Re: [OSL | CCIE_Voice] QoS Policy Map

[Daniel Berlinski]

and you wont see any output form show policy map interface because in these
switches it is rubish - left over from developers.

if you want to see any stats incrementing on 3750 or 3560 you better use
show mls qos interface statistics.

by the way your acl could be only matching half of the traffic.

2001 and 2002 is skinny analog stuff, you may not need it.

It would be safer to permit traffic both ways. alternatively you could also
attach this interface not only to the cucm port but also to the phones port.


On Sat, Oct 9, 2010 at 6:49 PM, Mark Holloway  wrote:

> I'm trying to create a policy map that matches the skinny signaling
> protocol that will police it and rewrite the exceeded packets from dscp 24
> to 0.  I am pretty sure I have the policy map created correctly but when I
> do 'show policy-map interface ' I am not seeing the counters
> increment.  Am I missing something?
>
> ## Cat 3750 ##
>
> mls qos map policed-dscp 24 to 0
>
> access-list 100 remark SKINNY
> access-list 100 permit tcp any eq 2000 any
> access-list 100 permit tcp any eq 2001 any
> access-list 100 permit tcp any eq 2002 any
>
> class-map match-any class-map-skinny
>  match access-group 100
>
> policy-map policy-map-voip-signal
>  class class-map-skinny
>  set dscp cs3
>  police 32000 8000 exceed-action policed-dscp-transmit
>
> interface FastEthernet1/0/1
>  description Trunk Port to Router
>  switchport trunk encapsulation dot1q
>  switchport mode trunk
>  service-policy input policy-map-voip-signal
>
> sw1#show policy-map int fast 1/0/1
>  FastEthernet1/0/1
>
>  Service-policy input: policy-map-voip-signal
>
>Class-map: class-map-skinny (match-any)
>  0 packets, 0 bytes
>  5 minute offered rate 0 bps, drop rate 0 bps
>  Match: access-group 100
>0 packets, 0 bytes
>5 minute rate 0 bps
>
>Class-map: class-default (match-any)
>  0 packets, 0 bytes
>  5 minute offered rate 0 bps, drop rate 0 bps
>  Match: any
>0 packets, 0 bytes
>5 minute rate 0 bps
>
> sw1#show mls qos
> QoS is enabled
> QoS ip packet dscp rewrite is enabled
>
> ___
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>
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[OSL | CCIE_Voice] QoS Policy Map

[Mark Holloway]

I'm trying to create a policy map that matches the skinny signaling protocol 
that will police it and rewrite the exceeded packets from dscp 24 to 0.  I am 
pretty sure I have the policy map created correctly but when I do 'show 
policy-map interface ' I am not seeing the counters increment.  Am I 
missing something?

## Cat 3750 ##

mls qos map policed-dscp 24 to 0

access-list 100 remark SKINNY
access-list 100 permit tcp any eq 2000 any
access-list 100 permit tcp any eq 2001 any
access-list 100 permit tcp any eq 2002 any

class-map match-any class-map-skinny
 match access-group 100

policy-map policy-map-voip-signal
 class class-map-skinny
  set dscp cs3
  police 32000 8000 exceed-action policed-dscp-transmit

interface FastEthernet1/0/1
 description Trunk Port to Router
 switchport trunk encapsulation dot1q
 switchport mode trunk
 service-policy input policy-map-voip-signal

sw1#show policy-map int fast 1/0/1
 FastEthernet1/0/1 

  Service-policy input: policy-map-voip-signal

Class-map: class-map-skinny (match-any)
  0 packets, 0 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: access-group 100
0 packets, 0 bytes
5 minute rate 0 bps

Class-map: class-default (match-any)
  0 packets, 0 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: any 
0 packets, 0 bytes
5 minute rate 0 bps

sw1#show mls qos
QoS is enabled
QoS ip packet dscp rewrite is enabled

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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 96

[Sriharshaa Prabhakar]

Hi Prashanth &  Kobelski,

Even if you may want to use the Cisco Unity Connection for recording custom 
prompts for UCCX, by the way is it possible to download these prompts from 
CUCXN? and J2SE Runtime Environment is required to ensure that we can record 
the customer messages...

Regards,
Sriharshaa Prabhakar|

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[OSL | CCIE_Voice] Cannot route call through GK

[David Lee]

Hello,

Can anyone offer some suggestion on what might be the cause for the
gatekeeper call routing I'm having?  Thx.  /D



I have a GK with 3 UCM H225 trunks on it.  (i.e. CM-A 2XXX, CM-B 7XXX, and
CM-C 8XXX.)

CM-C can call CM-B and CM-A
CM-B can call CM-C and CM-A
CM-A can call CM-B, but cannot call CM-C.

Essentially, I cannot call from ext 2xxx to 8xxx, but I am able to dial from
7xxx to 8xxx.


This is the gatekeeper config

gatekeeper
 zone local CM-B DBCMYZFVOIP.COM 10.25.208.14
 zone local CM-C DBCMYZFVOIP.COM
 zone local CM-B DBCMYZFVOIP.COM
 zone local CM-A DBCMYZFVOIP.COM
 zone subnet CM-B 10.25.208.10/32 enable
 zone subnet CM-B 10.25.208.11/32 enable
 zone subnet CM-C 10.25.224.151/32 enable
 zone subnet CM-C 10.25.224.152/32 enable
 no zone subnet CM-C 10.25.208.0/21 enable
 zone prefix CM-A 2...
 zone prefix CM-B 73..
 zone prefix CM-C 8...
 gw-type-prefix 1#* default-technology
 bandwidth interzone zone CM-C 2000
 bandwidth interzone zone CM-A 64
 no shutdown


This is the main 10.  It seems that a technology GW is selected...

YZF-SC3-COM1-3-GK-01#
*Oct  9 04:06:09.752: gk_process: QUEUE_EVENT (minor 0) wakeup
*Oct  9 04:06:09.752: gk_rassrv_arq: arqp=0x45CD67C8, crv=0x1A3,
answerCall=0
*Oct  9 04:06:09.752: gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC
*Oct  9 04:06:09.752: gk_dns_query: No Name servers
*Oct  9 04:06:09.752: rassrv_get_addrinfo: (8916) Tech-prefix match failed.
*Oct  9 04:06:09.752: rassrv_get_addrinfo: (8916) Matched zone prefix 8 and
remainder 916
*Oct  9 04:06:09.752: rassrv_arq_select_viazone: about to check the source
side, src_zonep=0x470A1D50
*Oct  9 04:06:09.752: rassrv_arq_select_viazone: matched zone is CM-A, and
z_invianamelen=0
*Oct  9 04:06:09.752: rassrv_arq_select_viazone: about to check the
destination side, dst_zonep=0x470A1890
*Oct  9 04:06:09.752: rassrv_arq_select_viazone: matched zone is CM-C, and
z_outvianamelen=0
*Oct  9 04:06:09.752: rassrv_get_addrinfo: No tech prefix

*Oct  9 04:06:09.752: rassrv_get_addrinfo: Alias not found

*Oct  9 04:06:09.752: gk_zone_get_proxy_usage: local zone= CM-C, remote
zone= CM-A, call direction= 0, eptype= 2050 be_entry= 0
*Oct  9 04:06:09.752: gk_zone_get_proxy_usage: returns proxied = 0
*Oct  9 04:06:09.752: gk_gw_select_px: Source and destination endpoints in
different local zones
*Oct  9 04:06:09.752: gk_zone_get_proxy_usage: local zone= CM-A, remote
zone= CM-C, call direction= 1, eptype= 2050 be_entry= 0
*Oct  9 04:06:09.752: gk_zone_get_proxy_usage: returns proxied = 0
*Oct  9 04:06:09.752: rassrv_get_addrinfo: Technology GW selected

*Oct  9 04:06:12.268: gk_process: got a TIMER event

*Oct  9 04:06:12.268: gk_handle_timers

*Oct  9 04:06:12.268: gk_handle_timers: managed timer expired 0x45962220
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[OSL | CCIE_Voice] RES: Call Forward Unregistered

[Marcelo Alexandria]

Mark , dont worry,This is a image issue..i got this many times in my
labs…

 

 

De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Em nome de Mark Holloway
Enviada em: sexta-feira, 8 de outubro de 2010 20:15
Para: Mark Holloway
Cc: CCIE Voice Maillist
Assunto: Re: [OSL | CCIE_Voice] Call Forward Unregistered

 

I have had it working before, but it's odd because sometimes when I reset
the lab rack I can get it work and other times it does not work the way I
want.  I'm trying to figure out if I keep overlooking something.

 

 

On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote:





I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking
CFUR) which is the Redirecting number. 

 

 

On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote:





Hi Mark,
 
The easiest way is to use calling party Transformation on the outbound
gateway.
 
For example - 5002 calling 3002 out of local gateway. create a pt and assign
it to a css. Assign css to the gateway "calling party transformation css"
and uncheck use dp box. Now create a calling party transformation for 5XXX
in the pt and modify the ANI to use extenal mask. 
 
This will modify the ANI from 5xxx to external mask everytime the 5xxx makes
a call out of that gateway.
 
HTH
Prashant

On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway  wrote:

I'm trying to get my CFUR to work so it shows the External Mask in the For
and By part of the call presentation but instead I am only getting it to
show the 4 digit extension.  For example, lets say HQ 5001 calls BR1 3001
(3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because
that site is in SRST mode).  The presentation on the BR1 phones is Forwarded
HqPh1 5001, For 3001 By 3001.  Instead of 3001 I want to display the
External Mask.  Does anyone know the proper way to do this?

Thanks,
Mark

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Verificado por AVG - www.avgbrasil.com.br
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Lançamento: 10/08/10 15:34:00

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Re: [OSL | CCIE_Voice] UCCX Prompt

[Arun Kumar]

under SNU directory in UCCX folder.

On Fri, Oct 8, 2010 at 10:47 PM, Mark Holloway  wrote:

> Does anyone know if/what UCCX wav file says "Please try again later"
>
> Thanks,
> Mark
>
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>
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[OSL | CCIE_Voice] Vouchers for sale

[Mike Hurley]

CCIE #27139!   Was lucky enough to pass it on my first try!

 

I now have some vouchers left over...anyone looking for extra rack
time??   We can work something out via paypal.  

 

-Mike

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Re: [OSL | CCIE_Voice] Call Forward Unregistered

[Mark Holloway]

I have had it working before, but it's odd because sometimes when I reset the 
lab rack I can get it work and other times it does not work the way I want.  
I'm trying to figure out if I keep overlooking something.


On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote:

> I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking 
> CFUR) which is the Redirecting number. 
> 
> 
> On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote:
> 
>> Hi Mark,
>>  
>> The easiest way is to use calling party Transformation on the outbound 
>> gateway.
>>  
>> For example - 5002 calling 3002 out of local gateway. create a pt and assign 
>> it to a css. Assign css to the gateway "calling party transformation css" 
>> and uncheck use dp box. Now create a calling party transformation for 5XXX 
>> in the pt and modify the ANI to use extenal mask. 
>>  
>> This will modify the ANI from 5xxx to external mask everytime the 5xxx makes 
>> a call out of that gateway.
>>  
>> HTH
>> Prashant
>> 
>> On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway  wrote:
>> I'm trying to get my CFUR to work so it shows the External Mask in the For 
>> and By part of the call presentation but instead I am only getting it to 
>> show the 4 digit extension.  For example, lets say HQ 5001 calls BR1 3001 
>> (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because 
>> that site is in SRST mode).  The presentation on the BR1 phones is Forwarded 
>> HqPh1 5001, For 3001 By 3001.  Instead of 3001 I want to display the 
>> External Mask.  Does anyone know the proper way to do this?
>> 
>> Thanks,
>> Mark
>> 
>> ___
>> 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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Re: [OSL | CCIE_Voice] MVA and MGCP

[CCIE Voice GMAIL]

Bernhard,

 

It now just sunk in what you were getting at.  

 

I do have a GK configured with the ip address that would be the same as the
H323 GW (10.5.200.1).  

 

Does this mean I could potentially send the MVA communication through the GK
and then to CUCM?

 

Has anyone ever accomplished this?

 

Thanks,

Jeff

 

From: Stutz, Bernhard [mailto:st...@pandacom.de] 
Sent: Friday, October 08, 2010 12:58 PM
To: CCIE Voice GMAIL; ccie_voice@onlinestudylist.com
Subject: AW: [OSL | CCIE_Voice] MVA and MGCP

 

Maybe you have a gatekeeper with that ip address configured?

Your mgcp should be registered with the domain name something like
HQ-RTR.domain.com

 

Hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE Voice
GMAIL
Gesendet: Freitag, 8. Oktober 2010 20:29
An: ccie_voice@onlinestudylist.com
Betreff: [OSL | CCIE_Voice] MVA and MGCP

 

Hi everyone,

 

 

I am trying to figure out how I can configure an MGCP GW to work with MVA.
I have a two sites, HQ (MGCP) and SB (H323) that I am configuring to work
with MVA.  

 

I have successfully configure my SB router (H323) to allow MVA capability
with this configuration:

 

Service Parameters > CallManager > Clusterwide Parameters - Mobility

Enable Mobile Voice Access - True

Mobile Voice Access Number - 3777

 

Media Resouces > Mobile Voice Access 

MVA Directory Number - 3777

Partition -  

Locale - English United States

 

application 

service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml

 

voice translation-rule 1

rule 1 /^1312301\(3777$\)/ /\1/

 

voice translation-profile MVA

translate called 1

 

dial-peer voice 6 pots

description IN MVA

incoming called 13123013777

service MVA

translation-profile out MVA

 

dial-peer voice 7 voip

description TO SUB MVA

destination-pattern 3777

session target ipv4:172.21.51.205

preference 1

codec g711ulaw

no vad

dtmf-relay h245-signal

 

dial-peer voice 8 voip

description TO PUB MVA

destination-pattern 3777

session target ipv4:172.21.51.204

preference 2

codec g711ulaw

no vad

dtmf-relay h245-signal

 

I have also configured a Remote Destination Profile and Remote Destination
that is working completely.

 

I only have a single interface on the HQ gateway (MGCP) that can be used for
communications with CUCM as my lab is hooked up to a corporate network.
Every time I try to add the H323 GW for this router to the CUCM database I
get an error saying that a required field matches an existing database
entry.  I assume that this error is related to the fact that I have the HQ
GW registered as MGCP with the same interface's IP address.

 

Is there any way to get this working with MGCP?

 

Thanks for your help,

 

Jeff

 

 

 

 

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Re: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

[Tamer Ismail]

Hello Prashant,

It works… thanks.

 

Tamer,

 

From: Prashant Patel [mailto:prashantpatel...@gmail.com] 
Sent: Saturday, October 09, 2010 12:02 AM
To: Tamer Ismail
Cc: Stutz, Bernhard; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

 

Hi Tamer,

 

You can add any domain name. I usually add lab.com and fqdn as
cupspub.lab.com.  Also make sure you change the hostame to IP address is
Topology.

 

HTH

Prashant

On Fri, Oct 8, 2010 at 5:52 PM, Tamer Ismail  wrote:

Hello Stutz,
So which value I can put?
I don’t have any domain and I don't use DNS.
This value can't be IP Address.
Please advice.

Tamer,


-Original Message-
From: Stutz, Bernhard [mailto:st...@pandacom.de]
Sent: Friday, October 08, 2010 9:54 PM
To: Tamer Ismail; ccie_voice@onlinestudylist.com
Subject: AW: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

Hi Tamer,

You need to select the Cisco UP SIP Proxy Service at Presence Server Service
Settings.
There you can change Proxy Domain.

Cheers,
Bernhard

-Ursprüngliche Nachricht-
Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Tamer Ismail
Gesendet: Freitag, 8. Oktober 2010 18:22
An: ccie_voice@onlinestudylist.com
Betreff: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,

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Re: [OSL | CCIE_Voice] Call Forward Unregistered

[Mark Holloway]

I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking 
CFUR) which is the Redirecting number. 


On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote:

> Hi Mark,
>  
> The easiest way is to use calling party Transformation on the outbound 
> gateway.
>  
> For example - 5002 calling 3002 out of local gateway. create a pt and assign 
> it to a css. Assign css to the gateway "calling party transformation css" and 
> uncheck use dp box. Now create a calling party transformation for 5XXX in the 
> pt and modify the ANI to use extenal mask. 
>  
> This will modify the ANI from 5xxx to external mask everytime the 5xxx makes 
> a call out of that gateway.
>  
> HTH
> Prashant
> 
> On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway  wrote:
> I'm trying to get my CFUR to work so it shows the External Mask in the For 
> and By part of the call presentation but instead I am only getting it to show 
> the 4 digit extension.  For example, lets say HQ 5001 calls BR1 3001 (3001 is 
> unregistered and has CFUR set in CUCM to dial out the PSTN because that site 
> is in SRST mode).  The presentation on the BR1 phones is Forwarded HqPh1 
> 5001, For 3001 By 3001.  Instead of 3001 I want to display the External Mask. 
>  Does anyone know the proper way to do this?
> 
> Thanks,
> Mark
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 

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Re: [OSL | CCIE_Voice] Call Forward Unregistered

[Prashant Patel]

Hi Mark,

The easiest way is to use calling party Transformation on the outbound
gateway.

For example - 5002 calling 3002 out of local gateway. create a pt and assign
it to a css. Assign css to the gateway "calling party transformation css"
and uncheck use dp box. Now create a calling party transformation for 5XXX
in the pt and modify the ANI to use extenal mask.

This will modify the ANI from 5xxx to external mask everytime the 5xxx makes
a call out of that gateway.

HTH
Prashant

On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway  wrote:

> I'm trying to get my CFUR to work so it shows the External Mask in the For
> and By part of the call presentation but instead I am only getting it to
> show the 4 digit extension.  For example, lets say HQ 5001 calls BR1 3001
> (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because
> that site is in SRST mode).  The presentation on the BR1 phones is Forwarded
> HqPh1 5001, For 3001 By 3001.  Instead of 3001 I want to display the
> External Mask.  Does anyone know the proper way to do this?
>
> Thanks,
> Mark
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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[OSL | CCIE_Voice] CUE (unanswered questions)

[Pithog Oil]

I got CUE integrated with CUCM every thing was working except this , i really 
dont know if this is normal, after leaving a message for the cue user the 
caller hears a prompt wait while i transfer your calls and soon sees the call 
come back into callers phone.  THATS A LOOP

i get a similar issue when i integrate CUE with CME, when i try to leave a 
message for the cue user i get a prompt , record a message that is at least 2 
seconds long, even after making the message longer than 2 seconds i keep 
getting same prompt.
Experts thanks for the fix.

Please i would like to know how to fix this.
.
--- On Mon, 9/20/10, Amy Ryan  wrote:

From: Amy Ryan 
Subject: Re: [OSL | CCIE_Voice] After Hour block patterns
To: "Pithog Oil" , ccie_voice@onlinestudylist.com
Date: Monday, September 20, 2010, 8:59 PM





 
This would be the one to best bit the required solution.





after-hours day Sun 12:00 06:59

after-hours day Mon 19:00 06:59

after-hours day Tue 19:00 06:59

after-hours day Wed 19:00 06:59

after-hours day Thu 19:00 06:59

after-hours day Fri 19:00 06:59

after-hours day Sat 13:00 12:00







---

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, Wireless, 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: Pithog Oil 

Date: Mon, 20 Sep 2010 01:58:37 -0700 (PDT)

To: 

Subject: [OSL | CCIE_Voice] After Hour block patterns



Please i will like an expert to help me out with this solution



The Bls video conlict with work books, 



REQUIREMENT is to block international calls for this hours



Monday - Friday, 7pm-7am

All weekend, except Sat 7am-1pm



Please which of the solutions below best fits the required solution.



after-hours day Sun 12:00 07:00

after-hours day Mon 19:00 06:59

after-hours day Tue 19:00 06:59

after-hours day Wed 19:00 06:59

after-hours day Thu 19:00 06:59

after-hours day Fri 19:00 06:59

after-hours day Sat 13:00 12:00





after-hours day Sun 12:00 06:59

after-hours day Mon 19:00 06:59

after-hours day Tue 19:00 06:59

after-hours day Wed 19:00 06:59

after-hours day Thu 19:00 06:59

after-hours day Fri 19:00 06:59

after-hours day Sat 13:00 12:00







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[OSL | CCIE_Voice] I GOT INTO A LOOP INTEGRATING (CUE & CUCM)

[Pithog Oil]

I got CUE integrated with CUCM every thing was working except this , i really 
dont know if this is normal, after leaving a message for the cue user the 
caller hears a prompt wait while i transfer your calls and soon sees the call 
come into callers phone.  THATS A LOOP
Experts kindly assist to solve this issue.


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[OSL | CCIE_Voice] Call Forward Unregistered

[Mark Holloway]

I'm trying to get my CFUR to work so it shows the External Mask in the For and 
By part of the call presentation but instead I am only getting it to show the 4 
digit extension.  For example, lets say HQ 5001 calls BR1 3001 (3001 is 
unregistered and has CFUR set in CUCM to dial out the PSTN because that site is 
in SRST mode).  The presentation on the BR1 phones is Forwarded HqPh1 5001, For 
3001 By 3001.  Instead of 3001 I want to display the External Mask.  Does 
anyone know the proper way to do this?

Thanks,
Mark

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

Re: [OSL | CCIE_Voice] Unity Connection - Error when trying to record a customer greeting

[Prashant Patel]

Hi,

Is the screenshot from Users page ---> Greetings ---> record/play? I have
7.0.1.1-323 running and can record both in IE and FF from here calling
to a phone.

Thanks
Prashant

On Fri, Oct 8, 2010 at 6:28 PM, Miron Kobelski  wrote:

> Hi,
>
> I subscribe to this question... I've seen that, tried different browsers
> but couldn't find any solution. This is the quickest way to record a custom
> prompt for UCCX, so it would be a pain to see such issue on the exam...
>
> regards
> kobel
>
>  On Fri, Oct 8, 2010 at 20:56, Mark Holloway  wrote:
>
>> Has anyone ever seen this before?
>>
>>
>> I login to Unity Connection then click on my BR1PH1 user so I can record a
>> custom greeting.
>>
>>
>>
>>
>>
>>
>>
>> When I press the Record button I get the following error.
>>
>>
>> ___
>> 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] Unity Connection - Error when trying to record a customer greeting

[Miron Kobelski]

Hi,

I subscribe to this question... I've seen that, tried different browsers but
couldn't find any solution. This is the quickest way to record a custom
prompt for UCCX, so it would be a pain to see such issue on the exam...

regards
kobel

On Fri, Oct 8, 2010 at 20:56, Mark Holloway  wrote:

> Has anyone ever seen this before?
>
>
> I login to Unity Connection then click on my BR1PH1 user so I can record a
> custom greeting.
>
>
>
>
>
>
>
> When I press the Record button I get the following error.
>
>
> ___
> 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] Proxy Domain Parameter on CUPS

[Prashant Patel]

Hi Tamer,

You can add any domain name. I usually add lab.com and fqdn as
cupspub.lab.com.  Also make sure you change the hostame to IP address is
Topology.

HTH
Prashant

On Fri, Oct 8, 2010 at 5:52 PM, Tamer Ismail  wrote:

> Hello Stutz,
> So which value I can put?
> I don’t have any domain and I don't use DNS.
> This value can't be IP Address.
> Please advice.
>
> Tamer,
>
> -Original Message-
> From: Stutz, Bernhard [mailto:st...@pandacom.de]
> Sent: Friday, October 08, 2010 9:54 PM
> To: Tamer Ismail; ccie_voice@onlinestudylist.com
> Subject: AW: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS
>
> Hi Tamer,
>
> You need to select the Cisco UP SIP Proxy Service at Presence Server
> Service
> Settings.
> There you can change Proxy Domain.
>
> Cheers,
> Bernhard
>
> -Ursprüngliche Nachricht-
> Von: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Tamer
> Ismail
> Gesendet: Freitag, 8. Oktober 2010 18:22
> An: ccie_voice@onlinestudylist.com
> Betreff: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS
>
> Hello Experts,
> I found that I have to modify Proxy Domain parameter in Presence service
> parameter.
> And to set it as CUCM enterprise parameter.
> Actually I don't know the value of that parameter.
>
> Please advice.
>
> Tamer,
>
> ___
> 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] Proxy Domain Parameter on CUPS

[Tamer Ismail]

Hello Stutz,
So which value I can put?
I don’t have any domain and I don't use DNS.
This value can't be IP Address.
Please advice.

Tamer,

-Original Message-
From: Stutz, Bernhard [mailto:st...@pandacom.de] 
Sent: Friday, October 08, 2010 9:54 PM
To: Tamer Ismail; ccie_voice@onlinestudylist.com
Subject: AW: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

Hi Tamer,

You need to select the Cisco UP SIP Proxy Service at Presence Server Service
Settings.
There you can change Proxy Domain.

Cheers,
Bernhard

-Ursprüngliche Nachricht-
Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Tamer Ismail
Gesendet: Freitag, 8. Oktober 2010 18:22
An: ccie_voice@onlinestudylist.com
Betreff: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,

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

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

Re: [OSL | CCIE_Voice] How do you determine what the sccp version is

[Randall Crumm]

Hi Warren,
I don't see the sccp version when I do show run.

RC

-Original Message-
From: Warren Heaviside (wheavisi) [mailto:wheav...@cisco.com] 
Sent: Friday, October 08, 2010 2:26 PM
To: ccie_voice@onlinestudylist.com
Cc: Randall Crumm
Subject: RE: How do you determine what the sccp version is 

Hi Randall,

You can see the sccp version in the show run.  If you want to know the
DSP firmware version you can use "show voice dsp detail".

Warren

Warren Heavisidewheav...@cisco.com
ENGINEER.CUSTOMER SUPPORT
High Touch Technical Support
Phone: +1 408 853 7995

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


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Friday, October 08, 2010 11:57 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 56, Issue 93

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. How do you determine what the sccp version is (Randall Crumm)
   2. Proxy Domain Parameter on CUPS (Tamer Ismail)
   3. UCCX Prompt (Mark Holloway)
   4. MVA and MGCP (CCIE Voice GMAIL)
   5. Unity Connection - Error when trying to record a  customer
  greeting (Mark Holloway)


--

Message: 1
Date: Fri, 8 Oct 2010 09:13:27 -0700
From: Randall Crumm 
To: "ccie_voice@onlinestudylist.com" 
Subject: [OSL | CCIE_Voice] How do you determine what the sccp version
is
Message-ID:
<9473270a65ca67458d287f3da3c9f37d11a3368...@exch-cms.hlit.local>
Content-Type: text/plain; charset="us-ascii"

Hi is there a command to run to see the version of sccp on the router?

Thanks,
RC


--

Message: 2
Date: Fri, 8 Oct 2010 18:22:22 +0200
From: "Tamer Ismail" 
To: 
Subject: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS
Message-ID: <00a601cb6704$fddf48b0$f99dda...@com>
Content-Type: text/plain;   charset="us-ascii"

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,



--

Message: 3
Date: Fri, 8 Oct 2010 10:17:38 -0700
From: Mark Holloway 
To: CCIE Voice Maillist 
Subject: [OSL | CCIE_Voice] UCCX Prompt
Message-ID: <32e05610-e8c7-41fd-9400-8e2a8ced8...@markholloway.com>
Content-Type: text/plain; charset=us-ascii

Does anyone know if/what UCCX wav file says "Please try again later"

Thanks,
Mark



--

Message: 4
Date: Fri, 8 Oct 2010 11:29:28 -0700
From: "CCIE Voice GMAIL" 
To: 
Subject: [OSL | CCIE_Voice] MVA and MGCP
Message-ID: <000601cb6716$befe9d50$3cfbd7...@com>
Content-Type: text/plain; charset="us-ascii"

Hi everyone,

 

 

I am trying to figure out how I can configure an MGCP GW to work with
MVA.
I have a two sites, HQ (MGCP) and SB (H323) that I am configuring to
work
with MVA.  

 

I have successfully configure my SB router (H323) to allow MVA
capability
with this configuration:

 

Service Parameters > CallManager > Clusterwide Parameters - Mobility

Enable Mobile Voice Access - True

Mobile Voice Access Number - 3777

 

Media Resouces > Mobile Voice Access 

MVA Directory Number - 3777

Partition -  

Locale - English United States

 

application 

service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml

 

voice translation-rule 1

rule 1 /^1312301\(3777$\)/ /\1/

 

voice translation-profile MVA

translate called 1

 

dial-peer voice 6 pots

description IN MVA

incoming called 13123013777

service MVA

translation-profile out MVA

 

dial-peer voice 7 voip

description TO SUB MVA

destination-pattern 3777

session target ipv4:172.21.51.205

preference 1

codec g711ulaw

no vad

dtmf-relay h245-signal

 

dial-peer voice 8 voip

description TO PUB MVA

destination-pattern 3777

session target ipv4:172.21.51.204

preference 2

codec g711ulaw

no vad

dtmf-relay h245-signal

 

I have also configured a Remote Destination Profile and Remote
Destination
that is working completely.

 

I only have a single interface on the HQ gateway (MGCP) that can be used
for
communications with CUCM as my lab is hooked up to a corporate network.
Every time I try to add the H323 GW for this router to the CUCM database
I
get an error saying that a required field matches an existing databa

Re: [OSL | CCIE_Voice] MVA and MGCP

[CCIE]

I'm not using proctor labs. Thanks though. 

On Oct 8, 2010, at 12:57 PM, "Stutz, Bernhard"  wrote:

> Maybe you have a gatekeeper with that ip address configured?
> 
> Your mgcp should be registered with the domain name something like 
> HQ-RTR.domain.com
> 
>  
> 
> Hth,
> 
> Bernhard
> 
>  
> 
> Von: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE Voice 
> GMAIL
> Gesendet: Freitag, 8. Oktober 2010 20:29
> An: ccie_voice@onlinestudylist.com
> Betreff: [OSL | CCIE_Voice] MVA and MGCP
> 
>  
> 
> Hi everyone,
> 
>  
> 
>  
> 
> I am trying to figure out how I can configure an MGCP GW to work with MVA.  I 
> have a two sites, HQ (MGCP) and SB (H323) that I am configuring to work with 
> MVA. 
> 
>  
> 
> I have successfully configure my SB router (H323) to allow MVA capability 
> with this configuration:
> 
>  
> 
> Service Parameters > CallManager > Clusterwide Parameters – Mobility
> 
> Enable Mobile Voice Access – True
> 
> Mobile Voice Access Number - 3777
> 
>  
> 
> Media Resouces > Mobile Voice Access
> 
> MVA Directory Number – 3777
> 
> Partition - 
> 
> Locale – English United States
> 
>  
> 
> application
> 
> service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml
> 
>  
> 
> voice translation-rule 1
> 
> rule 1 /^1312301\(3777$\)/ /\1/
> 
>  
> 
> voice translation-profile MVA
> 
> translate called 1
> 
>  
> 
> dial-peer voice 6 pots
> 
> description IN MVA
> 
> incoming called 13123013777
> 
> service MVA
> 
> translation-profile out MVA
> 
>  
> 
> dial-peer voice 7 voip
> 
> description TO SUB MVA
> 
> destination-pattern 3777
> 
> session target ipv4:172.21.51.205
> 
> preference 1
> 
> codec g711ulaw
> 
> no vad
> 
> dtmf-relay h245-signal
> 
>  
> 
> dial-peer voice 8 voip
> 
> description TO PUB MVA
> 
> destination-pattern 3777
> 
> session target ipv4:172.21.51.204
> 
> preference 2
> 
> codec g711ulaw
> 
> no vad
> 
> dtmf-relay h245-signal
> 
>  
> 
> I have also configured a Remote Destination Profile and Remote Destination 
> that is working completely.
> 
>  
> 
> I only have a single interface on the HQ gateway (MGCP) that can be used for 
> communications with CUCM as my lab is hooked up to a corporate network.  
> Every time I try to add the H323 GW for this router to the CUCM database I 
> get an error saying that a required field matches an existing database entry. 
>  I assume that this error is related to the fact that I have the HQ GW 
> registered as MGCP with the same interface’s IP address.
> 
>  
> 
> Is there any way to get this working with MGCP?
> 
>  
> 
> Thanks for your help,
> 
>  
> 
> Jeff
> 
>  
> 
>  
> 
>  
> 
>  
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] How do you determine what the sccp version is

[Warren Heaviside (wheavisi)]

Hi Randall,

You can see the sccp version in the show run.  If you want to know the
DSP firmware version you can use "show voice dsp detail".

Warren

Warren Heavisidewheav...@cisco.com
ENGINEER.CUSTOMER SUPPORT
High Touch Technical Support
Phone: +1 408 853 7995

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


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Friday, October 08, 2010 11:57 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 56, Issue 93

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. How do you determine what the sccp version is (Randall Crumm)
   2. Proxy Domain Parameter on CUPS (Tamer Ismail)
   3. UCCX Prompt (Mark Holloway)
   4. MVA and MGCP (CCIE Voice GMAIL)
   5. Unity Connection - Error when trying to record a  customer
  greeting (Mark Holloway)


--

Message: 1
Date: Fri, 8 Oct 2010 09:13:27 -0700
From: Randall Crumm 
To: "ccie_voice@onlinestudylist.com" 
Subject: [OSL | CCIE_Voice] How do you determine what the sccp version
is
Message-ID:
<9473270a65ca67458d287f3da3c9f37d11a3368...@exch-cms.hlit.local>
Content-Type: text/plain; charset="us-ascii"

Hi is there a command to run to see the version of sccp on the router?

Thanks,
RC


--

Message: 2
Date: Fri, 8 Oct 2010 18:22:22 +0200
From: "Tamer Ismail" 
To: 
Subject: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS
Message-ID: <00a601cb6704$fddf48b0$f99dda...@com>
Content-Type: text/plain;   charset="us-ascii"

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,



--

Message: 3
Date: Fri, 8 Oct 2010 10:17:38 -0700
From: Mark Holloway 
To: CCIE Voice Maillist 
Subject: [OSL | CCIE_Voice] UCCX Prompt
Message-ID: <32e05610-e8c7-41fd-9400-8e2a8ced8...@markholloway.com>
Content-Type: text/plain; charset=us-ascii

Does anyone know if/what UCCX wav file says "Please try again later"

Thanks,
Mark



--

Message: 4
Date: Fri, 8 Oct 2010 11:29:28 -0700
From: "CCIE Voice GMAIL" 
To: 
Subject: [OSL | CCIE_Voice] MVA and MGCP
Message-ID: <000601cb6716$befe9d50$3cfbd7...@com>
Content-Type: text/plain; charset="us-ascii"

Hi everyone,

 

 

I am trying to figure out how I can configure an MGCP GW to work with
MVA.
I have a two sites, HQ (MGCP) and SB (H323) that I am configuring to
work
with MVA.  

 

I have successfully configure my SB router (H323) to allow MVA
capability
with this configuration:

 

Service Parameters > CallManager > Clusterwide Parameters - Mobility

Enable Mobile Voice Access - True

Mobile Voice Access Number - 3777

 

Media Resouces > Mobile Voice Access 

MVA Directory Number - 3777

Partition -  

Locale - English United States

 

application 

service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml

 

voice translation-rule 1

rule 1 /^1312301\(3777$\)/ /\1/

 

voice translation-profile MVA

translate called 1

 

dial-peer voice 6 pots

description IN MVA

incoming called 13123013777

service MVA

translation-profile out MVA

 

dial-peer voice 7 voip

description TO SUB MVA

destination-pattern 3777

session target ipv4:172.21.51.205

preference 1

codec g711ulaw

no vad

dtmf-relay h245-signal

 

dial-peer voice 8 voip

description TO PUB MVA

destination-pattern 3777

session target ipv4:172.21.51.204

preference 2

codec g711ulaw

no vad

dtmf-relay h245-signal

 

I have also configured a Remote Destination Profile and Remote
Destination
that is working completely.

 

I only have a single interface on the HQ gateway (MGCP) that can be used
for
communications with CUCM as my lab is hooked up to a corporate network.
Every time I try to add the H323 GW for this router to the CUCM database
I
get an error saying that a required field matches an existing database
entry.  I assume that this error is related to the fact that I have the
HQ
GW registered as MGCP with the same interface's IP address.

 

Is there any way to get this working with MGCP?

 

Thanks for your help,

 

Jeff

 

 

 

 

-- next part --
An HTML attachment was scrubbe

Re: [OSL | CCIE_Voice] UCCX Prompt

[Warren Heaviside (wheavisi)]

Hi Mark,

Do a search on *.wav and look for try-again.wav

Warren

Warren Heavisidewheav...@cisco.com
ENGINEER.CUSTOMER SUPPORT
High Touch Technical Support
Phone: +1 408 853 7995

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

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Friday, October 08, 2010 11:57 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 56, Issue 93

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. How do you determine what the sccp version is (Randall Crumm)
   2. Proxy Domain Parameter on CUPS (Tamer Ismail)
   3. UCCX Prompt (Mark Holloway)
   4. MVA and MGCP (CCIE Voice GMAIL)
   5. Unity Connection - Error when trying to record a  customer
  greeting (Mark Holloway)


--

Message: 1
Date: Fri, 8 Oct 2010 09:13:27 -0700
From: Randall Crumm 
To: "ccie_voice@onlinestudylist.com" 
Subject: [OSL | CCIE_Voice] How do you determine what the sccp version
is
Message-ID:
<9473270a65ca67458d287f3da3c9f37d11a3368...@exch-cms.hlit.local>
Content-Type: text/plain; charset="us-ascii"

Hi is there a command to run to see the version of sccp on the router?

Thanks,
RC


--

Message: 2
Date: Fri, 8 Oct 2010 18:22:22 +0200
From: "Tamer Ismail" 
To: 
Subject: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS
Message-ID: <00a601cb6704$fddf48b0$f99dda...@com>
Content-Type: text/plain;   charset="us-ascii"

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,



--

Message: 3
Date: Fri, 8 Oct 2010 10:17:38 -0700
From: Mark Holloway 
To: CCIE Voice Maillist 
Subject: [OSL | CCIE_Voice] UCCX Prompt
Message-ID: <32e05610-e8c7-41fd-9400-8e2a8ced8...@markholloway.com>
Content-Type: text/plain; charset=us-ascii

Does anyone know if/what UCCX wav file says "Please try again later"

Thanks,
Mark



--

Message: 4
Date: Fri, 8 Oct 2010 11:29:28 -0700
From: "CCIE Voice GMAIL" 
To: 
Subject: [OSL | CCIE_Voice] MVA and MGCP
Message-ID: <000601cb6716$befe9d50$3cfbd7...@com>
Content-Type: text/plain; charset="us-ascii"

Hi everyone,

 

 

I am trying to figure out how I can configure an MGCP GW to work with
MVA.
I have a two sites, HQ (MGCP) and SB (H323) that I am configuring to
work
with MVA.  

 

I have successfully configure my SB router (H323) to allow MVA
capability
with this configuration:

 

Service Parameters > CallManager > Clusterwide Parameters - Mobility

Enable Mobile Voice Access - True

Mobile Voice Access Number - 3777

 

Media Resouces > Mobile Voice Access 

MVA Directory Number - 3777

Partition -  

Locale - English United States

 

application 

service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml

 

voice translation-rule 1

rule 1 /^1312301\(3777$\)/ /\1/

 

voice translation-profile MVA

translate called 1

 

dial-peer voice 6 pots

description IN MVA

incoming called 13123013777

service MVA

translation-profile out MVA

 

dial-peer voice 7 voip

description TO SUB MVA

destination-pattern 3777

session target ipv4:172.21.51.205

preference 1

codec g711ulaw

no vad

dtmf-relay h245-signal

 

dial-peer voice 8 voip

description TO PUB MVA

destination-pattern 3777

session target ipv4:172.21.51.204

preference 2

codec g711ulaw

no vad

dtmf-relay h245-signal

 

I have also configured a Remote Destination Profile and Remote
Destination
that is working completely.

 

I only have a single interface on the HQ gateway (MGCP) that can be used
for
communications with CUCM as my lab is hooked up to a corporate network.
Every time I try to add the H323 GW for this router to the CUCM database
I
get an error saying that a required field matches an existing database
entry.  I assume that this error is related to the fact that I have the
HQ
GW registered as MGCP with the same interface's IP address.

 

Is there any way to get this working with MGCP?

 

Thanks for your help,

 

Jeff

 

 

 

 

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Message: 5
Date: Fri, 8 Oct 2010 11:56:37

Re: [OSL | CCIE_Voice] MVA and MGCP

[Stutz, Bernhard]

Maybe you have a gatekeeper with that ip address configured?

Your mgcp should be registered with the domain name something like
HQ-RTR.domain.com

 

Hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von CCIE
Voice GMAIL
Gesendet: Freitag, 8. Oktober 2010 20:29
An: ccie_voice@onlinestudylist.com
Betreff: [OSL | CCIE_Voice] MVA and MGCP

 

Hi everyone,

 

 

I am trying to figure out how I can configure an MGCP GW to work with
MVA.  I have a two sites, HQ (MGCP) and SB (H323) that I am configuring
to work with MVA.  

 

I have successfully configure my SB router (H323) to allow MVA
capability with this configuration:

 

Service Parameters > CallManager > Clusterwide Parameters - Mobility

Enable Mobile Voice Access - True

Mobile Voice Access Number - 3777

 

Media Resouces > Mobile Voice Access 

MVA Directory Number - 3777

Partition -  

Locale - English United States

 

application 

service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml

 

voice translation-rule 1

rule 1 /^1312301\(3777$\)/ /\1/

 

voice translation-profile MVA

translate called 1

 

dial-peer voice 6 pots

description IN MVA

incoming called 13123013777

service MVA

translation-profile out MVA

 

dial-peer voice 7 voip

description TO SUB MVA

destination-pattern 3777

session target ipv4:172.21.51.205

preference 1

codec g711ulaw

no vad

dtmf-relay h245-signal

 

dial-peer voice 8 voip

description TO PUB MVA

destination-pattern 3777

session target ipv4:172.21.51.204

preference 2

codec g711ulaw

no vad

dtmf-relay h245-signal

 

I have also configured a Remote Destination Profile and Remote
Destination that is working completely.

 

I only have a single interface on the HQ gateway (MGCP) that can be used
for communications with CUCM as my lab is hooked up to a corporate
network.  Every time I try to add the H323 GW for this router to the
CUCM database I get an error saying that a required field matches an
existing database entry.  I assume that this error is related to the
fact that I have the HQ GW registered as MGCP with the same interface's
IP address.

 

Is there any way to get this working with MGCP?

 

Thanks for your help,

 

Jeff

 

 

 

 

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Re: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

[Stutz, Bernhard]

Hi Tamer,

You need to select the Cisco UP SIP Proxy Service at Presence Server Service 
Settings.
There you can change Proxy Domain.

Cheers,
Bernhard

-Ursprüngliche Nachricht-
Von: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Tamer Ismail
Gesendet: Freitag, 8. Oktober 2010 18:22
An: ccie_voice@onlinestudylist.com
Betreff: [OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,

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___
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Re: [OSL | CCIE_Voice] Unity Connection - Error when trying to record a customer greeting

[ayman labib]

The only time I saw this message or similar is when I use my Firefox and not IE.








From: Mark Holloway 
To: CCIE Voice Maillist 
Sent: Fri, October 8, 2010 2:56:37 PM
Subject: [OSL | CCIE_Voice] Unity Connection - Error when trying to record a 
customer greeting

Has anyone ever seen this before?


I login to Unity Connection then click on my BR1PH1 user so I can record a 
custom greeting.






When I press the Record button I get the following error.

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[OSL | CCIE_Voice] Unity Connection - Error when trying to record a customer greeting

[Mark Holloway]

Has anyone ever seen this before?


I login to Unity Connection then click on my BR1PH1 user so I can record a 
custom greeting.


<>



When I press the Record button I get the following error.

<>___
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[OSL | CCIE_Voice] MVA and MGCP

[CCIE Voice GMAIL]

Hi everyone,

 

 

I am trying to figure out how I can configure an MGCP GW to work with MVA.
I have a two sites, HQ (MGCP) and SB (H323) that I am configuring to work
with MVA.  

 

I have successfully configure my SB router (H323) to allow MVA capability
with this configuration:

 

Service Parameters > CallManager > Clusterwide Parameters - Mobility

Enable Mobile Voice Access - True

Mobile Voice Access Number - 3777

 

Media Resouces > Mobile Voice Access 

MVA Directory Number - 3777

Partition -  

Locale - English United States

 

application 

service MVA http://172.21.51.204:8080/ccmivr/pages/IVRMainpage.vxml

 

voice translation-rule 1

rule 1 /^1312301\(3777$\)/ /\1/

 

voice translation-profile MVA

translate called 1

 

dial-peer voice 6 pots

description IN MVA

incoming called 13123013777

service MVA

translation-profile out MVA

 

dial-peer voice 7 voip

description TO SUB MVA

destination-pattern 3777

session target ipv4:172.21.51.205

preference 1

codec g711ulaw

no vad

dtmf-relay h245-signal

 

dial-peer voice 8 voip

description TO PUB MVA

destination-pattern 3777

session target ipv4:172.21.51.204

preference 2

codec g711ulaw

no vad

dtmf-relay h245-signal

 

I have also configured a Remote Destination Profile and Remote Destination
that is working completely.

 

I only have a single interface on the HQ gateway (MGCP) that can be used for
communications with CUCM as my lab is hooked up to a corporate network.
Every time I try to add the H323 GW for this router to the CUCM database I
get an error saying that a required field matches an existing database
entry.  I assume that this error is related to the fact that I have the HQ
GW registered as MGCP with the same interface's IP address.

 

Is there any way to get this working with MGCP?

 

Thanks for your help,

 

Jeff

 

 

 

 

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[OSL | CCIE_Voice] UCCX Prompt

[Mark Holloway]

Does anyone know if/what UCCX wav file says "Please try again later"

Thanks,
Mark

___
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[OSL | CCIE_Voice] Proxy Domain Parameter on CUPS

[Tamer Ismail]

Hello Experts,
I found that I have to modify Proxy Domain parameter in Presence service
parameter.
And to set it as CUCM enterprise parameter.
Actually I don't know the value of that parameter.

Please advice.

Tamer,

___
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[OSL | CCIE_Voice] How do you determine what the sccp version is

[Randall Crumm]

Hi is there a command to run to see the version of sccp on the router?

Thanks,
RC
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Re: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

[CCIE Voice GMAIL]

Thanks Goran and Ayman, that command did the trick.

 

Jeff

 

From: CCIE Voice GMAIL [mailto:givemeccievoice2...@gmail.com] 
Sent: Friday, October 08, 2010 7:45 AM
To: 'Goran Selthofer'
Cc: 'ccie_voice@onlinestudylist.com'
Subject: RE: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

 

Here is the dial-peer for incoming calls from the GK

 

dial-peer voice 8 voip

 description FROM GK

 translation-profile incoming FROM_GK

 session target ipv4:10.5.202.1

 incoming called-number 852

 dtmf-relay h245-signal

 no vad

 

I hardcode g729r8 as I normally write my dial-peers in notepad, however I
realize that by default the codec is g729r8 and that is why it is not listed
here.

 

I am not in the lab right now, so I will have to try that command "sdspfarm
transcode session x" under telephony and let you all know how it goes.

 

Jeff 

 

From: Goran Selthofer [mailto:seltho...@gmail.com] 
Sent: Friday, October 08, 2010 5:30 AM
To: CCIE Voice GMAIL
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

 

do you have a separate dial-peer voip for INCOMING calls from GK to CME ?

i don't see it posted here...

 

if you are getting calls from GK via specific incoming VOIP, then you should
NOT configure anything related to codecs on that incoming VOIP dial-peer
(i.e. no codec or no voice-class codec) and leave it to use by default g729.
That part and the fact that you have hardcoded g711u on dialpeer TO CUE will
trigger internal transcoder which you registeredd to telephony-service (of
course, as Ayman mentioned, you have to configure xcode sessions under
telephony-service - command is sdspfarm transcode sessions )

 

 

On Fri, Oct 8, 2010 at 6:24 AM, CCIE Voice GMAIL
 wrote:

Hey everyone,

 

I'm struggling to get calls to work from HQ or SB to SC through the GK to
CUE.  I'm figuring it's a problem with the Transcoder b/c the calls across
the WAN are G729 and CUE only accepts G711ulaw.

 

Any ideas what to do for this? 

 

Here is the relevant configurations:

 

 

 

<  VOICE SERVICE VOIP  >

 

voice service voip 

 allow-connections h323 to h323

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 no supplementary-service h225-notify cid-update

 fax protocol cisco 

 sip

  bind control source-interface Vlan250

  bind media source-interface Vlan250

  registrar server expires max 1500 min 300

 

 

<  DIAL PEER TO CUE  >

 

dial-peer voice 17 voip

 description TO CUE

 translation-profile outgoing TO_CUE

 destination-pattern 45[056][056]

 session protocol sipv2

 session target ipv4:10.5.202.254

 dtmf-relay sip-notify

 codec g711ulaw

 no vad

 

 

<  TRANSLATION TO CUE  >

 

voice translation-profile TO_CUE

 translate calling 14

 translate called 13

 

voice translation-rule 13

 rule 1 /^[234]...$/ /\0/

 

 

<  MEDIA RESOURCES  >

 

 

telephony-service

 sdspfarm units 3

 sdspfarm tag 1 SC_CONF

 sdspfarm tag 2 SC_MTP

 sdspfarm tag 3 SC_XCODE

 conference hardware

 

sccp local Vlan250

sccp ccm 10.5.202.1 identifier 1 version 7.0 

sccp

!

sccp ccm group 1

 associate ccm 1 priority 1

 associate profile 1 register SC_CONF

 associate profile 2 register SC_MTP

 associate profile 3 register SC_XCODE

!

dspfarm profile 3 transcode  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 maximum sessions 3

 associate application SCCP

!

dspfarm profile 1 conference  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 codec g729br8

 maximum sessions 1

 associate application SCCP

!

dspfarm profile 2 mtp  

 codec g729r8

 maximum sessions software 3

 associate application SCCP

 

<  EPHONE CONFIGURATION  >

 

ephone-dn  1  octo-line

 number 4001 no-reg primary

 label 4001

 description +442321314001

 name Site C Phone 1

 mobility

 snr 999 delay 2 timeout 10 cfwd-noan 4500

 allow watch

 call-forward busy 4500

 call-forward noan 4500 timeout 10

 

 

<  CUE SERVICE ENGINE   >

 

ip route 10.5.202.254 255.255.255.255 Service-Engine1/0

interface Service-Engine1/0

 ip unnumbered Vlan250

 service-module ip address 10.5.202.254 255.255.255.0

 service-module ip default-gateway 10.5.202.1

 

 

I was concerned that maybe the media resources didn't register, so I did a
"show sccp" command.  When doing this I found that the MTP wasn't
registering.  Does CME not support MTPs?  Or is it that I have a transcoder
registering as well?  I figure that the Transcoder should be able to handle
the calls from the GK to CUE as the transcoder is supporting both g711ulaw
and g729r8.

 

SCCP Admin State: UP

Gateway Local Interface: Vlan250

IPv4 Address: 10.5.202.1

Port Number: 2000

IP Precedence: 5

User Masked Codec list: No

Re: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

[CCIE Voice GMAIL]

Here is the dial-peer for incoming calls from the GK

 

dial-peer voice 8 voip

 description FROM GK

 translation-profile incoming FROM_GK

 session target ipv4:10.5.202.1

 incoming called-number 852

 dtmf-relay h245-signal

 no vad

 

I hardcode g729r8 as I normally write my dial-peers in notepad, however I
realize that by default the codec is g729r8 and that is why it is not listed
here.

 

I am not in the lab right now, so I will have to try that command "sdspfarm
transcode session x" under telephony and let you all know how it goes.

 

Jeff 

 

From: Goran Selthofer [mailto:seltho...@gmail.com] 
Sent: Friday, October 08, 2010 5:30 AM
To: CCIE Voice GMAIL
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

 

do you have a separate dial-peer voip for INCOMING calls from GK to CME ?

i don't see it posted here...

 

if you are getting calls from GK via specific incoming VOIP, then you should
NOT configure anything related to codecs on that incoming VOIP dial-peer
(i.e. no codec or no voice-class codec) and leave it to use by default g729.
That part and the fact that you have hardcoded g711u on dialpeer TO CUE will
trigger internal transcoder which you registeredd to telephony-service (of
course, as Ayman mentioned, you have to configure xcode sessions under
telephony-service - command is sdspfarm transcode sessions )

 

 

On Fri, Oct 8, 2010 at 6:24 AM, CCIE Voice GMAIL
 wrote:

Hey everyone,

 

I'm struggling to get calls to work from HQ or SB to SC through the GK to
CUE.  I'm figuring it's a problem with the Transcoder b/c the calls across
the WAN are G729 and CUE only accepts G711ulaw.

 

Any ideas what to do for this? 

 

Here is the relevant configurations:

 

 

 

<  VOICE SERVICE VOIP  >

 

voice service voip 

 allow-connections h323 to h323

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 no supplementary-service h225-notify cid-update

 fax protocol cisco 

 sip

  bind control source-interface Vlan250

  bind media source-interface Vlan250

  registrar server expires max 1500 min 300

 

 

<  DIAL PEER TO CUE  >

 

dial-peer voice 17 voip

 description TO CUE

 translation-profile outgoing TO_CUE

 destination-pattern 45[056][056]

 session protocol sipv2

 session target ipv4:10.5.202.254

 dtmf-relay sip-notify

 codec g711ulaw

 no vad

 

 

<  TRANSLATION TO CUE  >

 

voice translation-profile TO_CUE

 translate calling 14

 translate called 13

 

voice translation-rule 13

 rule 1 /^[234]...$/ /\0/

 

 

<  MEDIA RESOURCES  >

 

 

telephony-service

 sdspfarm units 3

 sdspfarm tag 1 SC_CONF

 sdspfarm tag 2 SC_MTP

 sdspfarm tag 3 SC_XCODE

 conference hardware

 

sccp local Vlan250

sccp ccm 10.5.202.1 identifier 1 version 7.0 

sccp

!

sccp ccm group 1

 associate ccm 1 priority 1

 associate profile 1 register SC_CONF

 associate profile 2 register SC_MTP

 associate profile 3 register SC_XCODE

!

dspfarm profile 3 transcode  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 maximum sessions 3

 associate application SCCP

!

dspfarm profile 1 conference  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 codec g729br8

 maximum sessions 1

 associate application SCCP

!

dspfarm profile 2 mtp  

 codec g729r8

 maximum sessions software 3

 associate application SCCP

 

<  EPHONE CONFIGURATION  >

 

ephone-dn  1  octo-line

 number 4001 no-reg primary

 label 4001

 description +442321314001

 name Site C Phone 1

 mobility

 snr 999 delay 2 timeout 10 cfwd-noan 4500

 allow watch

 call-forward busy 4500

 call-forward noan 4500 timeout 10

 

 

<  CUE SERVICE ENGINE   >

 

ip route 10.5.202.254 255.255.255.255 Service-Engine1/0

interface Service-Engine1/0

 ip unnumbered Vlan250

 service-module ip address 10.5.202.254 255.255.255.0

 service-module ip default-gateway 10.5.202.1

 

 

I was concerned that maybe the media resources didn't register, so I did a
"show sccp" command.  When doing this I found that the MTP wasn't
registering.  Does CME not support MTPs?  Or is it that I have a transcoder
registering as well?  I figure that the Transcoder should be able to handle
the calls from the GK to CUE as the transcoder is supporting both g711ulaw
and g729r8.

 

SCCP Admin State: UP

Gateway Local Interface: Vlan250

IPv4 Address: 10.5.202.1

Port Number: 2000

IP Precedence: 5

User Masked Codec list: None

Call Manager: 10.5.202.1, Port Number: 2000

Priority: N/A, Version: 7.0, Identifier: 1

Trustpoint: N/A

 

Conferencing Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.5.202.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Iden

Re: [OSL | CCIE_Voice] CUCM - UCCS upgrading options

[Shadab Abbasi]

Guys,

This is something pre-sales business oppurtunity query. I dont think this
forum is a appropriate place to discuss such thing.

Saeed - Please engage your local cisco SE to find out more options.


Cheers,
Shadab

On Fri, Oct 8, 2010 at 7:04 PM, Matthew Saskin  wrote:

> If UCSS wasn't purchased initially or within 90 days (if I recall
> correctly) of the initial purchase, they can't purchase UCSS to enable the
> upgrade.  They need to re-purchase 8.0 software + licenses.
>
> Matthew Saskin
> msas...@gmail.com
> 203-253-9571
>
> July 18, 2010 - 1500m swim (in the hudson), 40k bike, 10k run
> Please support the Leukemia & Lyphoma Society
> http://pages.teamintraining.org/nyc/nyctri10/msaskin
>
>
>   On Wed, Oct 6, 2010 at 4:48 PM, Saeed IDris  wrote:
>
>>*Hi everyone,*
>>
>> *I need to ask regarding Cisco UCCS, is it enough to include the below
>> part number according to number of users in case this is new implementation:
>> *
>>
>> *Other question, one of my clients already is hosted CUCMBE version 6.5.1
>> but now they plan to upgrade their CUCM to version 8.0 …..They didn’t have
>> UCCS ….what is the possible Scenario to achieve such upgrade?*
>>
>> **
>>
>> *Regards,*
>>
>> *Saeed *
>>
>> ___
>> 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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[OSL | CCIE_Voice] (no subject)

[bmaxim]

http://www.divittoeditore.com/und9.html


  
___
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Re: [OSL | CCIE_Voice] CUCM - UCCS upgrading options

[Matthew Saskin]

If UCSS wasn't purchased initially or within 90 days (if I recall correctly)
of the initial purchase, they can't purchase UCSS to enable the upgrade.
They need to re-purchase 8.0 software + licenses.

Matthew Saskin
msas...@gmail.com
203-253-9571

July 18, 2010 - 1500m swim (in the hudson), 40k bike, 10k run
Please support the Leukemia & Lyphoma Society
http://pages.teamintraining.org/nyc/nyctri10/msaskin


On Wed, Oct 6, 2010 at 4:48 PM, Saeed IDris  wrote:

> *Hi everyone,*
>
> *I need to ask regarding Cisco UCCS, is it enough to include the below
> part number according to number of users in case this is new implementation:
> *
>
> *Other question, one of my clients already is hosted CUCMBE version 6.5.1
> but now they plan to upgrade their CUCM to version 8.0 …..They didn’t have
> UCCS ….what is the possible Scenario to achieve such upgrade?*
>
> **
>
> *Regards,*
>
> *Saeed *
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

[Goran Selthofer]

do you have a separate dial-peer voip for INCOMING calls from GK to CME ?
i don't see it posted here...

if you are getting calls from GK via specific incoming VOIP, then you should
NOT configure anything related to codecs on that incoming VOIP dial-peer
(i.e. no codec or no voice-class codec) and leave it to use by default g729.
That part and the fact that you have hardcoded g711u on dialpeer TO CUE will
trigger internal transcoder which you registeredd to telephony-service (of
course, as Ayman mentioned, you have to configure xcode sessions under
telephony-service - command is sdspfarm transcode sessions )


On Fri, Oct 8, 2010 at 6:24 AM, CCIE Voice GMAIL <
givemeccievoice2...@gmail.com> wrote:

>  Hey everyone,
>
>
>
> I’m struggling to get calls to work from HQ or SB to SC through the GK to
> CUE.  I’m figuring it’s a problem with the Transcoder b/c the calls across
> the WAN are G729 and CUE only accepts G711ulaw.
>
>
>
> Any ideas what to do for this?
>
>
>
> Here is the relevant configurations:
>
>
>
>
>
>
>
> <  VOICE SERVICE VOIP  >
>
>
>
> voice service voip
>
>  allow-connections h323 to h323
>
>  allow-connections h323 to sip
>
>  allow-connections sip to h323
>
>  allow-connections sip to sip
>
>  no supplementary-service h225-notify cid-update
>
>  fax protocol cisco
>
>  sip
>
>   bind control source-interface Vlan250
>
>   bind media source-interface Vlan250
>
>   registrar server expires max 1500 min 300
>
>
>
>
>
> <  DIAL PEER TO CUE  >
>
>
>
> dial-peer voice 17 voip
>
>  description TO CUE
>
>  translation-profile outgoing TO_CUE
>
>  destination-pattern 45[056][056]
>
>  session protocol sipv2
>
>  session target ipv4:10.5.202.254
>
>  dtmf-relay sip-notify
>
>  codec g711ulaw
>
>  no vad
>
>
>
>
>
> <  TRANSLATION TO CUE  >
>
>
>
> voice translation-profile TO_CUE
>
>  translate calling 14
>
>  translate called 13
>
>
>
> voice translation-rule 13
>
>  rule 1 /^[234]...$/ /\0/
>
>
>
>
>
> <  MEDIA RESOURCES  >
>
>
>
>
>
> telephony-service
>
>  sdspfarm units 3
>
>  sdspfarm tag 1 SC_CONF
>
>  sdspfarm tag 2 SC_MTP
>
>  sdspfarm tag 3 SC_XCODE
>
>  conference hardware
>
>
>
> sccp local Vlan250
>
> sccp ccm 10.5.202.1 identifier 1 version 7.0
>
> sccp
>
> !
>
> sccp ccm group 1
>
>  associate ccm 1 priority 1
>
>  associate profile 1 register SC_CONF
>
>  associate profile 2 register SC_MTP
>
>  associate profile 3 register SC_XCODE
>
> !
>
> dspfarm profile 3 transcode
>
>  codec g711ulaw
>
>  codec g711alaw
>
>  codec g729ar8
>
>  codec g729abr8
>
>  codec g729r8
>
>  maximum sessions 3
>
>  associate application SCCP
>
> !
>
> dspfarm profile 1 conference
>
>  codec g711ulaw
>
>  codec g711alaw
>
>  codec g729ar8
>
>  codec g729abr8
>
>  codec g729r8
>
>  codec g729br8
>
>  maximum sessions 1
>
>  associate application SCCP
>
> !
>
> dspfarm profile 2 mtp
>
>  codec g729r8
>
>  maximum sessions software 3
>
>  associate application SCCP
>
>
>
> <  EPHONE CONFIGURATION  >
>
>
>
> ephone-dn  1  octo-line
>
>  number 4001 no-reg primary
>
>  label 4001
>
>  description +442321314001
>
>  name Site C Phone 1
>
>  mobility
>
>  snr 999 delay 2 timeout 10 cfwd-noan 4500
>
>  allow watch
>
>  call-forward busy 4500
>
>  call-forward noan 4500 timeout 10
>
>
>
>
>
> <  CUE SERVICE ENGINE   >
>
>
>
> ip route 10.5.202.254 255.255.255.255 Service-Engine1/0
>
> interface Service-Engine1/0
>
>  ip unnumbered Vlan250
>
>  service-module ip address 10.5.202.254 255.255.255.0
>
>  service-module ip default-gateway 10.5.202.1
>
>
>
>
>
> I was concerned that maybe the media resources didn’t register, so I did a
> “show sccp” command.  When doing this I found that the MTP wasn’t
> registering.  Does CME not support MTPs?  Or is it that I have a transcoder
> registering as well?  I figure that the Transcoder should be able to handle
> the calls from the GK to CUE as the transcoder is supporting both g711ulaw
> and g729r8.
>
>
>
> SCCP Admin State: UP
>
> Gateway Local Interface: Vlan250
>
> IPv4 Address: 10.5.202.1
>
> Port Number: 2000
>
> IP Precedence: 5
>
> User Masked Codec list: None
>
> Call Manager: 10.5.202.1, Port Number: 2000
>
> Priority: N/A, Version: 7.0, Identifier: 1
>
> Trustpoint: N/A
>
>
>
> Conferencing Oper State: ACTIVE - Cause Code: NONE
>
> Active Call Manager: 10.5.202.1, Port Number: 2000
>
> TCP Link Status: CONNECTED, Profile Identifier: 1
>
> Reported Max Streams: 8, Reported Max OOS Streams: 0
>
> Supported Codec: g711ulaw, Maximum Packetization Period: 30
>
> Supported Codec: g711alaw, Maximum Packetization Period: 30
>
> Supported Codec: g729ar8, Maximum Packetization Period: 60
>
> Supported Codec: g729abr8, Maximum Packetization Period: 60
>
> Supported Codec: g729r8, Maximum Packetization Period: 60
>
> Supported Codec: g729br8

Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

[Stutz, Bernhard]

If I see this right you are trying to call 5621891 which is the Iceland
local number at BR2 right?

So you are doing a TEHO from US to the Iceland Gateway

Not sure but has the Iceland Breakout not the requirement to send this
with called party type subscriber? And calling party type international?

 

 

Hth,

Bernhard

 

Von: Graham Hopkins [mailto:ghopk...@wolf-rock.co.uk] 
Gesendet: Freitag, 8. Oktober 2010 10:47
An: amr thabt
Cc: Stutz, Bernhard; ccie_voice@onlinestudylist.com; Pithog Oil
Betreff: Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

 

Currently have a similar issue with the same lab -  symptoms are:

 

MVA call connects OK and calls placed to internal numbers are fine (
except 5002 but that is the number that the mobile is linked to so may
be normal - why would you call yourself)

Calls placed to local/ld numbers never reach the HQ MGCP gateway

Calls placed to international numbers at BR2 reach the BR2 UCME and then
hang up after one ring - Cause i = 0x80AF - Resource unavailable,
unspecified

 

 Time for some CUCM debugs - any other ideas ?

 

Bits from config  and debug

 

HQ RTR

 

voice translation-rule 100

 rule 1 /^5002$/ /2123942123/

!

voice translation-profile MVA

 translate calling 100   

 

application

 service MVA http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml   

 

dial-peer voice 5010 voip

 translation-profile incoming MVA

 service mva

 destination-pattern 5010

 session target ipv4:10.10.210.10

 incoming called-number 5010

 dtmf-relay h245-alphanumeric

 codec g711ulaw

 no vad  

 

CUCM

 

5010 in pt-internal is matched by the MGCP gateway and points to to
H.323 Gateway which points to 5010 in pt-mva which is the MVA access
number

 

FROM Br2 RTR

 

Oct  8 08:39:17.932: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type
0x12 is 0x0 0x1, Calling num 12123945002

Oct  8 08:39:17.932: ISDN Se0/0/0:15 Q931: Sending SETUP  callref =
0x0082 callID = 0x8003 switch = primary-net5 interface = User

Oct  8 08:39:17.936: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref =
0x0082

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98383

Exclusive, Channel 3

Calling Party Number i = 0x0181, '12123945002'

Plan:ISDN, Type:Unknown

Called Party Number i = 0x80, '5621891'

Plan:Unknown, Type:Unknown

BR2-RTR#

Oct  8 08:39:17.972: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8
callref = 0x8082

Channel ID i = 0xA98383

Exclusive, Channel 3

Oct  8 08:39:17.992: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8
callref = 0x8082

Progress Ind i = 0x8188 - In-band info or appropriate now
available

Oct  8 08:39:18.084: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0082

Cause i = 0x80AF - Resource unavailable, unspecified

Oct  8 08:39:18.096: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref
= 0x8082

Oct  8 08:39:18.100: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x0082

BR2-RTR#

 


 

 

 

 

 

Regards

 

Graham 

 

 

 

On 7 Oct 2010, at 20:00, amr thabt wrote:





Hi Stutz,

 1- add translation rule& profile to dial-p 1997 to change the calling
number to be  '8884343' .

 2- if still have a problem , check css of RDP and may restart Mobile
Voice Service

 I hpoe this may help

HTH

AMR

 

On Thu, Oct 7, 2010 at 9:26 PM, Stutz, Bernhard 
wrote:



Hi,

 

I run into the same issue.

furthermore i have to hairpin the call through a h323 gateway as all
incoming calls come per mgcp to the callmanager. You have then to add a
H.323 gateway to the same mgcp gateway which is possible.

 

I got following dial--peers configured:

 

dial-peer voice 1999 voip
 service cmm
 incoming called-number 1999
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 101 voip
 preference 1
 destination-pattern 1997
 voice-class h323 1
 session target ipv4:10.10.210.10
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

Under callmanager i have 1997 as MVA Number defined at Media
Ressources->Mobile Voice Access and also at service parameter

 

When i call the mva the call comes in via mgcp, on ccm i have a route
pattern that sends 1999 back to the h.323 configured gateway, then the
service gets invoked. so far so good.

 

I have remote destination configured with 8884343 and the call comes in
as following:

 

Oct  7 21:41:58.277: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref =
0x00B4
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Progress Ind i = 0x8583 - Origination address is non-ISD

Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

[Ayman_labib]

I will tear everything down and start from scratch again.  I will report back 
this morning.

Sent from my iPhone

On Oct 8, 2010, at 2:06 AM, Mark Holloway  wrote:

> Hmm, PSTN to BR1 and IP to IP (inter and intra site) play multicast MoH piano 
> music from route flash just fine, but for some reason when calling from BR1 
> to the PSTN and pressing HOLD on the BR1 phone it plays beep beep beep.  
> 
> Usually the issue is PSTN to IP because you need a voice class codec on the 
> SUB/PUB dial peers that support G711, which I have, and PSTN to BR1 piano 
> music streams multicast ok. Not sure what would cause IP to PSTN calls to 
> fail streaming MoH and play beep beep beep.  Any ideas?
> 
> 
> 
> On Oct 7, 2010, at 1:36 PM, ayman labib wrote:
> 
>> Thanks for the reply. 
>> 
>> As it turns out.  Loopback interface is a required step.  Now everything is 
>> working.  Thanks
>> 
>> Next challenge is to get Site HQ and SRST to use MoH with CME using the 
>> Gatekeeper.  Thanks
>> 
>> From: ayman labib 
>> To: amr thabt 
>> Cc: ccie_voice@onlinestudylist.com
>> Sent: Thu, October 7, 2010 3:49:41 PM
>> Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site
>> 
>> Thanks for the reply.
>> 
>> I do have the max ephone etc..  I removed my config to keep it short.
>> I tried it with bind command and without.  Same Issue.
>> I don't have Lo0 configured.  Everything is configured using the fa0/1 
>> interface.  
>> 
>> Please have a look at the screen shots of my config.  I really appreciate 
>> everyone's help.  2 days and it's driving me crazy. 
>> 
>> call-manager-fallback
>>  secondary-dialtone 9
>>  max-conferences 8 gain -6
>>  transfer-system full-consult
>>  ip source-address 192.168.31.10 port 2000 strict-match
>>  max-ephones 10
>>  max-dn 10
>>  transfer-pattern .T
>>  voicemail 912123945020
>>  call-forward pattern .T
>>  call-forward busy 12123945020
>>  call-forward noan 12123945020 timeout 20
>>  moh music-on-hold.au
>>  multicast moh 239.1.1.1 port 16384 route 192.168.31.10
>>  time-zone 8
>> !
>> 
>> 
>> From: amr thabt 
>> To: ayman labib 
>> Cc: ccie_voice@onlinestudylist.com
>> Sent: Thu, October 7, 2010 3:07:59 PM
>> Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site
>> 
>> Hi Ayman,
>> I have three comments that may help
>> 1 Do you add max-dn and max-ephone under call-manager-fallback
>> 2-in "ccm-manager music-on-hold bind fa0/1 " remove the bind use only 
>> ccm-manager music-on-hold
>> 3- in multicast command add both loopback and VLan SVI ip address.
>>  
>>  
>> HTH
>> AMR
>> 
>> 
>> On Thu, Oct 7, 2010 at 9:56 PM, ayman labib  wrote:
>> Just wondering if anyone encountered this problem.
>> 
>> I still can't get MOH when calling the PSTN phone and the site is not in 
>> SRST mode.  According to the sh command below.  The call manager has done 
>> its job  but the GWY is not responding.  Any ideas?  MOH local and between 
>> HQ works fine.  Just need a sanity check.  Thanks for all your help
>> 
>> SRST-Site#sh ccm-manager music-on-hold
>> Current active multicast sessions : 1
>>  Multicast   RTP port   Packets   Call   CodecIncoming
>>  Address number in/outidInterface
>> ===
>> 239.1.1.1 16384   0/0  12   g711ulaw
>> 
>> ccm-manager music-on-hold bind fa0/1
>> 
>> call-manager-fallback
>>  ip source-address 192.168.31.10 port 2000 strict-match
>>  moh music-on-hold.au
>>  multicast moh 239.1.1.1 port 16384 route 192.168.31.10
>>  
>> 
>> http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789
>> 
>> From: ayman labib 
>> To: ccie_voice@onlinestudylist.com
>> Cc: ccie_voice@onlinestudylist.com
>> Sent: Wed, October 6, 2010 9:45:12 AM
>> Subject: MoH to PSTN from SRST site
>> 
>> 
>> Hello Experts,
>> 
>> Follow up to Mark's email about Moh to PSTN.  I don't hear the Piano music 
>> as well.  Inter-site and Intra-site with HQ works.  
>> 
>> I see the Muticast on the gateway is invoked and on the server, but don't 
>> hear anything.  Any idea?  Thanks in advance
>> 
>> admin:show perf query class "Cisco MOH Device"
>> ==>query class :
>> 
>>  - Perf class (Cisco MOH Device) has instances and values:
>> MOH_2   -> MOHHighestActiveResources  = 1
>> MOH_2   -> MOHMulticastResourceActive = 0
>> MOH_2   -> MOHMulticastResourceAvailable  = 25
>> MOH_2   -> MOHOutOfResources  = 0
>> MOH_2   -> MOHTotalMulticastResources = 25
>> MOH_2   -> MOHTotalUnicastResources   = 250
>> MOH_2   -> MOHUnicastResourceActive   = 0
>> MOH_2   -> MOHUnicastResourceAvailable= 250
>> MOH_3   -> MOHHighestActiveResources  = 1
>> MOH_3   -> MOHMulticastResourceActive = 1
>> MOH_3   -> MOHMulticastResourceAvailable  = 24
>> MOH_3

Re: [OSL | CCIE_Voice] Forwarding GK Calls to CUE

[Ayman_labib]

I think you're missing one command under telephony- service
Sdspfarm transcode session 4 
Or something similar to this.  Too early in the morning to remember

Sent from my iPhone

On Oct 8, 2010, at 12:24 AM, "CCIE Voice GMAIL"  
wrote:

> Hey everyone,
> 
>  
> 
> I’m struggling to get calls to work from HQ or SB to SC through the GK to 
> CUE.  I’m figuring it’s a problem with the Transcoder b/c the calls across 
> the WAN are G729 and CUE only accepts G711ulaw.
> 
>  
> 
> Any ideas what to do for this?
> 
>  
> 
> Here is the relevant configurations:
> 
>  
> 
>  
> 
>  
> 
> <  VOICE SERVICE VOIP  >
> 
>  
> 
> voice service voip
> 
>  allow-connections h323 to h323
> 
>  allow-connections h323 to sip
> 
>  allow-connections sip to h323
> 
>  allow-connections sip to sip
> 
>  no supplementary-service h225-notify cid-update
> 
>  fax protocol cisco
> 
>  sip
> 
>   bind control source-interface Vlan250
> 
>   bind media source-interface Vlan250
> 
>   registrar server expires max 1500 min 300
> 
>  
> 
>  
> 
> <  DIAL PEER TO CUE  >
> 
>  
> 
> dial-peer voice 17 voip
> 
>  description TO CUE
> 
>  translation-profile outgoing TO_CUE
> 
>  destination-pattern 45[056][056]
> 
>  session protocol sipv2
> 
>  session target ipv4:10.5.202.254
> 
>  dtmf-relay sip-notify
> 
>  codec g711ulaw
> 
>  no vad
> 
>  
> 
>  
> 
> <  TRANSLATION TO CUE  >
> 
>  
> 
> voice translation-profile TO_CUE
> 
>  translate calling 14
> 
>  translate called 13
> 
>  
> 
> voice translation-rule 13
> 
>  rule 1 /^[234]...$/ /\0/
> 
>  
> 
>  
> 
> <  MEDIA RESOURCES  >
> 
>  
> 
>  
> 
> telephony-service
> 
>  sdspfarm units 3
> 
>  sdspfarm tag 1 SC_CONF
> 
>  sdspfarm tag 2 SC_MTP
> 
>  sdspfarm tag 3 SC_XCODE
> 
>  conference hardware
> 
>  
> 
> sccp local Vlan250
> 
> sccp ccm 10.5.202.1 identifier 1 version 7.0 
> 
> sccp
> 
> !
> 
> sccp ccm group 1
> 
>  associate ccm 1 priority 1
> 
>  associate profile 1 register SC_CONF
> 
>  associate profile 2 register SC_MTP
> 
>  associate profile 3 register SC_XCODE
> 
> !
> 
> dspfarm profile 3 transcode 
> 
>  codec g711ulaw
> 
>  codec g711alaw
> 
>  codec g729ar8
> 
>  codec g729abr8
> 
>  codec g729r8
> 
>  maximum sessions 3
> 
>  associate application SCCP
> 
> !
> 
> dspfarm profile 1 conference 
> 
>  codec g711ulaw
> 
>  codec g711alaw
> 
>  codec g729ar8
> 
>  codec g729abr8
> 
>  codec g729r8
> 
>  codec g729br8
> 
>  maximum sessions 1
> 
>  associate application SCCP
> 
> !
> 
> dspfarm profile 2 mtp 
> 
>  codec g729r8
> 
>  maximum sessions software 3
> 
>  associate application SCCP
> 
>  
> 
> <  EPHONE CONFIGURATION  >
> 
>  
> 
> ephone-dn  1  octo-line
> 
>  number 4001 no-reg primary
> 
>  label 4001
> 
>  description +442321314001
> 
>  name Site C Phone 1
> 
>  mobility
> 
>  snr 999 delay 2 timeout 10 cfwd-noan 4500
> 
>  allow watch
> 
>  call-forward busy 4500
> 
>  call-forward noan 4500 timeout 10
> 
>  
> 
>  
> 
> <  CUE SERVICE ENGINE   >
> 
>  
> 
> ip route 10.5.202.254 255.255.255.255 Service-Engine1/0
> 
> interface Service-Engine1/0
> 
>  ip unnumbered Vlan250
> 
>  service-module ip address 10.5.202.254 255.255.255.0
> 
>  service-module ip default-gateway 10.5.202.1
> 
>  
> 
>  
> 
> I was concerned that maybe the media resources didn’t register, so I did a 
> “show sccp” command.  When doing this I found that the MTP wasn’t 
> registering.  Does CME not support MTPs?  Or is it that I have a transcoder 
> registering as well?  I figure that the Transcoder should be able to handle 
> the calls from the GK to CUE as the transcoder is supporting both g711ulaw 
> and g729r8.
> 
>  
> 
> SCCP Admin State: UP
> 
> Gateway Local Interface: Vlan250
> 
> IPv4 Address: 10.5.202.1
> 
> Port Number: 2000
> 
> IP Precedence: 5
> 
> User Masked Codec list: None
> 
> Call Manager: 10.5.202.1, Port Number: 2000
> 
> Priority: N/A, Version: 7.0, Identifier: 1
> 
> Trustpoint: N/A
> 
>  
> 
> Conferencing Oper State: ACTIVE - Cause Code: NONE
> 
> Active Call Manager: 10.5.202.1, Port Number: 2000
> 
> TCP Link Status: CONNECTED, Profile Identifier: 1
> 
> Reported Max Streams: 8, Reported Max OOS Streams: 0
> 
> Supported Codec: g711ulaw, Maximum Packetization Period: 30
> 
> Supported Codec: g711alaw, Maximum Packetization Period: 30
> 
> Supported Codec: g729ar8, Maximum Packetization Period: 60
> 
> Supported Codec: g729abr8, Maximum Packetization Period: 60
> 
> Supported Codec: g729r8, Maximum Packetization Period: 60
> 
> Supported Codec: g729br8, Maximum Packetization Period: 60
> 
> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
> 
> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
> 
> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization 
> Period: 

Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

[Graham Hopkins]

Currently have a similar issue with the same lab -  symptoms are:

MVA call connects OK and calls placed to internal numbers are fine ( except 
5002 but that is the number that the mobile is linked to so may be normal - why 
would you call yourself)
Calls placed to local/ld numbers never reach the HQ MGCP gateway
Calls placed to international numbers at BR2 reach the BR2 UCME and then hang 
up after one ring - Cause i = 0x80AF - Resource unavailable, unspecified

 Time for some CUCM debugs - any other ideas ?

Bits from config  and debug

HQ RTR

voice translation-rule 100
 rule 1 /^5002$/ /2123942123/
!
voice translation-profile MVA
 translate calling 100   

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

dial-peer voice 5010 voip
 translation-profile incoming MVA
 service mva
 destination-pattern 5010
 session target ipv4:10.10.210.10
 incoming called-number 5010
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad  

CUCM

5010 in pt-internal is matched by the MGCP gateway and points to to H.323 
Gateway which points to 5010 in pt-mva which is the MVA access number

FROM Br2 RTR

Oct  8 08:39:17.932: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 
is 0x0 0x1, Calling num 12123945002
Oct  8 08:39:17.932: ISDN Se0/0/0:15 Q931: Sending SETUP  callref = 0x0082 
callID = 0x8003 switch = primary-net5 interface = User
Oct  8 08:39:17.936: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0082
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98383
Exclusive, Channel 3
Calling Party Number i = 0x0181, '12123945002'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x80, '5621891'
Plan:Unknown, Type:Unknown
BR2-RTR#
Oct  8 08:39:17.972: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8  callref = 
0x8082
Channel ID i = 0xA98383
Exclusive, Channel 3
Oct  8 08:39:17.992: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8  callref = 
0x8082
Progress Ind i = 0x8188 - In-band info or appropriate now available
Oct  8 08:39:18.084: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 
0x0082
Cause i = 0x80AF - Resource unavailable, unspecified
Oct  8 08:39:18.096: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref = 
0x8082
Oct  8 08:39:18.100: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 
0x0082
BR2-RTR#

  





Regards

Graham 



On 7 Oct 2010, at 20:00, amr thabt wrote:

> Hi Stutz,
>  1- add translation rule& profile to dial-p 1997 to change the calling number 
> to be  '8884343' .
>  2- if still have a problem , check css of RDP and may restart Mobile Voice 
> Service
>  I hpoe this may help
> HTH
> AMR
> 
> 
> On Thu, Oct 7, 2010 at 9:26 PM, Stutz, Bernhard  wrote:
> Hi,
>  
> I run into the same issue.
> furthermore i have to hairpin the call through a h323 gateway as all incoming 
> calls come per mgcp to the callmanager. You have then to add a H.323 gateway 
> to the same mgcp gateway which is possible.
>  
> I got following dial--peers configured:
>  
> dial-peer voice 1999 voip
>  service cmm
>  incoming called-number 1999
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
> !
> dial-peer voice 101 voip
>  preference 1
>  destination-pattern 1997
>  voice-class h323 1
>  session target ipv4:10.10.210.10
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
> Under callmanager i have 1997 as MVA Number defined at Media 
> Ressources->Mobile Voice Access and also at service parameter
>  
> When i call the mva the call comes in via mgcp, on ccm i have a route pattern 
> that sends 1999 back to the h.323 configured gateway, then the service gets 
> invoked. so far so good.
>  
> I have remote destination configured with 8884343 and the call comes in as 
> following:
>  
> Oct  7 21:41:58.277: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref = 
> 0x00B4
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> Progress Ind i = 0x8583 - Origination address is non-ISDN
> Calling Party Number i = 0x4180, 
> Plan:ISDN, Type:Subscriber(local)
> Called Party Number i = 0xA1, '4158881999'
> Plan:ISDN, Type:National
> Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
>Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH

Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

[Stutz, Bernhard]

Hi, 

 

I got an idea how to solve this maybe.

I think the problem is that the call comes into the script via 1999 but
the MVA is 1997.

I will setup this that the MVA is also 1999 therefore you need to have a
CSS on the h.323 gateway that sees only 1999/pt-mva and not the
1999/pt-internal which will send the call back to h.323 gateway.

 

I will tell you if this is the trick...

 

Cheers,

Bernhard

 

 

Von: amr thabt [mailto:amrth...@gmail.com] 
Gesendet: Donnerstag, 7. Oktober 2010 21:01
An: Stutz, Bernhard
Cc: Pithog Oil; ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

 

Hi Stutz,

 1- add translation rule& profile to dial-p 1997 to change the calling
number to be  '8884343' .

 2- if still have a problem , check css of RDP and may restart Mobile
Voice Service

 I hpoe this may help

HTH

AMR

 

On Thu, Oct 7, 2010 at 9:26 PM, Stutz, Bernhard 
wrote:



Hi,

 

I run into the same issue.

furthermore i have to hairpin the call through a h323 gateway as all
incoming calls come per mgcp to the callmanager. You have then to add a
H.323 gateway to the same mgcp gateway which is possible.

 

I got following dial--peers configured:

 

dial-peer voice 1999 voip
 service cmm
 incoming called-number 1999
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 101 voip
 preference 1
 destination-pattern 1997
 voice-class h323 1
 session target ipv4:10.10.210.10
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

Under callmanager i have 1997 as MVA Number defined at Media
Ressources->Mobile Voice Access and also at service parameter

 

When i call the mva the call comes in via mgcp, on ccm i have a route
pattern that sends 1999 back to the h.323 configured gateway, then the
service gets invoked. so far so good.

 

I have remote destination configured with 8884343 and the call comes in
as following:

 

Oct  7 21:41:58.277: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref =
0x00B4
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Progress Ind i = 0x8583 - Origination address is non-ISDN
Calling Party Number i = 0x4180, 
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xA1, '4158881999'
Plan:ISDN, Type:National
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
   Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
Dial-peer=1999
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
   Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:

BR1-RTR#Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
Dial-peer=1999
Oct  7 21:41:58.385: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8
callref = 0x80B4
Channel ID i = 0xA98381
Exclusive, Channel 1
Oct  7 21:41:58.393: ISDN Se0/0/0:23 Q931: TX -> CONNECT pd = 8  callref
= 0x80B4
Oct  7 21:41:58.401: ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8
callref = 0x00B4

 

Then i am getting asked for the pin which is been accepted. after that i
choose option 1 and push 5002#

Then the call gets disconnected:

 

Oct  7 21:42:21.297: //-1//DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1997, Peer Info
Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1997
Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=101
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1997, Peer Info
Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1997
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=101
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Calling Number=, Called Number=8884343, Peer Info
Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Match Ru

[OSL | CCIE_Voice] need urgent solution looking for the study partner

[voice-gang voice-gang]

Need help

Section 12: High Availability

12.1 SiteB router high availability

Configure SRST on SiteB router so that it provides call processing for all
local IP
phones in case of CUCM is not reachable due to WAN issue. Configure
following
requirements,

1) Register all IP phones to SRST. Test inbound and outbound PSTN calls. All
IP phones should be able to make 911, long distance and international calls.
Such calls made should display 10-digit caller ID.

2) Enable IP phones to make maximum 2 3-party conference calls.

3) Make sure that voicemail functionality is restored in event of WAN
failure.
Voicemail forwarding feature should work between IP phones as well as
PSTN calls. When such forwarded call comes to Cisco Unity connection, it
should play user’s personal greeting. You are not allowed to use alternate
extension to achieve this.

(3 points)


I also want to know why people getting less marks in SRST and MRM in lab 4

I am looking for the real study partner if anyone interested please
interested let me know

thks
___
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[OSL | CCIE_Voice] need solution searching for the parter who has lab soon

[voice-gang voice-gang]

Hi,


8.1 Switch QoS

2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for

incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8
and

then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3.

Can anyone tell me how to achieve this.

Thks
___
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[OSL | CCIE_Voice] I need solution

[voice-gang voice-gang]

2.2 IP Phone customization (Part I)

HQ phone 1 should see the DND on the phone screen. Once the requirement has
been met. Do the following.

· DND softkey should be there in idle and active conversation state.


· DND softkey should be available while incoming calls ringing.
___
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