[OSL | CCIE_Voice] CME - direct incoming call from PSTN

[Terrence Ovi]

Hi Guys,

Is there any way to direct specific incoming call from PSTN to a
specific dial-peer range number on CME using COR? - 3 CMEs, only 1 cme
connected to PSTN.

for example, public phone number 7771234 that originate from PSTN only
allow to ring dial-peer range 7771000-7771005 on CME-A then public
phone number 7771235, only allow to ring dial-peer range
7771006-7771010 on CME-C.

please advice

Thanks in advance
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Re: [OSL | CCIE_Voice] WB 1 LAB 5C General question

[Rogers Ochieng]

Nope, you'll probably see it the early next day in the morning on email
telling you that you have either passed or failed :)

We need the 7 digit ANI to be met for 911 calls coming out of BR1 not to
lose that one point


-Original Message-
From: Randall Crumm [mailto:randall.cr...@harmonicinc.com] 
Sent: Monday, May 24, 2010 9:48 AM
To: r.ochi...@mfient.com
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] WB 1 LAB 5C General question

OK,
So how often does the proctor come by to verify? Would he come by and verify
all of lab 5 steps and score each step? 

Thanks,
Randall



hieng [mailto:r.ochi...@mfient.com] 
Sent: Sunday, May 23, 2010 11:34 PM
To: Randall Crumm
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] WB 1 LAB 5C General question

Then break say question 5.3 requirement that when calling 911 it should be 7
digits ANI? You'll get zero point there.

The target is to ensure that all the requirements spelt out in the question
are met at the end of all those configurations you can do.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm
Sent: Monday, May 24, 2010 8:57 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] WB 1 LAB 5C General question

HI,
In lab 5c we first start out br1 with 7 digit ani, then it moves to 10 and
we have to add a lot of translation patterns to make the ani 10 digits.

Why can we just adjust the calling transformation pattern to 10 digit? Can
we just change the calling transformation pattern in the real lab?

Thanks,
Randall

 

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Sunday, May 23, 2010 9:00 AM
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Subject: CCIE_Voice Digest, Vol 51, Issue 129

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Today's Topics:

   1. Re: Not getting "PLUS" on my Phones (Ashar Siddiqui)
   2. CME background image 7961 (Ashar Siddiqui)


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Message: 1
Date: Sun, 23 May 2010 15:29:55 +0100
From: Ashar Siddiqui 
Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones
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Cc: ccie_voice@onlinestudylist.com
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Message: 2
Date: Sun, 23 May 2010 16:19:06 +0100
From: Ashar Siddiqui 
Subject: [OSL | CCIE_Voice] CME background image 7961
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Re: [OSL | CCIE_Voice] WB 1 LAB 5C General question

[Randall Crumm]

OK,
So how often does the proctor come by to verify? Would he come by and verify 
all of lab 5 steps and score each step? 

Thanks,
Randall



hieng [mailto:r.ochi...@mfient.com] 
Sent: Sunday, May 23, 2010 11:34 PM
To: Randall Crumm
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] WB 1 LAB 5C General question

Then break say question 5.3 requirement that when calling 911 it should be 7
digits ANI? You'll get zero point there.

The target is to ensure that all the requirements spelt out in the question
are met at the end of all those configurations you can do.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm
Sent: Monday, May 24, 2010 8:57 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] WB 1 LAB 5C General question

HI,
In lab 5c we first start out br1 with 7 digit ani, then it moves to 10 and
we have to add a lot of translation patterns to make the ani 10 digits.

Why can we just adjust the calling transformation pattern to 10 digit? Can
we just change the calling transformation pattern in the real lab?

Thanks,
Randall

 

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Sunday, May 23, 2010 9:00 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 51, Issue 129

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Today's Topics:

   1. Re: Not getting "PLUS" on my Phones (Ashar Siddiqui)
   2. CME background image 7961 (Ashar Siddiqui)


--

Message: 1
Date: Sun, 23 May 2010 15:29:55 +0100
From: Ashar Siddiqui 
Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones
To: Ehab Salem 
Cc: ccie_voice@onlinestudylist.com
Message-ID: <4bf93be3.8070...@gmail.com>
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Message: 2
Date: Sun, 23 May 2010 16:19:06 +0100
From: Ashar Siddiqui 
Subject: [OSL | CCIE_Voice] CME background image 7961
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Re: [OSL | CCIE_Voice] WB 1 LAB 5C General question

[Rogers Ochieng]

Then break say question 5.3 requirement that when calling 911 it should be 7
digits ANI? You'll get zero point there.

The target is to ensure that all the requirements spelt out in the question
are met at the end of all those configurations you can do.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm
Sent: Monday, May 24, 2010 8:57 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] WB 1 LAB 5C General question

HI,
In lab 5c we first start out br1 with 7 digit ani, then it moves to 10 and
we have to add a lot of translation patterns to make the ani 10 digits.

Why can we just adjust the calling transformation pattern to 10 digit? Can
we just change the calling transformation pattern in the real lab?

Thanks,
Randall

 

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Sunday, May 23, 2010 9:00 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 51, Issue 129

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Today's Topics:

   1. Re: Not getting "PLUS" on my Phones (Ashar Siddiqui)
   2. CME background image 7961 (Ashar Siddiqui)


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Message: 1
Date: Sun, 23 May 2010 15:29:55 +0100
From: Ashar Siddiqui 
Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones
To: Ehab Salem 
Cc: ccie_voice@onlinestudylist.com
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Message: 2
Date: Sun, 23 May 2010 16:19:06 +0100
From: Ashar Siddiqui 
Subject: [OSL | CCIE_Voice] CME background image 7961
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[OSL | CCIE_Voice] WB 1 LAB 5C General question

[Randall Crumm]

HI,
In lab 5c we first start out br1 with 7 digit ani, then it moves to 10 and we 
have to add a lot of translation patterns to make the ani 10 digits.

Why can we just adjust the calling transformation pattern to 10 digit? Can we 
just change the calling transformation pattern in the real lab?

Thanks,
Randall

 

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
ccie_voice-requ...@onlinestudylist.com
Sent: Sunday, May 23, 2010 9:00 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 51, Issue 129

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Today's Topics:

   1. Re: Not getting "PLUS" on my Phones (Ashar Siddiqui)
   2. CME background image 7961 (Ashar Siddiqui)


--

Message: 1
Date: Sun, 23 May 2010 15:29:55 +0100
From: Ashar Siddiqui 
Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones
To: Ehab Salem 
Cc: ccie_voice@onlinestudylist.com
Message-ID: <4bf93be3.8070...@gmail.com>
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From: Ashar Siddiqui 
Subject: [OSL | CCIE_Voice] CME background image 7961
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[OSL | CCIE_Voice] EM CUCME Problem

[kerboute kerboute]

Hi guys,

I have an issue with EM on CME BR2, I've created the logout profile and 
assign it to the br2 phone2 but when i press the button service I've got 
"No services Configured", is there any restriction due to the firmware 
of IP phones ??

Any Idea?

note: ip http server already configured

Thank you
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Re: [OSL | CCIE_Voice] Direct Call Parl Problem

[kerboute kerboute]

Amy,

Thank you, It's working now by entering 80 then press BLF button, You're 
right :)



On 05/23/2010 10:23 PM, Amy Ryan wrote:
> When you are attempting to retrieve the call via the BLF button, which way
> are you conducting this?
>
> 1.  Directly pressing the BLF button.
> 2.  Entering the prefix (80) then pressing the BLF button.
>
> Answer is #2.  :-)
>
> HTH,
> Amy
>
> ---
> Amy Ryan ­ CCIE #24677 (Voice)
> Technical Instructor - IPexpert, Inc.
> Mailto: ar...@ipexpert.com
> Telephone: +1.810.326.1444
> Live Assistance, Please visit: www.ipexpert.com/chat
> 
> eFax: +1.810.454.0130
>
> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
> Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
> CCIE (R&S, Voice, Security&  Service Provider) certification(s) with
> training locations throughout the United States, Europe, South Asia and
> Australia. Be sure to visit our online communities at
> www.ipexpert.com/communities   and our
> public website at www.ipexpert.com
>
>
>
>
>> From: kerboute kerboute
>> Date: Sun, 23 May 2010 22:16:04 +0100
>> To: CCIE Voice Maillist
>> Subject: [OSL | CCIE_Voice] Direct Call Parl Problem
>>
>> Hi guys,
>>
>> I'm working on lab8A Vol1, I've configured the direct call park number
>> 8555 with a prefix of 80, also I've configured the call park BLF on the
>> phone button template.
>> When I transfer a call to the direct call park number 8555 I can monitor
>> the status on both phone for BLF and I can retrieve the call if I dial
>> 808555, however I can't retrieve call by pressing the BLF button I've
>> got "Park Slot Unavailable".
>>
>>
>> Any Idea?
>>
>> Regards
>> Naoufal
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>  
>
>
>

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Re: [OSL | CCIE_Voice] Direct Call Parl Problem

[Amy Ryan]

When you are attempting to retrieve the call via the BLF button, which way
are you conducting this?

1.  Directly pressing the BLF button.
2.  Entering the prefix (80) then pressing the BLF button.

Answer is #2.  :-)

HTH, 
Amy

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

eFax: +1.810.454.0130

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



> From: kerboute kerboute 
> Date: Sun, 23 May 2010 22:16:04 +0100
> To: CCIE Voice Maillist 
> Subject: [OSL | CCIE_Voice] Direct Call Parl Problem
> 
> Hi guys,
> 
> I'm working on lab8A Vol1, I've configured the direct call park number
> 8555 with a prefix of 80, also I've configured the call park BLF on the
> phone button template.
> When I transfer a call to the direct call park number 8555 I can monitor
> the status on both phone for BLF and I can retrieve the call if I dial
> 808555, however I can't retrieve call by pressing the BLF button I've
> got "Park Slot Unavailable".
> 
> 
> Any Idea?
> 
> Regards
> Naoufal
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com


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[OSL | CCIE_Voice] Direct Call Parl Problem

[kerboute kerboute]

Hi guys,

I'm working on lab8A Vol1, I've configured the direct call park number 
8555 with a prefix of 80, also I've configured the call park BLF on the 
phone button template.
When I transfer a call to the direct call park number 8555 I can monitor 
the status on both phone for BLF and I can retrieve the call if I dial 
808555, however I can't retrieve call by pressing the BLF button I've 
got "Park Slot Unavailable".


Any Idea?

Regards
Naoufal
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Re: [OSL | CCIE_Voice] CME background image 7961

[Ashar Siddiqui]

Yes I was missing that. It's all working now :)

Ash>

Mohammed Al-Assadi wrote:

  Did
you add the command:
  
tftp-server: flash::Desktops/320x196x4/[Images
name]
  
  Date: Sun, 23 May 2010 16:19:06 +0100
From: siddas...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME background image 7961
  
  Has anyone tried this?
  
I have copied the files List.xml, Small.png and Large.png for my 7961
phone from tftp server to flash:Desktops/320x196x4/ but still I am not
getting the image on my phone.
Even though at the router it says that it is looking for the file:
  
R3#copy tftp://10.10.210.5/List.xml flash:Desktops/320x196x4/List.xml
Destination filename [Desktops/320x196x4/List.xml]?
%Warning:There is a file already existing with this name
Do you want to over write? [confirm]
Accessing tftp://10.10.210.5/List.xml...
Loading List.xml from 10.10.210.5 (via Serial0/1/0:0.1): !
[OK - 152 bytes]
  
152 bytes copied in 0.720 secs (211 bytes/sec)
R3#
R3#
R3#
R3#
R3#
May 23 15:14:52.135: TFTP: Looking for Desktops/320x196x4/List.xml
R3#
  
  
Any clue? i have reset the phones as well. Is there any command we need
under telephony-service or ephones?
  
Thanks,
Ash>
   
  The New Busy is not the old busy. Search, chat and e-mail from
your inbox. Get started.



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Re: [OSL | CCIE_Voice] CME background image 7961

[Mohammed Al-Assadi]

Did you add the command:

tftp-server: flash::Desktops/320x196x4/[Images name]

Date: Sun, 23 May 2010 16:19:06 +0100
From: siddas...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME background image 7961








Has anyone tried this?



I have copied the files List.xml, Small.png and Large.png for my 7961
phone from tftp server to flash:Desktops/320x196x4/ but still I am not
getting the image on my phone.

Even though at the router it says that it is looking for the file:



R3#copy tftp://10.10.210.5/List.xml flash:Desktops/320x196x4/List.xml

Destination filename [Desktops/320x196x4/List.xml]?

%Warning:There is a file already existing with this name

Do you want to over write? [confirm]

Accessing tftp://10.10.210.5/List.xml...

Loading List.xml from 10.10.210.5 (via Serial0/1/0:0.1): !

[OK - 152 bytes]



152 bytes copied in 0.720 secs (211 bytes/sec)

R3#

R3#

R3#

R3#

R3#

May 23 15:14:52.135: TFTP: Looking for Desktops/320x196x4/List.xml

R3#





Any clue? i have reset the phones as well. Is there any command we need
under telephony-service or ephones?



Thanks,

Ash>

  
_
The New Busy is not the old busy. Search, chat and e-mail from your inbox.
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[OSL | CCIE_Voice] CME background image 7961

[Ashar Siddiqui]

Has anyone tried this?

I have copied the files List.xml, Small.png and Large.png for my 7961
phone from tftp server to flash:Desktops/320x196x4/ but still I am not
getting the image on my phone.
Even though at the router it says that it is looking for the file:

R3#copy tftp://10.10.210.5/List.xml flash:Desktops/320x196x4/List.xml
Destination filename [Desktops/320x196x4/List.xml]?
%Warning:There is a file already existing with this name
Do you want to over write? [confirm]
Accessing tftp://10.10.210.5/List.xml...
Loading List.xml from 10.10.210.5 (via Serial0/1/0:0.1): !
[OK - 152 bytes]

152 bytes copied in 0.720 secs (211 bytes/sec)
R3#
R3#
R3#
R3#
R3#
May 23 15:14:52.135: TFTP: Looking for Desktops/320x196x4/List.xml
R3#


Any clue? i have reset the phones as well. Is there any command we need
under telephony-service or ephones?

Thanks,
Ash>



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Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

[Ashar Siddiqui]

Thanks man!

Solved! I prefixed + for unknown type at GW in CUCM..I actually had
this in my mind and the reason I didn't try that because I wanted to
solve it using translation rules...why? because the same problem I have
calling from Site B to Site C (CME-H323). I am not getting + when
dialing from Site B to site C...I mean I am getting at the bottom of
the phone as From +16178631001 but at the top of site C phone it is
still From SBPH1 (16178631001).

Is this something to do with CME?

Thanks,
Ash


Ehab Salem wrote:

  
  
  

  
  
  You
can prefix plus to the incoming calling number from Gateway
page in CUCM…
   
   
  Ehab
M. Salem
  
   
  
  
  From:
Ashar Siddiqui
[mailto:siddas...@gmail.com] 
  Sent: Sunday, May 23, 2010 4:29 PM
  To: Wael Agina
  Cc: ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my
Phones
  
  
   
  I am glad you mentioned those rules...I have
tried all those
rules before exactly...no joy...this is why I wrote in my last email
that I
tried all my translation rule skills..  :)
  
  
  
voice translation-rule 99
 rule 1 /^34\(.*\)/ /+34\1/ type any unknown plan any unknown
 rule 2 // /+/
 rule 3 // /+/ type any unknown plan any unknown
 rule 4 /^34/ /+\0/
  
  
Even after all this...
  
R2#
May 23 12:27:59.440: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 
callref
= 0x00AB
    Bearer Capability i = 0x9090A2
   
Standard = CCITT
   
Transfer Capability = 3.1kHz Audio
   
Transfer Mode = Circuit
   
Transfer Rate = 64 kbit/s
    Channel ID i = 0xA98381
   
Exclusive, Channel 1
    Progress Ind i = 0x8583 -
Origination address is non-ISDN
    Display i = 'SCPH1'
    Calling Party Number i = 0x0080,
'3432143001'
   
Plan:Unknown, Type:Unknown
    Called Party Number i = 0x80,
'6178631001'
   
Plan:Unknown, Type:Unknown
May 23
R2# 12:27:59.440: ISDN Se0/0/0:23 Q931: Received SETUP  callref =
0x80AB
callID = 0x0019 switch = primary-ni interface = User
May 23 12:27:59.460: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 
callref = 0x80AB
    Channel ID i = 0xA98381
   
Exclusive, Channel 1
May 23 12:27:59.588: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 
callref = 0x80AB
R2#
May 23 12:28:01.972: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 
callref = 0x00AB
    Cause i = 0x8290 - Normal call
clearing
May 23 12:28:01.976: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 
callref = 0x80AB
May 23 12:28:01.988: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd =
8 
callref = 0x00AB
  
  
Thanks,
Ash>
  
Wael Agina wrote: 
  
  Try This and keep us updated
  
1-
voice translation-rule 99
 rule 1 // /+\0/
  
2-
If above working then make it specific for 34* numbers
voice translation-rule 99
 rule 1 /^34/ /+\0/ 
  
3- Last resort try num-exp  ===  this will affect both direction
calls and any calling passing the router !!!
  num-exp 3432143... +3432143...
  
  
  !
Thanks and Best Regards,
Wael Agina
  
   
  




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Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

[Ehab Salem]

You can prefix plus to the incoming calling number from Gateway page in
CUCM.

 

 

Ehab M. Salem

 

From: Ashar Siddiqui [mailto:siddas...@gmail.com] 
Sent: Sunday, May 23, 2010 4:29 PM
To: Wael Agina
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

 

I am glad you mentioned those rules...I have tried all those rules before
exactly...no joy...this is why I wrote in my last email that I tried all my
translation rule skills..  :)



voice translation-rule 99
 rule 1 /^34\(.*\)/ /+34\1/ type any unknown plan any unknown
 rule 2 // /+/
 rule 3 // /+/ type any unknown plan any unknown
 rule 4 /^34/ /+\0/


Even after all this...

R2#
May 23 12:27:59.440: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref =
0x00AB
Bearer Capability i = 0x9090A2
Standard = CCITT
Transfer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Progress Ind i = 0x8583 - Origination address is non-ISDN
Display i = 'SCPH1'
Calling Party Number i = 0x0080, '3432143001'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '6178631001'
Plan:Unknown, Type:Unknown
May 23
R2# 12:27:59.440: ISDN Se0/0/0:23 Q931: Received SETUP  callref = 0x80AB
callID = 0x0019 switch = primary-ni interface = User
May 23 12:27:59.460: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref =
0x80AB
Channel ID i = 0xA98381
Exclusive, Channel 1
May 23 12:27:59.588: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8  callref =
0x80AB
R2#
May 23 12:28:01.972: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8  callref
= 0x00AB
Cause i = 0x8290 - Normal call clearing
May 23 12:28:01.976: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8  callref =
0x80AB
May 23 12:28:01.988: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x00AB


Thanks,
Ash>

Wael Agina wrote: 

Try This and keep us updated

1-
voice translation-rule 99
 rule 1 // /+\0/

2-
If above working then make it specific for 34* numbers
voice translation-rule 99
 rule 1 /^34/ /+\0/ 

3- Last resort try num-exp  ===  this will affect both direction calls and
any calling passing the router !!!
num-exp 3432143... +3432143...


!
Thanks and Best Regards,
Wael Agina

 

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Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

[Ashar Siddiqui]

I am glad you mentioned those rules...I have tried all those rules
before exactly...no joy...this is why I wrote in my last email that I
tried all my translation rule skills..  :)



voice translation-rule 99
 rule 1 /^34\(.*\)/ /+34\1/ type any unknown plan any unknown
 rule 2 // /+/
 rule 3 // /+/ type any unknown plan any unknown
 rule 4 /^34/ /+\0/


Even after all this...

R2#
May 23 12:27:59.440: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 
callref = 0x00AB
    Bearer Capability i = 0x9090A2
    Standard = CCITT
    Transfer Capability = 3.1kHz Audio
    Transfer Mode = Circuit
    Transfer Rate = 64 kbit/s
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Progress Ind i = 0x8583 - Origination address is non-ISDN
    Display i = 'SCPH1'
    Calling Party Number i = 0x0080, '3432143001'
    Plan:Unknown, Type:Unknown
    Called Party Number i = 0x80, '6178631001'
    Plan:Unknown, Type:Unknown
May 23
R2# 12:27:59.440: ISDN Se0/0/0:23 Q931: Received SETUP  callref =
0x80AB callID = 0x0019 switch = primary-ni interface = User
May 23 12:27:59.460: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 
callref = 0x80AB
    Channel ID i = 0xA98381
    Exclusive, Channel 1
May 23 12:27:59.588: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 
callref = 0x80AB
R2#
May 23 12:28:01.972: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 
callref = 0x00AB
    Cause i = 0x8290 - Normal call clearing
May 23 12:28:01.976: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 
callref = 0x80AB
May 23 12:28:01.988: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd =
8  callref = 0x00AB


Thanks,
Ash>

Wael Agina wrote:

  Try This and keep us updated
  
1-
voice translation-rule 99
 rule 1 // /+\0/
  
2-
If above working then make it specific for 34* numbers
voice translation-rule 99
 rule 1 /^34/ /+\0/ 
  
3- Last resort try num-exp  ===  this will affect both direction calls
and any calling passing the router !!!
  num-exp 3432143... +3432143...
  
  
  !
Thanks and Best Regards,
Wael Agina
  




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Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

[Wael Agina]

Try This and keep us updated

1-
voice translation-rule 99
 rule 1 // /+\0/

2-
If above working then make it specific for 34* numbers
voice translation-rule 99
 rule 1 /^34/ /+\0/

3- Last resort try num-exp  ===  this will affect both direction calls and
any calling passing the router !!!
*num-exp 3432143... +**3432143...


* !
Thanks and Best Regards,
Wael Agina
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[OSL | CCIE_Voice] Not getting "PLUS" on my Phones

[Ashar Siddiqui]

Hi,

Having an strange issue...

Cannot make + appear when calling thru pstn from Site C to Site B or
from Site B to Site C. Site B is h323 and Site C is h323 (CME).
Used all my translation rule skills...no joy.

When call is routing from Site C, I can see Calling party number with
Plus at Site C gateway.

PSTN router is stripping +  ..look at this from PSTN router:

May 22 01:53:35.472: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 
callref = 0x008A
    Sending Complete
    Bearer Capability i = 0x8090A3
    Standard = CCITT
    Transfer Capability = Speech
    Transfer Mode = Circuit
    Transfer Rate = 64 kbit/s
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Progress Ind i = 0x8183 - Origination address is non-ISDN
    Display i = 'SCPH1'
    Calling Party Number i = 0x1180, '+3432143001'
    Plan:ISDN, Type:International
    Called Party Number i = 0x91, '0016178631001'
    Plan:ISDN, Type:International
May 22 01:53:35.492: ISDN Se0/3/1:23 Q931: Applying typeplan for
sw-type 0xD is 0x0 0x0, Calling num 85224044001
May 22 01:53:35.496: ISDN Se0/3/1:23 Q931: Applying typeplan for
sw-type 0xD is 0x0 0x0, Called num 9723033001
May 22 01:53:35.496: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 
callref = 0x808A
    Channel ID i = 0xA98381
    Exclusive, Channel 1
May 22 01:53:35.496: ISDN Se0/3/1:23 Q931: TX -> SETUP pd = 8 
callref = 0x009E
    Bearer Capability i = 0x9090A2
    Standard = CCITT
    Transfer Capability = 3.1kHz Audio
    Transfer Mode = Circuit
    Transfer Rate = 64 kbit/s
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Progress Ind i = 0x8583 - Origination address is non-ISDN
    Display i = 'SCPH1'
    Calling Party Number i = 0x0080, '3432143001'
    Plan:Unknown, Type:Unknown
    Called Party Number i = 0x80, '6178631001'
    Plan:Unknown, Type:Unknown
May 22 01:53:35.536: ISDN Se0/3/1:23 Q931: RX <- CALL_PROC pd = 8 
callref = 0x809E
    Channel ID i = 0xA98381
    Exclusive, Channel 1
May 22 01:53:35.668: ISDN Se0/3/1:23 Q931: RX <- ALERTING pd = 8 
callref = 0x809E
May 22 01:53:35.680: ISDN Se0/2/0:15 Q931: TX -> ALERTING pd = 8 
callref = 0x808A
    Progress Ind i = 0x8088 - In-band info or appropriate now
available


I made several rules at site B to add + to incoming calling number but
its still not showing the effing plus

voice translation-rule 1
 rule 1 /^617863\(1...$\)/ /\1/

voice translation-rule 99
 rule 1 /^34\(.*\)/ /+34\1/ type any unknown plan any unknown
!
!
voice translation-profile DNIS
 translate calling 99
 translate called 1

voice-port 0/0/0:23
 translation-profile incoming DNIS


When making calls to PSTN from site C, I am getting + perfect fine.
Am I hiting a bug or something? why I cannot see + on my Site B
phones...Why it's not even matching the translation rule 99 - any help
would be much appreciated.

Thanks,
Ash>




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Re: [OSL | CCIE_Voice] CME Globalization ?

[Angel Perez]

hi, it´s a cme 7 version issue, please read this:

 

https://learningnetwork.cisco.com/message/5

 

hth
 


Date: Sat, 22 May 2010 14:17:57 -0400
From: 2xcci...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Globalization ?


Is it possible to globalize a "calling" number inbound to a SCCP CME phone ?
 
On BR2 phone 1, I want the calling number to show up as +12123945001.  
Currently it just shows up on the screen as 12123945001.  In the bottom left of 
the screen it displays "From: +12123945001", but is it possible for it to be on 
the top of the screen and show up in the missed calls directory in the 
globalized format ?
 
By the way in the debug isdn, I am seeing the calling number in globalized 
format:
 
Display i = 'HQ PH1' 
Calling Party Number i = 0x0081, '+12123945001' 
Plan:Unknown, Type:Unknown 
Called Party Number i = 0x91, '32143001' 
Plan:ISDN, Type:International
 
I know that an H323 gateway strips the plus sign, but can I add it back as the 
calling number gets delivered to the phone ?  I can't find a way to do it.
 
Thanks,
Mike  
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Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links

[Graham Hopkins]

Would appear that the issue is with ip cef per destination load sharing. The 
two links are load shared but rsvp attempts to always use the same link between 
the two loopbacks of the gateways

so traffic to 30.30.30.30 ( BR1 loopback) is load-shared

HQ-GW#sh ip cef 30.30.30.30 internal
30.30.30.30/32, epoch 0, RIB[I], refcount 5, per-destination sharing
  sources: RIB
  feature space:
   IPRM: 0x00038000
  ifnums:
   Serial1/0.1(16): 10.10.10.2
   Serial1/0.3(18): 10.10.10.10
  path 68452BE8, path list 6845191C, share 0/1, type attached nexthop, for IPv4
  nexthop 10.10.10.2 Serial1/0.1, adjacency IP adj out of Serial1/0.1 66A19B40
  path 68452CD0, path list 6845191C, share 1/1, type attached nexthop, for IPv4
  nexthop 10.10.10.10 Serial1/0.3, adjacency IP adj out of Serial1/0.3 67CB9E80
  output chain:
loadinfo 669D93E4, per-session, 2 choices, flags 0003, 6 locks
flags: Per-session, for-rx-IPv4
16 hash buckets
  < 0 > IP adj out of Serial1/0.1 66A19B40
  < 1 > IP adj out of Serial1/0.3 67CB9E80
  < 2 > IP adj out of Serial1/0.1 66A19B40
  < 3 > IP adj out of Serial1/0.3 67CB9E80
  < 4 > IP adj out of Serial1/0.1 66A19B40
  < 5 > IP adj out of Serial1/0.3 67CB9E80
  < 6 > IP adj out of Serial1/0.1 66A19B40
  < 7 > IP adj out of Serial1/0.3 67CB9E80
  < 8 > IP adj out of Serial1/0.1 66A19B40 

but the specific route used by rsvp always takes one route

May 23 09:19:11.847: RSVP-API: 11.11.11.11_29550->30.30.30.30_24948[0.0.0.0]: 
Processing PATH request [id=0x67AC3BD0]...

HQ-GW# sh ip cef exact-route 11.11.11.11 30.30.30.30
11.11.11.11 -> 30.30.30.30 => IP adj out of Serial1/0.1

Other traffic - for example the rdp streams between phones on 192.168.60.x (HQ) 
and 192.168.50.x (BR1)  does load share but never gets the chance !

HQ-GW# sh ip cef exact-route 192.168.60.2 192.168.50.10
192.168.60.2 -> 192.168.50.10 => IP adj out of Serial1/0.1
HQ-GW# sh ip cef exact-route 192.168.60.4 192.168.50.10
192.168.60.4 -> 192.168.50.10 => IP adj out of Serial1/0.3   




Regards

Graham Hopkins



On 22 May 2010, at 19:19, Graham Hopkins wrote:

> I think we mean the same thing although my use of the term call setup is 
> probably not a good one - when the request for bandwidth for call setup is 
> made with G729 then 40 kbps is requested - worse case bandwidth for a 10ms 
> sample rate. After call established this drops to 24kbps leaving 40kbps 
> available for the bandwidth request of the second call. 
> 
> I think this is more likely to be a routing issue as the router makes no 
> attempt to request bandwidth on the second link
> 
> 
> Gateways and debug follow - btw the configs have some legacy stuff from other 
> testing - this is the dynamips  version rather than the physical one so are 
> 7200s
> 
> 
> HQ-GW#sh run
> Building configuration...
> 
> Current configuration : 4800 bytes
> !
> ! Last configuration change at 15:55:33 BST Sat May 22 2010
> ! NVRAM config last updated at 15:56:06 BST Sat May 22 2010
> !
> upgrade fpd auto
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname HQ-GW
> !
> boot-start-marker
> boot-end-marker
> !
> logging message-counter syslog
> !
> no aaa new-model
> clock timezone GMT 0
> clock summer-time BST recurring last Sun Mar 1:00 last Sun Oct 1:00
> clock calendar-valid
> ip source-route
> ip cef
> !
> !
> ip dhcp excluded-address 192.168.60.1 192.168.60.9
> ip dhcp excluded-address 192.168.60.21 192.168.60.254
> !
> ip dhcp pool PHONES
>   network 192.168.60.0 255.255.255.0
>   default-router 192.168.60.1
>   option 150 ip 192.168.60.2
> !
> !
> no ip domain lookup
> no ipv6 cef
> !
> multilink bundle-name authenticated
> !
> !
> !
> voice service voip
> fax protocol cisco
> h323
>  ras rrq dynamic prefixes
> !
> !
> !
> voice class codec 1
> codec preference 1 g729r8
> codec preference 2 g711ulaw
> !
> !
> archive
> log config
>  hidekeys
> !
> !
> class-map match-all VOIP
> match ip dscp ef
> class-map match-any CONTROL
> match ip dscp cs3  af31
> class-map match-any AutoQoS-VoIP-RTP-Trust
> match ip dscp ef
> class-map match-any AutoQoS-VoIP-Control-Trust
> match ip dscp cs3
> match ip dscp af31
> !
> !
> policy-map AutoQoS-Policy-Trust
> class AutoQoS-VoIP-RTP-Trust
>priority percent 70
> class AutoQoS-VoIP-Control-Trust
>bandwidth percent 5
> class class-default
>fair-queue
> policy-map WAN
> class VOIP
>priority percent 25
>   compress header ip rtp
> class CONTROL
>bandwidth percent 30
> class class-default
>fair-queue
> !
> !
> !
> !
> !
> interface Loopback0
> ip address 11.11.11.11 255.255.255.255
> h323-gateway voip interface
> h323-gateway voip id GK1 ipaddr 20.20.20.20 1719
> h323-gateway voip tech-prefix 1#
> h323-gateway voip bind srcaddr 11.11.11.11
> !
> interface FastEthernet0/0
> ip address 192.168.60.1 255.255.255.0
> duplex half
> speed 100
> !
> interface FastEthernet0/1
> description to GK