Re: [OSL | CCIE_Voice] CIPC strange behavior !

[jonn cozak]

Dear Sir, 

I have found it !! 

Now if possible can you kindly also let me know what is SRND all about and what 
benefit i may get from these docs ?

The question is for you and for anyone who wishes to answer.

Thanks again for the support 






From: Brian Mulgrew 
To: cisco.jon...@yahoo.com; ccie_voice@onlinestudylist.com
Sent: Sun, March 14, 2010 1:12:53 AM
Subject: RE: [OSL | CCIE_Voice] CIPC strange behavior !

  Hi - think it is touched on briefly on the SRND under Network Infrastructure 
/ Network Svcs / TFTP section:

When a device requests a configuration file from the TFTP server, the TFTP 
server searches for the configuration file in its internal caches, the disk, 
and then alternate Cisco file servers (if specified). If the TFTP server finds 
the configuration file, it sends it to the device. If the configuration file 
provides Unified CM names, the device resolves the name by using DNS and opens 
a connection to the Unified CM. If the device does not receive an IP address or 
name, it uses the TFTP server name or IP address to attempt a registration 
connection. If the TFTP server cannot find the configuration file, it sends a 
"file not found" message to the device.


cheers
b



Date: Sat, 13 Mar 2010 08:06:47 -0800
From: cisco.jon...@yahoo.com
Subject: Re: [OSL | CCIE_Voice] CIPC strange behavior !
To: btmulg...@hotmail.com; ccie_voice@onlinestudylist.com


Dear Brian, thats exactly what is confusing me. Why this happens ? i mean i 
havent read this on any cisco technology doc no matter i much i search. Why 
this behaviour is not documented ? is it a random behavior ? 

i tried the same thing with IP Blue with 7960 selected but it didnt register 
!!, it wasnt able to resolve the hostname and hence registeration wasnt 
successful. Perhaps this is the behaviour that is defined in various study 
guides, is it a special case with CIPC ?






From: Brian Mulgrew 
To: cisco.jon...@yahoo.com; ccie_voice@onlinestudylist.com
Sent: Sat, March 13, 2010 5:46:31 PM
Subject: RE: [OSL | CCIE_Voice] CIPC strange behavior !

 Hi - If I remember correctly if the IP Phone / CIPC cannot find the hostname 
of the CUCM in the cnf.xml it will register with  ccm service running on the 
tftp server - so if your tftp is set as IP this could be the reason why it 
still registers.

thks
b


Date: Fri, 12 Mar 2010 23:45:10 -0800
From: cisco.jon...@yahoo.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CIPC strange behavior !


Hi all. I am using CUCM 6. CIPC is installed on my pc. I am using single 
callmanager server that is also my tftp server for endpoints. Now while doing 
my studies i wanted to check how DNS might cause issues. In System->Server, i 
am using hostname instead of IP. Now what happens is that, CIPC after getting 
the .cnf.xml file, registers with tftp server successfully (which is also my 
callmanager server). Now what i read in student guide was, ip phone should not 
be able to register if its not able to resolve the hostname through DNS. (i am 
not using any DNS server and nor the entry for the hostname is present in my pc 
host file).

Can someone tell me why is this the case ? i have searched alot but i couldnt 
find any thing stating that ip phone will fall back to tftp server in case 
primary callmanager fails !!

Any input on this pls ?



Got a cool Hotmail story? Tell us now


Do you have a story that started on Hotmail? Tell us now


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

Re: [OSL | CCIE_Voice] CIPC strange behavior !

[jonn cozak]

Dear Brian, 

I dont have words to express my gratitude. Thanks alot sir. I just have 2 
queries

What is this SRND ? i mean what shall i expect to find here as compared to 
normal configuration guides or other technologies notes ?

Can you send me the link from where you quoted this 

Thanks alot again Sir





From: Brian Mulgrew 
To: cisco.jon...@yahoo.com; ccie_voice@onlinestudylist.com
Sent: Sun, March 14, 2010 1:12:53 AM
Subject: RE: [OSL | CCIE_Voice] CIPC strange behavior !

  Hi - think it is touched on briefly on the SRND under Network Infrastructure 
/ Network Svcs / TFTP section:

When a device requests a configuration file from the TFTP server, the TFTP 
server searches for the configuration file in its internal caches, the disk, 
and then alternate Cisco file servers (if specified). If the TFTP server finds 
the configuration file, it sends it to the device. If the configuration file 
provides Unified CM names, the device resolves the name by using DNS and opens 
a connection to the Unified CM. If the device does not receive an IP address or 
name, it uses the TFTP server name or IP address to attempt a registration 
connection. If the TFTP server cannot find the configuration file, it sends a 
"file not found" message to the device.


cheers
b



Date: Sat, 13 Mar 2010 08:06:47 -0800
From: cisco.jon...@yahoo.com
Subject: Re: [OSL | CCIE_Voice] CIPC strange behavior !
To: btmulg...@hotmail.com; ccie_voice@onlinestudylist.com


Dear Brian, thats exactly what is confusing me. Why this happens ? i mean i 
havent read this on any cisco technology doc no matter i much i search. Why 
this behaviour is not documented ? is it a random behavior ? 

i tried the same thing with IP Blue with 7960 selected but it didnt register 
!!, it wasnt able to resolve the hostname and hence registeration wasnt 
successful. Perhaps this is the behaviour that is defined in various study 
guides, is it a special case with CIPC ?






From: Brian Mulgrew 
To: cisco.jon...@yahoo.com; ccie_voice@onlinestudylist.com
Sent: Sat, March 13, 2010 5:46:31 PM
Subject: RE: [OSL | CCIE_Voice] CIPC strange behavior !

 Hi - If I remember correctly if the IP Phone / CIPC cannot find the hostname 
of the CUCM in the cnf.xml it will register with  ccm service running on the 
tftp server - so if your tftp is set as IP this could be the reason why it 
still registers.

thks
b


Date: Fri, 12 Mar 2010 23:45:10 -0800
From: cisco.jon...@yahoo.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CIPC strange behavior !


Hi all. I am using CUCM 6. CIPC is installed on my pc. I am using single 
callmanager server that is also my tftp server for endpoints. Now while doing 
my studies i wanted to check how DNS might cause issues. In System->Server, i 
am using hostname instead of IP. Now what happens is that, CIPC after getting 
the .cnf.xml file, registers with tftp server successfully (which is also my 
callmanager server). Now what i read in student guide was, ip phone should not 
be able to register if its not able to resolve the hostname through DNS. (i am 
not using any DNS server and nor the entry for the hostname is present in my pc 
host file).

Can someone tell me why is this the case ? i have searched alot but i couldnt 
find any thing stating that ip phone will fall back to tftp server in case 
primary callmanager fails !!

Any input on this pls ?



Got a cool Hotmail story? Tell us now


Do you have a story that started on Hotmail? Tell us now


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

Re: [OSL | CCIE_Voice] SIP Phone Registration Rejected

[Pavan]

Pretty common mistake.
Looks like you registered a 79xx phone as 79yy while manually adding it.
Double check the phone type from the phone settings menu

Sent from my phone

On Mar 14, 2010, at 8:37 PM, Robertico Gonzalez > wrote:



Hi,

Device > Phones > SIP Phone says "Rejected".

SDI
15:48:07.403 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message  
from 10.94.169.62 on port 52028 index 197 with 2055 bytes:


REGISTER sip:10.94.169.5 SIP/2.0



15:44:47.335 |SIPStationD(1,100,58,181), SEP001BD5E86F14,  
10.94.169.62:49901, primaryDN=9503502, DevStat-StopClose:  
LineRegisterReq send failure| 
1,100,56,1.559^10.94.169.62^SEP001BD5E86F14




SDL

31428| 2010/03/14 15:48:07.510| 001| SdlSig|  
SIPRegisterResp   |  
wait  | SIPHandler(1,100,63,1)  |  
SIPStationD(1,100,58,189)   | (1,100,56,1).581- 
(SEP001BD5E86F14:10.94.169.62)| [T:NP - HP: 0, NP: 1, LP: 0, VLP: 0,  
LZP: 0 DBP: 0] ccbID= 386 --TransType=1 --TransSecurity=0 PeerAddr=  
10.94.169.62:52028 respCode= 404



Syslog:

: 71: bof-cma1.cisco.com: Mar 14 2010 20:56:22.912 UTC :  
%UC_CALLMANAGER-3-DbInfoError: %[DeviceName=SEP001BD5E86F14] 
[ClusterID=StandAloneCluster][NodeID=bof-cma1]: Configuration  
information may be out of sync for the device and Unified CM database




: 72: bof-cma1.cisco.com: Mar 14 2010 20:56:26.237 UTC :  
%UC_CALLMANAGER-3-DeviceTypeMismatch: %[DeviceName=SEP001BD5E86D54] 
[DeviceType=30006][DBDeviceType=255][ClusterID=StandAloneCluster] 
[NodeID=bof-cma1]: Device type mismatch between the information  
contained in the device's TFTP configuration file and what is  
configured in Unified CM Administration for that device



Any suggestions on what to do?

Thanks,
RG
___
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] MVA

[Roger Källberg]

Maybe an obvious thing, but has you set the MVA DN under Media Resources and 
can the RDP/RD see that DN in its CSS? And also can the RDP/RD see the RP that 
are used for external calls?
You might also want to restart the Mobile Voice service in UCM, that's 
sometimes needed after you have setup MVA-
Roger Källberg
Unified Communication Consultant
Cygate AB


From: anupam TYAGI [mailto:anuf...@gmail.com]
Sent: den 15 mars 2010 04:44
To: Otto Sanchez
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MVA

it is configured , i am able to reach MVA prompts , but when i am dialing the 
external when in MVA my call disconnects
On Mon, Mar 15, 2010 at 8:37 AM, Otto Sanchez 
mailto:o...@ipexpert.com>> wrote:
Hi,

I'm referring to the mva number you have configured in the ucm->media 
resources->mobile voice access config, do you have a voip dial-peer matching 
this number as destination and pointing to the ucm?


On Sun, Mar 14, 2010 at 10:33 PM, anupam TYAGI 
mailto:anuf...@gmail.com>> wrote:
I have the dial-peer on the gateway which matches the external number when i am 
dialing through MVA .

i am able to dial this external number when i am not in MVA

On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez 
mailto:o...@ipexpert.com>> wrote:
Hi,

Did you make sure that you have a dial peer in the hq rtr which destination 
number match the mva number you have in the ucm configuration?, this is needed 
to route calls outbound calls from the remote devices connected to the mva 
service,

Thanks,
On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI 
mailto:anuf...@gmail.com>> wrote:
i have the route pattern partion assigned to the CSS and this CSS  is assigned 
to RDP >but still the call disconnect when i dial the external number in MVZ
On Fri, Mar 12, 2010 at 11:06 PM, Omotayo 
mailto:adefilabi...@gmail.com>> wrote:
Hello,

Berry is right.

create a partition called pt-mva

crease a CSS called css-mva

put the partition in the css

create a route pattern like 9.011! in partition pt-mva. the gateway can be the 
hq gateway if you wish
discard predot

assign the css to the remote destination profile

this will work for you




On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI 
mailto:anuf...@gmail.com>> wrote:
if i dial that external number without MVA it goes through ,but when in MVA i 
get a disconnect when calling this external number ( so don't seems to be codec 
issue )

On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer 
mailto:myciscov...@gmail.com>> wrote:
are you maybe calling to a remote location (g.729) and therefore a xcoder is 
required, but not set up correctly?
2010/3/12 anupam TYAGI mailto:anuf...@gmail.com>>
i saw the call hit the gateway .  RDP is having the same CSS as phone CSS


On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. 
mailto:mjbe...@krollontrack.com>> wrote:
Check the CSS on the remote destination profile you're calling from.
If you do a "debug isdn q931" on the PSTN gateway, do you see the call hit the 
gateway?

Your rerouting CSS on the RDP is used for calls out to your RD.
Your CSS on the RDP is used for calls through MVA that are routed out through 
your PSTN gateway.

From: 
ccie_voice-boun...@onlinestudylist.com
 
[mailto:ccie_voice-boun...@onlinestudylist.com]
 On Behalf Of anupam TYAGI
Sent: Friday, March 12, 2010 9:27 AM
To: 
ccie_voice-requ...@onlinestudylist.com;
 ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MVA

Hi Folks

I am doing MVA , When i dial the MVA number ,  I am able to hear the  prompt.  
I dial a  PSTN number , but the call disconnect . Can any body suggest me what 
can be the reason .


Rgds
Anu.


___
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


--
Regards,

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




--
Regards,

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

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

Re: [OSL | CCIE_Voice] MVA

[Otto Sanchez]

Would you please post your gw config?

thanks,

On Sun, Mar 14, 2010 at 11:13 PM, anupam TYAGI  wrote:

> it is configured , i am able to reach MVA prompts , but when i am dialing
> the external when in MVA my call disconnects
>
>
> On Mon, Mar 15, 2010 at 8:37 AM, Otto Sanchez  wrote:
>
>> Hi,
>>
>> I'm referring to the mva number you have configured in the ucm->media
>> resources->mobile voice access config, do you have a voip dial-peer matching
>> this number as destination and pointing to the ucm?
>>
>>
>> On Sun, Mar 14, 2010 at 10:33 PM, anupam TYAGI  wrote:
>>
>>> I have the dial-peer on the gateway which matches the external number
>>> when i am dialing through MVA .
>>>
>>> i am able to dial this external number when i am not in MVA
>>>
>>>
>>> On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez  wrote:
>>>
 Hi,

 Did you make sure that you have a dial peer in the hq rtr which
 destination number match the mva number you have in the ucm configuration?,
 this is needed to route calls outbound calls from the remote devices
 connected to the mva service,

 Thanks,

   On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI wrote:

> i have the route pattern partion assigned to the CSS and this CSS  is
> assigned to RDP >but still the call disconnect when i dial the external
> number in MVZ
>
> On Fri, Mar 12, 2010 at 11:06 PM, Omotayo wrote:
>
>> Hello,
>>
>> Berry is right.
>>
>> create a partition called pt-mva
>>
>> crease a CSS called css-mva
>>
>> put the partition in the css
>>
>> create a route pattern like 9.011! in partition pt-mva. the gateway
>> can be the hq gateway if you wish
>> discard predot
>>
>> assign the css to the remote destination profile
>>
>> this will work for you
>>
>>
>>
>>
>>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI 
>> wrote:
>>
>>> if i dial that external number without MVA it goes through ,but when
>>> in MVA i get a disconnect when calling this external number ( so don't 
>>> seems
>>> to be codec issue )
>>>
>>>
>>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer <
>>> myciscov...@gmail.com> wrote:
>>>
 are you maybe calling to a remote location (g.729) and therefore a
 xcoder is required, but not set up correctly?

 2010/3/12 anupam TYAGI 

>  i saw the call hit the gateway .  RDP is having the same CSS as
> phone CSS
>
>
>
> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
> mjbe...@krollontrack.com> wrote:
>
>>   Check the CSS on the remote destination profile you’re calling
>> from.
>>
>> If you do a “debug isdn q931” on the PSTN gateway, do you see the
>> call hit the gateway?
>>
>>
>>
>> Your rerouting CSS on the RDP is used for calls out to your RD.
>>
>> Your CSS on the RDP is used for calls through MVA that are routed
>> out through your PSTN gateway.
>>
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam
>> TYAGI
>> *Sent:* Friday, March 12, 2010 9:27 AM
>> *To:* ccie_voice-requ...@onlinestudylist.com;
>> ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] MVA
>>
>>
>>
>> Hi Folks
>>
>> I am doing MVA , When i dial the MVA number ,  I am able to hear
>> the  prompt.  I dial a  PSTN number , but the call disconnect . Can 
>> any body
>> suggest me what can be the reason .
>>
>>
>> Rgds
>> Anu.
>>
>
>
> ___
> 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
>
>


 --
 Regards,

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

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


-- 
Regards,

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

Re: [OSL | CCIE_Voice] MVA

[anupam TYAGI]

it is configured , i am able to reach MVA prompts , but when i am dialing
the external when in MVA my call disconnects

On Mon, Mar 15, 2010 at 8:37 AM, Otto Sanchez  wrote:

> Hi,
>
> I'm referring to the mva number you have configured in the ucm->media
> resources->mobile voice access config, do you have a voip dial-peer matching
> this number as destination and pointing to the ucm?
>
>
> On Sun, Mar 14, 2010 at 10:33 PM, anupam TYAGI  wrote:
>
>> I have the dial-peer on the gateway which matches the external number when
>> i am dialing through MVA .
>>
>> i am able to dial this external number when i am not in MVA
>>
>>
>> On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez  wrote:
>>
>>> Hi,
>>>
>>> Did you make sure that you have a dial peer in the hq rtr which
>>> destination number match the mva number you have in the ucm configuration?,
>>> this is needed to route calls outbound calls from the remote devices
>>> connected to the mva service,
>>>
>>> Thanks,
>>>
>>>   On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI wrote:
>>>
 i have the route pattern partion assigned to the CSS and this CSS  is
 assigned to RDP >but still the call disconnect when i dial the external
 number in MVZ

 On Fri, Mar 12, 2010 at 11:06 PM, Omotayo wrote:

> Hello,
>
> Berry is right.
>
> create a partition called pt-mva
>
> crease a CSS called css-mva
>
> put the partition in the css
>
> create a route pattern like 9.011! in partition pt-mva. the gateway can
> be the hq gateway if you wish
> discard predot
>
> assign the css to the remote destination profile
>
> this will work for you
>
>
>
>
>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI wrote:
>
>> if i dial that external number without MVA it goes through ,but when
>> in MVA i get a disconnect when calling this external number ( so don't 
>> seems
>> to be codec issue )
>>
>>
>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer <
>> myciscov...@gmail.com> wrote:
>>
>>> are you maybe calling to a remote location (g.729) and therefore a
>>> xcoder is required, but not set up correctly?
>>>
>>> 2010/3/12 anupam TYAGI 
>>>
  i saw the call hit the gateway .  RDP is having the same CSS as
 phone CSS



 On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
 mjbe...@krollontrack.com> wrote:

>   Check the CSS on the remote destination profile you’re calling
> from.
>
> If you do a “debug isdn q931” on the PSTN gateway, do you see the
> call hit the gateway?
>
>
>
> Your rerouting CSS on the RDP is used for calls out to your RD.
>
> Your CSS on the RDP is used for calls through MVA that are routed
> out through your PSTN gateway.
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam
> TYAGI
> *Sent:* Friday, March 12, 2010 9:27 AM
> *To:* ccie_voice-requ...@onlinestudylist.com;
> ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MVA
>
>
>
> Hi Folks
>
> I am doing MVA , When i dial the MVA number ,  I am able to hear
> the  prompt.  I dial a  PSTN number , but the call disconnect . Can 
> any body
> suggest me what can be the reason .
>
>
> Rgds
> Anu.
>


 ___
 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


>>>
>>>
>>> --
>>> Regards,
>>>
>>> Otto Sanchez
>>> CCIE #25592 (Voice)
>>> Support Engineer - IPexpert, Inc.
>>> URL: http://www.IPexpert.com 
>>>
>>
>>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] MVA

[Otto Sanchez]

Hi,

I'm referring to the mva number you have configured in the ucm->media
resources->mobile voice access config, do you have a voip dial-peer matching
this number as destination and pointing to the ucm?


On Sun, Mar 14, 2010 at 10:33 PM, anupam TYAGI  wrote:

> I have the dial-peer on the gateway which matches the external number when
> i am dialing through MVA .
>
> i am able to dial this external number when i am not in MVA
>
>
> On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez  wrote:
>
>> Hi,
>>
>> Did you make sure that you have a dial peer in the hq rtr which
>> destination number match the mva number you have in the ucm configuration?,
>> this is needed to route calls outbound calls from the remote devices
>> connected to the mva service,
>>
>> Thanks,
>>
>>   On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI wrote:
>>
>>> i have the route pattern partion assigned to the CSS and this CSS  is
>>> assigned to RDP >but still the call disconnect when i dial the external
>>> number in MVZ
>>>
>>> On Fri, Mar 12, 2010 at 11:06 PM, Omotayo wrote:
>>>
 Hello,

 Berry is right.

 create a partition called pt-mva

 crease a CSS called css-mva

 put the partition in the css

 create a route pattern like 9.011! in partition pt-mva. the gateway can
 be the hq gateway if you wish
 discard predot

 assign the css to the remote destination profile

 this will work for you




On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI wrote:

> if i dial that external number without MVA it goes through ,but when in
> MVA i get a disconnect when calling this external number ( so don't seems 
> to
> be codec issue )
>
>
> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer <
> myciscov...@gmail.com> wrote:
>
>> are you maybe calling to a remote location (g.729) and therefore a
>> xcoder is required, but not set up correctly?
>>
>> 2010/3/12 anupam TYAGI 
>>
>>>  i saw the call hit the gateway .  RDP is having the same CSS as
>>> phone CSS
>>>
>>>
>>>
>>> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
>>> mjbe...@krollontrack.com> wrote:
>>>
   Check the CSS on the remote destination profile you’re calling
 from.

 If you do a “debug isdn q931” on the PSTN gateway, do you see the
 call hit the gateway?



 Your rerouting CSS on the RDP is used for calls out to your RD.

 Your CSS on the RDP is used for calls through MVA that are routed
 out through your PSTN gateway.



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
 *Sent:* Friday, March 12, 2010 9:27 AM
 *To:* ccie_voice-requ...@onlinestudylist.com;
 ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA



 Hi Folks

 I am doing MVA , When i dial the MVA number ,  I am able to hear
 the  prompt.  I dial a  PSTN number , but the call disconnect . Can 
 any body
 suggest me what can be the reason .


 Rgds
 Anu.

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


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5

[CCIETalk.com]

Lab 5 is mgcp sir

On 3/14/10, Berry, Matthew J.  wrote:
> Might be simple, but do you have your H323 dial peers setup correctly?
> - Sent from my Blackberry
>
> 
> From: ccie_voice-boun...@onlinestudylist.com
> 
> To: Mike Todd 
> Cc: ccie_voice 
> Sent: Sun Mar 14 20:57:51 2010
> Subject: Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5
>
> Did you ever get this figured out?
>
> On Wed, Jan 6, 2010 at 1:14 PM, Mike Todd
> mailto:michaelt...@gmail.com>> wrote:
> I'm having problems figuring out how I'm supposed to be able to dial from
> certain lines to certain sites from the PSTN phone in this lab. I can dial
> fine from line 2 to the HQ site using 10 digit dialing, but when I try
> dialing from Line 3 or 4 to the same site I can't get any calls into the HQ
> router (no matter the way I dial). I've tried using the full E164 with and
> without various access codes (00, 000, 011, 900, 9000, 9011) and I get a
> busy signal for each call with no output on my HQ router debug ISDN q931.
>
> Any ideas? I'm sure I'm missing something stupid here...
>
> Thanks in advance!
>
> Mike Todd
> CCIE #10858 (Routing and Switching, Security) (and hopefully voice soon!)
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
> --
> www.ccietalk.com
>

-- 
Sent from my mobile device

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

Re: [OSL | CCIE_Voice] MVA

[anupam TYAGI]

I have the dial-peer on the gateway which matches the external number when i
am dialing through MVA .

i am able to dial this external number when i am not in MVA

On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez  wrote:

> Hi,
>
> Did you make sure that you have a dial peer in the hq rtr which destination
> number match the mva number you have in the ucm configuration?, this is
> needed to route calls outbound calls from the remote devices connected to
> the mva service,
>
> Thanks,
>
> On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI  wrote:
>
>> i have the route pattern partion assigned to the CSS and this CSS  is
>> assigned to RDP >but still the call disconnect when i dial the external
>> number in MVZ
>>
>> On Fri, Mar 12, 2010 at 11:06 PM, Omotayo  wrote:
>>
>>> Hello,
>>>
>>> Berry is right.
>>>
>>> create a partition called pt-mva
>>>
>>> crease a CSS called css-mva
>>>
>>> put the partition in the css
>>>
>>> create a route pattern like 9.011! in partition pt-mva. the gateway can
>>> be the hq gateway if you wish
>>> discard predot
>>>
>>> assign the css to the remote destination profile
>>>
>>> this will work for you
>>>
>>>
>>>
>>>
>>>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI wrote:
>>>
 if i dial that external number without MVA it goes through ,but when in
 MVA i get a disconnect when calling this external number ( so don't seems 
 to
 be codec issue )


 On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer <
 myciscov...@gmail.com> wrote:

> are you maybe calling to a remote location (g.729) and therefore a
> xcoder is required, but not set up correctly?
>
> 2010/3/12 anupam TYAGI 
>
>>  i saw the call hit the gateway .  RDP is having the same CSS as
>> phone CSS
>>
>>
>>
>> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
>> mjbe...@krollontrack.com> wrote:
>>
>>>   Check the CSS on the remote destination profile you’re calling
>>> from.
>>>
>>> If you do a “debug isdn q931” on the PSTN gateway, do you see the
>>> call hit the gateway?
>>>
>>>
>>>
>>> Your rerouting CSS on the RDP is used for calls out to your RD.
>>>
>>> Your CSS on the RDP is used for calls through MVA that are routed out
>>> through your PSTN gateway.
>>>
>>>
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
>>> *Sent:* Friday, March 12, 2010 9:27 AM
>>> *To:* ccie_voice-requ...@onlinestudylist.com;
>>> ccie_voice@onlinestudylist.com
>>> *Subject:* [OSL | CCIE_Voice] MVA
>>>
>>>
>>>
>>> Hi Folks
>>>
>>> I am doing MVA , When i dial the MVA number ,  I am able to hear the
>>> prompt.  I dial a  PSTN number , but the call disconnect . Can any body
>>> suggest me what can be the reason .
>>>
>>>
>>> Rgds
>>> Anu.
>>>
>>
>>
>> ___
>> 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
>>
>>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5

[Berry, Matthew J.]

Might be simple, but do you have your H323 dial peers setup correctly?
- Sent from my Blackberry


From: ccie_voice-boun...@onlinestudylist.com 

To: Mike Todd 
Cc: ccie_voice 
Sent: Sun Mar 14 20:57:51 2010
Subject: Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5

Did you ever get this figured out?

On Wed, Jan 6, 2010 at 1:14 PM, Mike Todd 
mailto:michaelt...@gmail.com>> wrote:
I'm having problems figuring out how I'm supposed to be able to dial from 
certain lines to certain sites from the PSTN phone in this lab. I can dial fine 
from line 2 to the HQ site using 10 digit dialing, but when I try dialing from 
Line 3 or 4 to the same site I can't get any calls into the HQ router (no 
matter the way I dial). I've tried using the full E164 with and without various 
access codes (00, 000, 011, 900, 9000, 9011) and I get a busy signal for each 
call with no output on my HQ router debug ISDN q931.

Any ideas? I'm sure I'm missing something stupid here...

Thanks in advance!

Mike Todd
CCIE #10858 (Routing and Switching, Security) (and hopefully voice soon!)

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




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

Re: [OSL | CCIE_Voice] Remote Destination/MVA/SNR in depth analysis

[CCIETalk.com]

I am working on this one right now and running into the same issue.
Following the PG and changing the service param to partial match DOES NOT
work. BR1 Ph2 is showing +4402059432785 rather than extension 4002 :\

On Wed, Dec 30, 2009 at 4:36 PM, Hawkins Jason L NGA-ES USA CTR <
jason.l.hawkins@nga.mil> wrote:

> Hello Otto,
>
> My thoughts exactly.  What is the purpose of "Number of Digits" service
> parameter?
>
>Service Param = Partial Match, 10 digits
> (Reset CCM and MVA service)
>BR1 GW CLID = 44 020 5943 2785
>
> Change HQ phone 2 remote destination to 1102059432785
> PSTN dials into BR1 phone 2 "04 4793 3002"
> BR1 phone 2 show 4402059432785
>
> Based on the above test the remote destination should have been matched
> but it wasn't.
>
> Maybe the number of digits is used to control how many digits is
> required from the PSTN not which digits to match.  Because the PSTN is
> sending 13 digits CLID the remote destination must match all 13 digits
> but the remote destination can be equal length or longer.  CUCM is
> configured to allow matching of only 10 digits if the PSTN only sent 10
> digits.  But what would happen if the PSTN only sent 8 digits but CUCM
> was configured to partially match 10, would the remote destination still
> work?
>
> Service Param = Partial Match, 10 digits
> BR1 GW CLID = 5943 2785 (reduced to 8 digits)
>
> HQ phone 2 remote destination = +4402059432785
>  PSTN dials into BR1 phone 2 "04 4793 3002"
>  BR1 phone 2 show +59432785
>
> Change Service Param = Partial Match, 8 digits
>  PSTN dials into BR1 phone 2 "04 4793 3002"
>  BR1 phone 2 show 4002
>
> The answer is no.
>
> The number of digits setting is used by CUCM to determine the minimum
> length the CLID from the PSTN must be to match a remote destination.
> Each digit from the PSTN must match a digit in the remote destination
> starting from the right.  When CUCM was configured to partially match 10
> digits but the PSTN was changed to only send 8 digits a remote
> destination was not matched even though all digits were matched in the
> remote destination number and the remote destination number was longer.
> When the number of digits was changed to 8 the remote destination
> matched.  This setting maybe useful in the event the PSTN sends a
> different length CLID into different gateways.
>
> Jason
>
>
> -Original Message-
> From: Otto Sanchez [mailto:o...@ipexpert.com]
> Sent: Thursday, December 24, 2009 3:09 PM
> To: Hawkins Jason L NGA-ES USA CTR
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Remote Destination/MVA/SNR in depth
> analysis
>
> Hello Jason,
>
> It seems that according to your tests, partial matching of CLID is not
> working as we might expect, If it was working right what would be the
> function of "Number of Digits for Caller ID Partial Match" parameter? it
> seems that no matter the number you set there, a full CLID match is
> still needed in the RD number,
>
> Did you take a look at the cucm traces when partial match and different
> Number of digits is used in the service parameter?,
>
> Thanks,
>
>
> On Mon, Dec 21, 2009 at 7:33 PM, Hawkins Jason L NGA-ES USA CTR
>  wrote:
>
>
>
>I am trying to understand the correlation of the remote
> destination
>number and the number that shows up on the CALLED phone.  I'm
> referring
>to Volume 2 Lab 5 question 3.1.
>
>On page 67 of the proctor guide it says that the "GW Incoming
> Calling
>Party Prefix is not used" for matching a remote destination but
> the
>proctor guide solution was not giving the result the question
> was asking
>for.
>
>When I tried configuring the "Destination Number" to 10 digits,
> inbound
>calls from the PSTN rang the CALLED phone but the display showed
> the
>full E.164 number (+44 020 5943 2785) and the desk phone didn't
> indicate
>a call was present.  I tried setting partial match with 10
> digits but
>that didn't work.  I confirmed that the CALLING number seen on
> the HQ
>gateway was only 10 digits.  The only way I could get the remote
>destination to match was to configure the "Destination Number"
> to the
>global form of the number which is done after the "GW Incoming
> Calling
>Party Prefix" is applied.  So unless I am reading something
> wrong or
>have something configured all wacky (which is probably the case)
> the
>remote destination number has to be the globalized number.
>
>The HQ gateway is MGCP.
> (no mgcp/mgcp)
>The BR1 gateway is MGCP.
> (no mgcp/mgcp)
>Service Param = Partial Match, 10 digits
> (Reset CCM and MVA service)
>BR1 GW CLID = 44 020 5943 2785
>
>Set HQ phone 2 (4002) remote destination to 20 5943 2785
> PSTN dials into BR1 phone 2 "04 4793 3002"
> BR1 phone 2 show +4402059432785
>
>Change

Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5

[CCIETalk.com]

Did you ever get this figured out?

On Wed, Jan 6, 2010 at 1:14 PM, Mike Todd  wrote:

> I'm having problems figuring out how I'm supposed to be able to dial from
> certain lines to certain sites from the PSTN phone in this lab. I can dial
> fine from line 2 to the HQ site using 10 digit dialing, but when I try
> dialing from Line 3 or 4 to the same site I can't get any calls into the HQ
> router (no matter the way I dial). I've tried using the full E164 with and
> without various access codes (00, 000, 011, 900, 9000, 9011) and I get a
> busy signal for each call with no output on my HQ router debug ISDN q931.
>
> Any ideas? I'm sure I'm missing something stupid here...
>
> Thanks in advance!
>
> Mike Todd
> CCIE #10858 (Routing and Switching, Security) (and hopefully voice soon!)
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


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

[OSL | CCIE_Voice] SIP Phone Registration Rejected

[Robertico Gonzalez]

Hi,

Device > Phones > SIP Phone says "Rejected".

SDI

15:48:07.403 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from
10.94.169.62 on port 52028 index 197 with 2055 bytes:

REGISTER sip:10.94.169.5 SIP/2.0


15:44:47.335 |SIPStationD(1,100,58,181), SEP001BD5E86F14, 10.94.169.62:49901,
primaryDN=9503502, DevStat-StopClose: *LineRegisterReq send failure*
|1,100,56,1.559^10.94.169.62^SEP001BD5E86F14



SDL

31428| 2010/03/14 15:48:07.510| 001| SdlSig|
SIPRegisterResp   | wait  |
SIPHandler(1,100,63,1)  | SIPStationD(1,100,58,189)   |
(1,100,56,1).581-(SEP001BD5E86F14:10.94.169.62)| [T:NP - HP: 0, NP: 1, LP:
0, VLP: 0, LZP: 0 DBP: 0] ccbID= 386 --TransType=1 --TransSecurity=0
PeerAddr= 10.94.169.62:52028 respCode= 404

Syslog:

: 71: bof-cma1.cisco.com: Mar 14 2010 20:56:22.912 UTC :
%UC_CALLMANAGER-3-DbInfoError:
%[DeviceName=*SEP001BD5E86F14*][ClusterID=StandAloneCluster][NodeID=bof-cma1]:
Configuration information may be out of sync for the device and Unified CM
database



: 72: bof-cma1.cisco.com: Mar 14 2010 20:56:26.237 UTC :
%UC_CALLMANAGER-3-DeviceTypeMismatch:
%[DeviceName=*SEP001BD5E86D54*][DeviceType=30006][DBDeviceType=255][ClusterID=StandAloneCluster][NodeID=bof-cma1]:
Device type mismatch between the information contained in the device's TFTP
configuration file and what is configured in Unified CM Administration for
that device

Any suggestions on what to do?

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

Re: [OSL | CCIE_Voice] No Audio to UCCX Agent

[Mike Brooks]

Please disregard.  Rebooting the HQ router (transcoder) resolved it.

Thx,
Mike Brooks
CCIE#16027 (R&S)

On Sun, Mar 14, 2010 at 5:49 PM, Mike Brooks <2xcci...@gmail.com> wrote:

> I am missing something and need some suggestions.
>
> 1. PSTN phone calls UCCX RP at HQ.
> 2. BR1PH2 agent picks up call.
> 3. No Audio Between PSTN Phone and BR1PH2
> 4. Transcoder is invoked on HQ.
>
> HQ-RTR#sho sccp connections
> sess_idconn_idstype mode codec   ripaddr rport sport
> 33555438   33554535   xcode recvonly g711u   0.0.0.0 0 17302
> 33555438   33554533   xcode recvonly g729ab  0.0.0.0 0 17112
> Total number of active session(s) 1, and connection(s) 2
>
> - According to help button on PSTN phone and BR1PH2 both are sending RTP
> packets but not receiving.
> - If agent HQPH1 picks up audio works.
>
> Any suggestions ?
>
>
> Thanks,
> Mike Brooks
> CCIE#16027 (R&S)
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] No Audio to UCCX Agent

[Mike Brooks]

I am missing something and need some suggestions.

1. PSTN phone calls UCCX RP at HQ.
2. BR1PH2 agent picks up call.
3. No Audio Between PSTN Phone and BR1PH2
4. Transcoder is invoked on HQ.

HQ-RTR#sho sccp connections
sess_idconn_idstype mode codec   ripaddr rport sport
33555438   33554535   xcode recvonly g711u   0.0.0.0 0 17302
33555438   33554533   xcode recvonly g729ab  0.0.0.0 0 17112
Total number of active session(s) 1, and connection(s) 2

- According to help button on PSTN phone and BR1PH2 both are sending RTP
packets but not receiving.
- If agent HQPH1 picks up audio works.

Any suggestions ?


Thanks,
Mike Brooks
CCIE#16027 (R&S)
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[Otto Sanchez]

Hi Jason,

Thanks for the information, what router/ interfaces are you using?

Thanks!

On Fri, Mar 12, 2010 at 1:04 PM, Jason Granat  wrote:

>  Hi Otto,
>
>
>
> Thanks for the advice. In your second paragraph the opposite was actually
> the case. The E1 voice-ports were originally showing a-law, and had
> distortion. I hard set u-law on the E1 ports between the gateway and PSTN
> router and the distortion went away. Perhaps that is what you meant?
>
>
>
> I took a look at the link you included. I’ll have to do some testing but my
> main question is how is this handled in the real world at the provider
> level?
>
>
>
> Thanks,
>
>
>
> Jason
>
>
>
> *From:* Otto Sanchez [mailto:o...@ipexpert.com]
> *Sent:* Friday, March 12, 2010 4:59 AM
> *To:* Jason Granat
> *Cc:* 
> *Subject:* Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1
>
>
>
> Hello Jason,
>
> E1's and T1's will always use a-law and u-law companding mechanism
> respectively, this is used to give more "resolution" to low voice
> frequencies when digitizing an analog signal (the mechanism is also used in
> the other end for digital to analogue conversion), each mechanism is
> designed exclusively to work with its voice digital standard and cannot be
> used conversely,
>
> In that sense, my guess is that before applying that command in your E1
> port, the companding type was u-law, you can verify this using the sh voice
> port command (perhaps the default configuration of a-law was somehow
> overwritten by a cptone command in the same port configuration), and when
> you hardcoded the a-law companding type everything worked as expected,
>
> I also found a note in the Cisco IOS Voice Port Configuration Guide, which
> says that the command is used when cross-connecting in a local router,
>
>
> http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871
>
>
> HTH,
>
> On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat  wrote:
>
> So I’ve got this partially figured out. It had to do with the compand-type.
> E1 was a-law and T1 was u-law. I set the E1 side for u-law and it sounds
> correct now.
>
>
>
> The final thing I am trying to figure out is how to ‘trans-compand’ (if
> that is the correct term) on the PSTN gateway. As it sits I had to change
> the compand-type between the PSTN and E1 gateway. I don’t have experience
> with foreign connectivity so maybe this is the way it is done in the real
> world but I am thinking that perhaps the E1 site may not want or be able to
> change their compand-type, so can it be changed at the PSTN level between
> a-law and u-law locations?
>
>
>
> Thanks,
>
>
>
> Jason
>
>
>
> *From:* Jason Granat
>
> *Sent:* Thursday, March 11, 2010 9:46 AM
> *To:* 
>
> *Subject:* PSTN Call Distortion Between T1/E1
>
>
>
> Perhaps this is something simple that I am overlooking but I have the
> generic setup running in my home lab with 3 gateways and one PSTN router. 2
> of the gateways are T1 and one is E1. The PSTN router is also running CME
> with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN
> phone are fine. Calls between T1 sites are fine. Calls between T1 and E1
> sites are distorted, like the gain is way too high. I tried playing with the
> gain on the voice-port but no luck. I’m not finding much online or in Cisco
> docs. Any suggestions?
>
>
>
> Thanks,
>
>
>
> Jason
>
>
>  --
>
>
>
> http://slash128.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com 
>
> --
>
>
> http://slash128.com
>



-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] MVA

[Otto Sanchez]

Hi,

Did you make sure that you have a dial peer in the hq rtr which destination
number match the mva number you have in the ucm configuration?, this is
needed to route calls outbound calls from the remote devices connected to
the mva service,

Thanks,

On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI  wrote:

> i have the route pattern partion assigned to the CSS and this CSS  is
> assigned to RDP >but still the call disconnect when i dial the external
> number in MVZ
>
> On Fri, Mar 12, 2010 at 11:06 PM, Omotayo  wrote:
>
>> Hello,
>>
>> Berry is right.
>>
>> create a partition called pt-mva
>>
>> crease a CSS called css-mva
>>
>> put the partition in the css
>>
>> create a route pattern like 9.011! in partition pt-mva. the gateway can be
>> the hq gateway if you wish
>> discard predot
>>
>> assign the css to the remote destination profile
>>
>> this will work for you
>>
>>
>>
>>
>>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI wrote:
>>
>>> if i dial that external number without MVA it goes through ,but when in
>>> MVA i get a disconnect when calling this external number ( so don't seems to
>>> be codec issue )
>>>
>>>
>>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer >> > wrote:
>>>
 are you maybe calling to a remote location (g.729) and therefore a
 xcoder is required, but not set up correctly?

 2010/3/12 anupam TYAGI 

>  i saw the call hit the gateway .  RDP is having the same CSS as phone
> CSS
>
>
>
> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. <
> mjbe...@krollontrack.com> wrote:
>
>>   Check the CSS on the remote destination profile you’re calling
>> from.
>>
>> If you do a “debug isdn q931” on the PSTN gateway, do you see the call
>> hit the gateway?
>>
>>
>>
>> Your rerouting CSS on the RDP is used for calls out to your RD.
>>
>> Your CSS on the RDP is used for calls through MVA that are routed out
>> through your PSTN gateway.
>>
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI
>> *Sent:* Friday, March 12, 2010 9:27 AM
>> *To:* ccie_voice-requ...@onlinestudylist.com;
>> ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] MVA
>>
>>
>>
>> Hi Folks
>>
>> I am doing MVA , When i dial the MVA number ,  I am able to hear the
>> prompt.  I dial a  PSTN number , but the call disconnect . Can any body
>> suggest me what can be the reason .
>>
>>
>> Rgds
>> Anu.
>>
>
>
> ___
> 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
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[Otto Sanchez]

Hi,

I think you are referring to the lab5c task 5.2, have you take a look at the
pg solution?, there's a very good explanation there on how the hq resources
are invoked, but always keep in mind that the transcoder resources are
invoked where the codec mismatch occurs,

Take a look and let us know,

Thanks,

On Fri, Mar 12, 2010 at 8:34 PM, Omotayo  wrote:

>  Hello Jeff,
>
> All calls worked when i configure the xcoder on the cme
>
> The question says use the hq router resources- that is where i have issues
>
> thanks
>
>   On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter  wrote:
>
>>  FYI, I was only able to get this to work using transcoder on CME.  Had
>> to match the codec between UCM trunk and incoming dial-peer on CME…then
>> xcoder would engage on CME for the SIP phone.  I have a hardware limitation
>> in my home lab so I am not able to configure a xcoder on both UCM and CME
>> simultaneously.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *From:* Omotayo [mailto:adefilabi...@gmail.com]
>> *Sent:* Friday, March 12, 2010 6:33 AM
>> *To:* Otto Sanchez
>> *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com
>>
>> *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
>>
>>
>>
>> Hello Otto,
>>
>>
>>
>> i had same issue
>>
>>
>>
>> The transcoder can be on the trunk?
>>
>>
>>
>> When i did the transcoder on the br2 router, i get a busy tone when the
>> sip phone is being called from the hq phone
>>
>>
>>
>> REgards
>>
>> On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez  wrote:
>>
>> Hi Jeff,
>>
>> Would you please tell us more about the call flow and the end to end codec
>> requirements for this call. If doing g.729 over the wan, and your sip phone
>> is using g.711 you should transcode at br2,
>>
>> Please let us know,
>>
>> Thanks,
>>
>> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter  wrote:
>>
>>   Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder
>> on UCM.  Can’t seem to get a call from Call Manager to CME sip phone
>> working.  I can call from CME to UCM but not the other way around. Rings but
>> disconnects when answered.  Transcoder shows registered in Call manager.
>> Thanks
>>
>>
>>
>>
>>
>> Jeff
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>>
>>
>> --
>> Regards,
>>
>> Otto Sanchez
>> CCIE #25592 (Voice)
>> Support Engineer - IPexpert, Inc.
>> URL: http://www.IPexpert.com 
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>>
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] UCM with SIP account Configuration Implementation

[Saeed IDris]

Hi ,
Can someone guide me whom have experience with configure SIP account from
SIP Provider with Cisco Voice Gateway, the idea is to point UCM users to
point for all International calls to pass via Cisco Voice Gateway router
then --INTERNET -SIP Provider :
Here is my SIP user account configuration:

sip-ua
authentication username 1777XXX password X
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
mwi-server ipv4:204.11.192.36 expires 3600 port 5060 transport udp
unsolicited
registrar ipv4:204.11.192.36 expires 3600
sip-server ipv4:204.11.192.36





Regards,



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

Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK

[CCIETalk.com]

I am not near my computer but it says V3

On 3/14/10, scott carruthers  wrote:
>
> Is the lab you are referring to - lab 5C - in CCIE Voice Workbook V6.0
> Volume 1?  When I download the section labeled Workbook 5A, 5b, and 5C - the
> PDF actually only contains 5A and 5B - I do not find a lab 5C - and thus I
> find no PSTN GK scenarios.  Am I going to the correct section?
>
>
>
> Thanks
> Scott
>
>> Date: Sun, 14 Mar 2010 11:17:56 -0500
>> Subject: Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK
>> From: cciet...@gmail.com
>> To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com
>>
>> Yes it does esp lab 5c
>>
>> On 3/13/10, scott carruthers  wrote:
>> >
>> > In some CCIE V2 labs Proctorlabs PSTN router was configured as GK so
>> > that we
>> > could practice remote zone scenarios. I have not reviewed all of the new
>> > lab IP Expert labs - do any call for sending calls thru our own HQ GK to
>> > a
>> > remote zone outside of our direct control? Do any of the Proctorlabs
>> > pre-configs configure the PSTN router as a GK?
>> >
>> > Thanks
>> > Scott
>> >
>> > _
>> > Hotmail has tools for the New Busy. Search, chat and e-mail from your
>> > inbox.
>> > http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_1
>>
>> --
>> Sent from my mobile device
>>
>> www.ccietalk.com
>   
> _
> Hotmail: Trusted email with powerful SPAM protection.
> http://clk.atdmt.com/GBL/go/210850553/direct/01/

-- 
Sent from my mobile device

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

Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK

[scott carruthers]

Is the lab you are referring to - lab 5C - in CCIE Voice Workbook V6.0 Volume 
1?  When I download the section labeled Workbook 5A, 5b, and 5C - the PDF 
actually only contains 5A and 5B - I do not find a lab 5C - and thus I find no 
PSTN GK scenarios.  Am I going to the correct section?

 

Thanks
Scott
 
> Date: Sun, 14 Mar 2010 11:17:56 -0500
> Subject: Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK
> From: cciet...@gmail.com
> To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com
> 
> Yes it does esp lab 5c
> 
> On 3/13/10, scott carruthers  wrote:
> >
> > In some CCIE V2 labs Proctorlabs PSTN router was configured as GK so that we
> > could practice remote zone scenarios. I have not reviewed all of the new
> > lab IP Expert labs - do any call for sending calls thru our own HQ GK to a
> > remote zone outside of our direct control? Do any of the Proctorlabs
> > pre-configs configure the PSTN router as a GK?
> >
> > Thanks
> > Scott
> > 
> > _
> > Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox.
> > http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_1
> 
> -- 
> Sent from my mobile device
> 
> www.ccietalk.com
  
_
Hotmail: Trusted email with powerful SPAM protection.
http://clk.atdmt.com/GBL/go/210850553/direct/01/___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK

[CCIETalk.com]

Yes it does esp lab 5c

On 3/13/10, scott carruthers  wrote:
>
> In some CCIE V2 labs Proctorlabs PSTN router was configured as GK so that we
> could practice remote zone scenarios.  I have not reviewed all of the new
> lab IP Expert labs - do any call for sending calls thru our own HQ GK to a
> remote zone outside of our direct control?  Do any of the Proctorlabs
> pre-configs configure the PSTN router as a GK?
>
> Thanks
> Scott
>   
> _
> Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox.
> http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_1

-- 
Sent from my mobile device

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

[OSL | CCIE_Voice] QoS Calculation Value for L2 MLPoFR

[Berry, Matthew J.]

Working on Vol 1 Lab 10A, Question 10.4

The Proctor Guide calculates L2 MLPoFR as 9 bytes per packet.  However, the QoS 
SRND defines the following on page 1-15:
- PPP = 12 bytes
- MLP = 13 bytes
- FR = 4 bytes
- FR with FRF.12 = 8 bytes

None of those match up.  Why did IPexpert chose 9 bytes per packet?

Matthew Berry

Digital Footprint:
Twitter: ciscovoiceguru
Skype: ciscovoiceguru
1st Lab Attempt: Aug 16th, 2010
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] UCCX License Location in Proctorlabs

[Arun Kumar]

send email to support and they will mail you.

On Sun, Mar 14, 2010 at 8:02 PM, Mike Brooks <2xcci...@gmail.com> wrote:

> I am trying to perform the UCCX integration in Proctorlabs but I am unable
> to find the licence file on the UCCX (10.10.210.5) server.
>
> Is it on the UCCX server or do I have to get it from somewhere else ?
>
> I preloaded lab 2a volume 1.  Perhaps its on a different VMWare instance of
> the UCCX.
>
> Regards,
>
> Mike Brooks
> CCIE# 16027 (R&S)
>
> ___
> 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] UCCX License Location in Proctorlabs

[Tanner Ezell]

You should not need the license file on an already installed UCCX server

On Sun, Mar 14, 2010 at 9:32 AM, Mike Brooks <2xcci...@gmail.com> wrote:
> I am trying to perform the UCCX integration in Proctorlabs but I am unable
> to find the licence file on the UCCX (10.10.210.5) server.
>
> Is it on the UCCX server or do I have to get it from somewhere else ?
>
> I preloaded lab 2a volume 1.  Perhaps its on a different VMWare instance of
> the UCCX.
>
> Regards,
>
> Mike Brooks
> CCIE# 16027 (R&S)
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>



-- 
Cheers,

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

[OSL | CCIE_Voice] UCCX License Location in Proctorlabs

[Mike Brooks]

I am trying to perform the UCCX integration in Proctorlabs but I am unable
to find the licence file on the UCCX (10.10.210.5) server.

Is it on the UCCX server or do I have to get it from somewhere else ?

I preloaded lab 2a volume 1.  Perhaps its on a different VMWare instance of
the UCCX.

Regards,

Mike Brooks
CCIE# 16027 (R&S)
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com