brooks <[EMAIL PROTECTED]>
Subject: Re: [OSL | CCIE_Voice] H323 COR ?
To: "kapil atrish" <[EMAIL PROTECTED]>, ccie_voice@onlinestudylist.com
Date: Monday, 6 October, 2008, 1:35 AM

Kapil,

I am referring to an H323 gateway not an MGCP gateway.  Therefore L3
info is not backhauled to CM.

Regards,

Mike Brooks
CCIE#16027 (R&S)

On Sun, Oct 5, 2008 at 3:58 PM, kapil atrish <[EMAIL PROTECTED]>
wrote:
> When not in SRST mode, all layer-3 information (DNIS, ANI) are back-hauled
> to CCM directly and COR won't trigger.
>
> Jacob Owen <[EMAIL PROTECTED]> wrote:
>
> Mike,
> I was under the impression since the call came into the H323 gateway from
> UCM (GW isn't in SRST) it wasn't "tagged" with an
incoming corlist and
> therefore could reach all remote PSTN numbers.  When the router drops back
> to SRST the phones would register with a corlist incoming and therefore be
> limited to where they could call.  Hopefully someone will let me know if I
> am incorrect.  You could also test this by adding a corlist incoming to
the
> inbound voip dial-peer and see if you can call.
>
> On Sun, Oct 5, 2008 at 12:35 PM, Mike Brooks <[EMAIL PROTECTED]>
wrote:
>>
>> If COR is configured on H323 dial-peers on an H323 gateway, is the
>> dial-peer COR only in affect when in SRST mode ?  If not, wouldn't
you
>> be performing COR twice  once on the CallManager and also on the
>> H323-GW ?
>>
>> for example:
>> phones/CSS > h323-gw inbound voip dial-peer (KEY) ---> 
h323gw
>> outbound pots dial-peer (LOCK)
>> or
>> h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
>> dial-peer (LOCK) --> h323-gw/CSS (on CM)
>>
>> If COR is in affect regardless of if it the site is in SRST mode
>> (which I assume it would be) should you just not configure COR
>> (keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?
>>
>> Regards,
>>
>> Mike Brooks
>> CCIE# 16027 (R&S)
>
>
>
> --
> Jacob Owen
> CCIE #14063 (R&S, Service Provider), CCDP, CCVP
>
>



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Re: [OSL | CCIE_Voice] SRST Voicemail Integration

[Balamurugan Singaram]

Hi All,
 
 
How can we block international (or specific calling number) call ONLY in SRST 
mode? in H323 gateway.
 
Thanks,
Bala.


--- On Mon, 6/10/08, Edi Hamlet <[EMAIL PROTECTED]> wrote:

From: Edi Hamlet <[EMAIL PROTECTED]>
Subject: Re: [OSL | CCIE_Voice] SRST Voicemail Integration
To: "Chris Parker" <[EMAIL PROTECTED]>, "Vikram Malhi" <[EMAIL PROTECTED]>
Cc: "" 
Date: Monday, 6 October, 2008, 7:13 AM






Hi Parker,

i think the cfw in alias will work if the PSTN accept 1212225 and pass it 
to HQ gateway. if the PSTN only accept 12122251xxx, then the cfw in alias will 
not work. i think the workaround if still want using alias is put cfw 
912122251xxx which is not already used in HQ, then use translation pattern to 
translate 1xxx to 200x. i haven't try this, but i think it's gonna work.

cmiiw..

cheers,
edi



- Original Message 
From: Chris Parker <[EMAIL PROTECTED]>
To: Vikram Malhi <[EMAIL PROTECTED]>
Cc: "" 
Sent: Sunday, October 5, 2008 3:50:50 AM
Subject: Re: [OSL | CCIE_Voice] SRST Voicemail Integration

The other method I've seen posted that has caught my interest involves 
setting the call forward noanswer/busy to a DID on the HQ PRI for each 
SRST phone using the alias command under call-manager-fallback. Then in 
UCM putting those DIDs on a CTI route point with call forward all to 
voicemail.

So basically on BR1:

call-manager-fallback
alias 1 2001 to 2001 cfw 912122252001 timeout 4
alias 1 2002 to 2002 cfw 912122252002 timeout 4
alias 1 2003 to 2003 cfw 912122252003 timeout 4

So for the extensions 2001-2003 at BR1 calls get forwarded to 
12122252001-3. You have a pots peer than puts those back out to the 
PSTN. They ring in on the 6608 PRI and if signifcant digits are set to 
four in the gateway config UCM will try and send the call to 2001-2002 
respectively. UCM sees the extension as OOS and sends it on to voicemail.

In the case where you dont have a DID on the HQ PRI that matches the BR1 
number on the last 4 digits you can do the same thing but you have to 
set up a CTI route point for each number that is forwarded to voice mail 
and then transform the number on UCM before it goes to Unity or use an 
alternate extension in Unity.

Chris



Vikram Malhi wrote:
> Know all possible workarounds...I personally don't like the 
> vm-integration method. Do you know any other methods?
>
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
>
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED]
>
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities
> IPexpert - The Global Leader in Self-Study, Classroom-Based, 
> Video-On-Demand and Audio Certification Training Tools for the Cisco 
> CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE 
> Voice Lab and CCIE Storage Lab Certifications.
>
>
>
> On Oct 4, 2008, at 6:46 AM, Chris Parker wrote:
>
>> Hello,
>>
>> I have been reviewing the methods of voice mail fall back with SRST, 
>> and I am wondering which method will actually work in the Lab? It 
>> seems that success relies on the behavior of PSTN. The 
>> "vm-integration" method seems to work fine on the Proctor Labs gear, 
>> but will that translate to the real lab? What is the safest / best 
>> way to do this?
>>
>> Chris Parker
>
>
>





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[OSL | CCIE_Voice] CallManager Security and Windows CA Server

[Paul and Bobs]

Hi Guys

Is it possible to run the CallManager in security mode and run SRTP
with secure phones without the Cisco USB token . Possibly using a
windows CA to creat the certificates.

Paul

Re: [OSL | CCIE_Voice] SRST Voicemail Integration

[Edi Hamlet]

Hi Parker,

i think the cfw in alias will work if the PSTN accept 1212225 and pass it 
to HQ gateway. if the PSTN only accept 12122251xxx, then the cfw in alias will 
not work. i think the workaround if still want using alias is put cfw 
912122251xxx which is not already used in HQ, then use translation pattern to 
translate 1xxx to 200x. i haven't try this, but i think it's gonna work.

cmiiw..

cheers,
edi



- Original Message 
From: Chris Parker <[EMAIL PROTECTED]>
To: Vikram Malhi <[EMAIL PROTECTED]>
Cc: "" 
Sent: Sunday, October 5, 2008 3:50:50 AM
Subject: Re: [OSL | CCIE_Voice] SRST Voicemail Integration

The other method I've seen posted that has caught my interest involves 
setting the call forward noanswer/busy to a DID on the HQ PRI for each 
SRST phone using the alias command under call-manager-fallback. Then in 
UCM putting those DIDs on a CTI route point with call forward all to 
voicemail.

So basically on BR1:

call-manager-fallback
alias 1 2001 to 2001 cfw 912122252001 timeout 4
alias 1 2002 to 2002 cfw 912122252002 timeout 4
alias 1 2003 to 2003 cfw 912122252003 timeout 4

So for the extensions 2001-2003 at BR1 calls get forwarded to 
12122252001-3. You have a pots peer than puts those back out to the 
PSTN. They ring in on the 6608 PRI and if signifcant digits are set to 
four in the gateway config UCM will try and send the call to 2001-2002 
respectively. UCM sees the extension as OOS and sends it on to voicemail.

In the case where you dont have a DID on the HQ PRI that matches the BR1 
number on the last 4 digits you can do the same thing but you have to 
set up a CTI route point for each number that is forwarded to voice mail 
and then transform the number on UCM before it goes to Unity or use an 
alternate extension in Unity.

Chris



Vikram Malhi wrote:
> Know all possible workarounds...I personally don't like the 
> vm-integration method. Do you know any other methods?
>
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
>
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED]
>
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities
> IPexpert - The Global Leader in Self-Study, Classroom-Based, 
> Video-On-Demand and Audio Certification Training Tools for the Cisco 
> CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE 
> Voice Lab and CCIE Storage Lab Certifications.
>
>
>
> On Oct 4, 2008, at 6:46 AM, Chris Parker wrote:
>
>> Hello,
>>
>> I have been reviewing the methods of voice mail fall back with SRST, 
>> and I am wondering which method will actually work in the Lab? It 
>> seems that success relies on the behavior of PSTN. The 
>> "vm-integration" method seems to work fine on the Proctor Labs gear, 
>> but will that translate to the real lab? What is the safest / best 
>> way to do this?
>>
>> Chris Parker
>
>
>


  

Re: [OSL | CCIE_Voice] H323 COR ?

[Edi Hamlet]

If COR is assign to incoming dial-peer (cor incoming) & outgoing dial-peer (cor 
outgoing), the COR will take affect event when the gateway is not in SRST mode.

My suggestions are:
1. Do configure outgoing dial-peer to PSTN with COR
2. Do not configure incoming dial-peer from CCM with COR
3. Let the COR comes to H323 gateway controlled by CCM using PT & CSS
4. Do configure COR for SRST mode



- Original Message 
From: Mike Brooks <[EMAIL PROTECTED]>
To: CCIE Voice Maillist 
Sent: Sunday, October 5, 2008 11:35:18 PM
Subject: [OSL | CCIE_Voice] H323 COR ?

If COR is configured on H323 dial-peers on an H323 gateway, is the
dial-peer COR only in affect when in SRST mode ?  If not, wouldn't you
be performing COR twice  once on the CallManager and also on the
H323-GW ?

for example:
phones/CSS > h323-gw inbound voip dial-peer (KEY) --->  h323gw
outbound pots dial-peer (LOCK)
or
h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
dial-peer (LOCK) --> h323-gw/CSS (on CM)

If COR is in affect regardless of if it the site is in SRST mode
(which I assume it would be) should you just not configure COR
(keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?

Regards,

Mike Brooks
CCIE# 16027 (R&S)



  

Re: [OSL | CCIE_Voice] Antw: Bandwidth Values used for FRTS

[Mark Snow]

Nope - you didn't have a typo - I just 'assumed' G729 and skipped  
actually 'reading' the part that clearly said g711.

:P

Sorry,

--
Mark Snow
CCIE #14073 (Voice, Security)

Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: [EMAIL PROTECTED]
--
Join our free online support and peer group communities: 
http://www.IPexpert.com/communities
--
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- 
Demand and Audio Certification Training Tools for the Cisco CCIE R&S  
Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
CCIE Storage Lab Certifications.

--

On Oct 4, 2008, at 7:08 AM, Robert Schuknecht wrote:


Hi Mark,

i think you have overread (dont know if it´s the right word)  
something in my  Mail. The Payload of G.729 is, of course, 20Byte  
and for G.711 it is 160Bytes. Or maybe i had a Typo in my mail. If  
it was a Typo, i am totally sorry.


/Robert



Mark Snow<[EMAIL PROTECTED]> schrieb am Freitag, 3. Oktober 2008  
um 22:13 in

Nachricht aa86984e555f26f5e9bbbfc53b42ff94:

Your calculations below are exactly correct.
Not sure where the gentleman below your email, Robert, heard us say
G729 payload was 160bytes?? It is 20 as you accurately state below -
at least at it's default sampling of 20ms (and thus 50pps).

I personally, however, would do the calcs in the lab - and only round
up once you had multiplied your "Per Call" BW by the number of calls.

That way your 10% margin of error is much closer! :)

Cheers

[OSL | CCIE_Voice] IPIPGW question 4.9 - Calling does not work

[Chris Kagadis (kagadis.com)]

Hello,
 In working through this problem and referencing the solution, I noticed
that the solution does not include dial-peers so that phones at BR2 at able
to call HQ phones via IPIPGW.  I'm not sure if this is an omission or if the
syntax has been added elsewhere.  The two problems I have are;

1) When an HQ phone calls BR2, the BR2 phone rings but hangs up when the BR2
goes off hook (call terminated)
2) BR2 can not call HQ phones at all (without adding dial peers)

So, I need help understanding what has to be done on BR2 in order to set up
a call to HQ's IPIPGW.  I've set up the following dial-peers on BR2, but
they do not work (fast busy)

dial-peer voice 2 voip
 incoming called-number 3...
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 3 voip
 destination-pattern 3...
 session protocol sipv2
 session target ipv4:10.1.202.1
 dtmf-relay rtp-nte digit-drop h245-alphanumeric
!
dial-peer voice 4 voip
 incoming called-number [12]...
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 5 voip
 destination-pattern [12]...
 session protocol sipv2
 session target ipv4:10.3.200.1
 dtmf-relay rtp-nte digit-drop h245-alphanumeric

Can someone help?

-- 
Chris Kagadis

Re: [OSL | CCIE_Voice] H323 COR ?

[Mike Brooks]

Kapil,

I am referring to an H323 gateway not an MGCP gateway.  Therefore L3
info is not backhauled to CM.

Regards,

Mike Brooks
CCIE#16027 (R&S)

On Sun, Oct 5, 2008 at 3:58 PM, kapil atrish <[EMAIL PROTECTED]> wrote:
> When not in SRST mode, all layer-3 information (DNIS, ANI) are back-hauled
> to CCM directly and COR won't trigger.
>
> Jacob Owen <[EMAIL PROTECTED]> wrote:
>
> Mike,
> I was under the impression since the call came into the H323 gateway from
> UCM (GW isn't in SRST) it wasn't "tagged" with an incoming corlist and
> therefore could reach all remote PSTN numbers.  When the router drops back
> to SRST the phones would register with a corlist incoming and therefore be
> limited to where they could call.  Hopefully someone will let me know if I
> am incorrect.  You could also test this by adding a corlist incoming to the
> inbound voip dial-peer and see if you can call.
>
> On Sun, Oct 5, 2008 at 12:35 PM, Mike Brooks <[EMAIL PROTECTED]> wrote:
>>
>> If COR is configured on H323 dial-peers on an H323 gateway, is the
>> dial-peer COR only in affect when in SRST mode ?  If not, wouldn't you
>> be performing COR twice  once on the CallManager and also on the
>> H323-GW ?
>>
>> for example:
>> phones/CSS > h323-gw inbound voip dial-peer (KEY) --->  h323gw
>> outbound pots dial-peer (LOCK)
>> or
>> h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
>> dial-peer (LOCK) --> h323-gw/CSS (on CM)
>>
>> If COR is in affect regardless of if it the site is in SRST mode
>> (which I assume it would be) should you just not configure COR
>> (keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?
>>
>> Regards,
>>
>> Mike Brooks
>> CCIE# 16027 (R&S)
>
>
>
> --
> Jacob Owen
> CCIE #14063 (R&S, Service Provider), CCDP, CCVP
>
>

[OSL | CCIE_Voice] Fw: AAR in CCM Cluster

[Michael Shavrov]

Hi David,

Do you have gateways registered to both, Pub and Sub? Would you check the 
gateways status when the publisher id down?

Sincerely,

Mike
http://cciev.headsetadapter.com
  - Original Message - 
  From: David Corbeil 
  To: 'ccie_voice@onlinestudylist.com' 
  Sent: Sunday, October 05, 2008 2:44 PM
  Subject: [OSL | CCIE_Voice] AAR in CCM Cluster


  Hi,

   

  Someone knows why when the publisher is down, the aar is not working?

   

  I have a busy after I shutdown the ccm pub service.

   

  Thanks

   

  David Corbeil

  Consultant en technologie | Technology Consultant

  Tel. 514-215-2490 | Fax. 514-748-5333

  Membre de l'équipe TELUS

   

Re: [OSL | CCIE_Voice] H323 COR ?

[kapil atrish]

When not in SRST mode, all layer-3 information (DNIS, ANI) are back-hauled to 
CCM directly and COR won't trigger.

Jacob Owen <[EMAIL PROTECTED]> wrote: Mike,
I was under the impression since the call came into the H323 gateway from UCM 
(GW isn't in SRST) it wasn't "tagged" with an incoming corlist and therefore 
could reach all remote PSTN numbers.  When the router drops back to SRST the 
phones would register with a corlist incoming and therefore be limited to where 
they could call.  Hopefully someone will let me know if I am incorrect.  You 
could also test this by adding a corlist incoming to the inbound voip dial-peer 
and see if you can call.  
 
On Sun, Oct 5, 2008 at 12:35 PM, Mike Brooks <[EMAIL PROTECTED]> wrote:
 If COR is configured on H323 dial-peers on an H323 gateway, is the
 dial-peer COR only in affect when in SRST mode ?  If not, wouldn't you
 be performing COR twice  once on the CallManager and also on the
 H323-GW ?
 
 for example:
 phones/CSS > h323-gw inbound voip dial-peer (KEY) --->  h323gw
 outbound pots dial-peer (LOCK)
 or
 h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
 dial-peer (LOCK) --> h323-gw/CSS (on CM)
 
 If COR is in affect regardless of if it the site is in SRST mode
 (which I assume it would be) should you just not configure COR
 (keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?
 
 Regards,
 
 Mike Brooks
 CCIE# 16027 (R&S)
 



-- 
Jacob Owen
CCIE #14063 (R&S, Service Provider), CCDP, CCVP
 
 

   

[OSL | CCIE_Voice] AAR in CCM Cluster

[David Corbeil]

Hi,

Someone knows why when the publisher is down, the aar is not working?

I have a busy after I shutdown the ccm pub service.

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-215-2490 | Fax. 514-748-5333
Membre de l'équipe TELUS

Re: [OSL | CCIE_Voice] H323 COR ?

[Jacob Owen]

Mike,
I was under the impression since the call came into the H323 gateway from
UCM (GW isn't in SRST) it wasn't "tagged" with an incoming corlist and
therefore could reach all remote PSTN numbers.  When the router drops back
to SRST the phones would register with a corlist incoming and therefore be
limited to where they could call.  Hopefully someone will let me know if I
am incorrect.  You could also test this by adding a corlist incoming to the
inbound voip dial-peer and see if you can call.

On Sun, Oct 5, 2008 at 12:35 PM, Mike Brooks <[EMAIL PROTECTED]> wrote:

> If COR is configured on H323 dial-peers on an H323 gateway, is the
> dial-peer COR only in affect when in SRST mode ?  If not, wouldn't you
> be performing COR twice  once on the CallManager and also on the
> H323-GW ?
>
> for example:
> phones/CSS > h323-gw inbound voip dial-peer (KEY) --->  h323gw
> outbound pots dial-peer (LOCK)
> or
> h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
> dial-peer (LOCK) --> h323-gw/CSS (on CM)
>
> If COR is in affect regardless of if it the site is in SRST mode
> (which I assume it would be) should you just not configure COR
> (keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?
>
> Regards,
>
> Mike Brooks
> CCIE# 16027 (R&S)
>



-- 
Jacob Owen
CCIE #14063 (R&S, Service Provider), CCDP, CCVP

[OSL | CCIE_Voice] QoS Question class-default?

[Ryan Hicks]

QoS - Question.

After setting up a policy to macth RTP and signalling in a LLQ policy in which 
I used 96k for rtp and 8k for SIG across the WAN.
The PVC is 384k. 

What if another question was asked to make sure that the remaining traffic's 
dscp marking must be maintained if the pvc is utilized under 60%. If the 
remaining traffic's utilization goes over 60% of the pvc speed then set the 
dscp to 0.

Any ideas?

I am assuming it be would be reffering to the class class-default. But I am not 
sure when this would be enaged when traffic is flowing through the WAN port. 
So I guess I need a better understanding of how what takes place and how I 
could configure it from the question abouve.


class-map match-any RTP
 match ip dscp ef
class-map match-any SIG
 match ip dscp cs3
 match ip dscp af31
!
policy-map LLQ
 class RTP
  priority 96
 class SIG
  bandwidth 8
class class-default
 fair-queue
 set ip dscp 0
!
int s0/0
 bandwidth 384
 service-policy out LLQ
-
Ryan Hicks

[OSL | CCIE_Voice] H323 COR ?

[Mike Brooks]

If COR is configured on H323 dial-peers on an H323 gateway, is the
dial-peer COR only in affect when in SRST mode ?  If not, wouldn't you
be performing COR twice  once on the CallManager and also on the
H323-GW ?

for example:
phones/CSS > h323-gw inbound voip dial-peer (KEY) --->  h323gw
outbound pots dial-peer (LOCK)
or
h323-gw inbound pots dial-peer (KEY) --> h323-gw outbound voip
dial-peer (LOCK) --> h323-gw/CSS (on CM)

If COR is in affect regardless of if it the site is in SRST mode
(which I assume it would be) should you just not configure COR
(keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ?

Regards,

Mike Brooks
CCIE# 16027 (R&S)

Re: [OSL | CCIE_Voice] GK calls failed - Receive wrong digit fromPSTN-WAN

[David Corbeil]

Not this time

I prefix 011331322 because I now on pod 12

And I receive that "Called Number=33132433313223001"

So I receive this extra digit "3313243"

So past Thursday a tech call me from IP EXPERT to change Translation in 
PSTN-WAN to fix the problem.

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-215-2490 | Fax. 514-748-5333
Membre de l'équipe TELUS

From: Michael Shavrov [mailto:[EMAIL PROTECTED]
Sent: Sunday, October 05, 2008 11:24 AM
To: David Corbeil
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] GK calls failed - Receive wrong digit 
fromPSTN-WAN

Hi David,

First of all, you should carefully check your "voice translation-rule" and your 
"num-exp". Looks like you have some runaway translations, which insert and/or 
drop digits into your dialing string.

When you look really close to your digits, you will notice that they are not 
that much "garbage":

1. You dial "3001" with the prefix "001331328", which results as:
011-331-328-3001 (by the way, what is the tech-prefix???)

2. Your "debug vpm signal" shows similar digits with some "missing and some 
"extra" digits within:[missing 3] 31 [missing 3] 2 [extra 4] 8-3001

3. And your "debug voice ccapi inout" shows: 331 324 8300 (another 
representation: 331-32 [extra 4] 8-300 [missing 1])

Good luck,

Mike

http://cciev.headsetadapter.com


- Original Message -
From: David Corbeil
To: 'ccie_voice@onlinestudylist.com'
Sent: Thursday, October 02, 2008 11:32 AM
Subject: [OSL | CCIE_Voice] GK calls failed - Receive wrong digit fromPSTN-WAN

I need help,

When I try to call BR2 via GK, I receive garbage of digit on my E2

Example:

I dial 3001- prefix 011331328 and the call go through my HQ-RTR and PSTN-WAN

I do
-debug vpm signal

I receive 312483001

-debug voice ccapi inout

I receive 331 324 8300

But for BR2 I'm suppose to receive 3313283001

Here the configuration of my HQ-RTR

Gatekeeper
zone local HQ-RTR ipexpert.com 172.28.100.1
zone remote PSTN-WAN ipexpert.com 10.28.200.2 1719
zone prefix PSTN-WAN 011*
no shutdown

Thanks


David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-215-2490 | Fax. 514-748-5333
Membre de l'équipe TELUS

Re: [OSL | CCIE_Voice] GK calls failed - Receive wrong digit fromPSTN-WAN

[Michael Shavrov]

Hi David,

First of all, you should carefully check your "voice translation-rule" and your 
"num-exp". Looks like you have some runaway translations, which insert and/or 
drop digits into your dialing string. 

When you look really close to your digits, you will notice that they are not 
that much "garbage":

1. You dial "3001" with the prefix "001331328", which results as:
011-331-328-3001 (by the way, what is the tech-prefix???)

2. Your "debug vpm signal" shows similar digits with some "missing and some 
"extra" digits within:[missing 3] 31 [missing 3] 2 [extra 4] 8-3001

3. And your "debug voice ccapi inout" shows: 331 324 8300 (another 
representation: 331-32 [extra 4] 8-300 [missing 1])

Good luck,

Mike

http://cciev.headsetadapter.com


- Original Message - 
  From: David Corbeil 
  To: 'ccie_voice@onlinestudylist.com' 
  Sent: Thursday, October 02, 2008 11:32 AM
  Subject: [OSL | CCIE_Voice] GK calls failed - Receive wrong digit fromPSTN-WAN


  I need help,

   

  When I try to call BR2 via GK, I receive garbage of digit on my E2

   

  Example:

   

  I dial 3001- prefix 011331328 and the call go through my HQ-RTR and PSTN-WAN

   

  I do 

  -debug vpm signal

   

  I receive 312483001

   

  -debug voice ccapi inout

   

  I receive 331 324 8300

   

  But for BR2 I'm suppose to receive 3313283001

   

  Here the configuration of my HQ-RTR

   

  Gatekeeper

  zone local HQ-RTR ipexpert.com 172.28.100.1

  zone remote PSTN-WAN ipexpert.com 10.28.200.2 1719

  zone prefix PSTN-WAN 011*

  no shutdown

   

  Thanks

   

   

  David Corbeil

  Consultant en technologie | Technology Consultant

  Tel. 514-215-2490 | Fax. 514-748-5333

  Membre de l'équipe TELUS

   

Re: [OSL | CCIE_Voice] Cat6K T1 Ports fail to register

[Edi Hamlet]

I also had same problem with once...

try do disable all the port
!
set port disable 7/2-4
!

then change dhcp enable to static for each port
!
set port voice interface 7/2 dhcp disable 10.10.10.1/24 gateway 10.10.10.10 
tftp 10.10.10.10 dns cisco.com
!

then change back static to dhcp enable
!
set port voice interface 7/2 dhcp enable vlan xxx (do not forget the vlan, 
cause port will not get the dhcp broadcast)
!

then enable the port
!
set port enable 7/2-4
!

verify each port get the dhcp ip address with correct tftp ip address & CCMs ip 
address.


hope this help.

cheers,
edi



- Original Message 
From: Kapil Atrish <[EMAIL PROTECTED]>
To: ccie_voice@onlinestudylist.com
Sent: Sunday, October 5, 2008 8:53:50 PM
Subject: [OSL | CCIE_Voice] Cat6K T1 Ports fail to register

 HI,

I am getting following when trying to register Cat6K port to CCM on Pod 20. 
I've tried enabling/disabling the ports but all three ports (T1, Xcode and CFB) 
are in same state. Reset the DHCP service, other devices are able to take IP 
Address from DHCP and enough IPs are available in the scope. Can't clear CDP 
table due to insufficient privileges.


ort  Name Status Vlan   Duplex Speed   Type
-  -- -- -- --- 
 7/4  POD20-PSTN-T1enabled400  full - unknown

Port DHCPMAC-Address   IP-Address  Subnet-Mask
 --- - --- ---
 7/4 enable  00-d0-c0-d3-12-c3 (Failed to obtain port interface information)

Appreciate any help...


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[OSL | CCIE_Voice] Help on Q24.26

[David Corbeil]

How you can reach the voicemail, when you have a WAN outage?

You can't dial 1600 directly like that, you need a dial-peer to call outside 
and reach the voicemail by the PSTN.

Can you confirm that 24.26 is not a good answer and missing some configuration ?

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-215-2490 | Fax. 514-748-5333
Membre de l'équipe TELUS

Re: [OSL | CCIE_Voice] GK calls failed - Receive wrong digit from PSTN-WAN

[David Corbeil]

Still have the same problem,

PSTN-WAN GK send extra digit to my E1 (BR2 Site)

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-215-2490 | Fax. 514-748-5333
Membre de l'équipe TELUS

From: David Corbeil
Sent: Thursday, October 02, 2008 11:32 AM
To: 'ccie_voice@onlinestudylist.com'
Subject: GK calls failed - Receive wrong digit from PSTN-WAN

I need help,

When I try to call BR2 via GK, I receive garbage of digit on my E2

Example:

I dial 3001- prefix 011331328 and the call go through my HQ-RTR and PSTN-WAN

I do
-debug vpm signal

I receive 312483001

-debug voice ccapi inout

I receive 331 324 8300

But for BR2 I'm suppose to receive 3313283001

Here the configuration of my HQ-RTR

Gatekeeper
zone local HQ-RTR ipexpert.com 172.28.100.1
zone remote PSTN-WAN ipexpert.com 10.28.200.2 1719
zone prefix PSTN-WAN 011*
no shutdown

Thanks


David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-215-2490 | Fax. 514-748-5333
Membre de l'équipe TELUS

[OSL | CCIE_Voice] Cat6K T1 Ports fail to register

[Kapil Atrish]

HI,

I am getting following when trying to register Cat6K port to CCM on Pod 20. 
I've tried enabling/disabling the ports but all three ports (T1, Xcode and CFB) 
are in same state. Reset the DHCP service, other devices are able to take IP 
Address from DHCP and enough IPs are available in the scope. Can't clear CDP 
table due to insufficient privileges.


ort  Name Status Vlan   Duplex Speed   Type
-  -- -- -- --- 
 7/4  POD20-PSTN-T1enabled400  full - unknown

Port DHCPMAC-Address   IP-Address  Subnet-Mask
 --- - --- ---
 7/4 enable  00-d0-c0-d3-12-c3 (Failed to obtain port interface information)

Appreciate any help...


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[OSL | CCIE_Voice] where is "Cisco IP Communicator Administration Tool" located? I want to install it but where is the file?

[jeremy co]

Hi,

Corporate Directory search on my cisco ip communicator doesn't work, so I
want to run Directory Wizard and in order to run it I should run Cisco
IP Communicator
Administration Tool ,but  where is this tool located ?,I couldn't  find any
info in Docs.


Jeremy