Re: [OSL | CCIE_Voice] Alerting and connected name through IPIP gateway.

[Voice Noob]

I never saw my original e-mail when I asked this question. When you send the
first e-mail to this group do you not see your own e-mail?

On Dec 7, 2007 8:38 AM, senthil natarajan <[EMAIL PROTECTED]> wrote:

> Configs of CME please? Did you try configuring the label and the caller
> mask on the CME side?
>
>
> On Dec 6, 2007 1:54 PM, Voice Noob <[EMAIL PROTECTED]> wrote:
>
> > Sending this again.
> >
> >
> > On Dec 5, 2007 8:48 AM, Voice Noob <[EMAIL PROTECTED]> wrote:
> >
> > >
> > >
> > > I have my CCM site connected to my CME site by and IPIPGW. On the CCM
> > > side I am sending the calls via H.323 to an ICT which is the IPIPGW.
> > > The calls are being received by the CME side as SIP translated by the
> > > IPIPGW. When I place a call from the HQ phone to my CME phone I do not get
> > > any caller Id info on the HQ phone. I do get the calling name and number 
> > > on
> > > the CME side, the side that was called. Even after we are connected I 
> > > still
> > > do not get any information on the phone that placed the call.
> > >
> > >
> > > If I reverse this and place the call from the CME side it is the same
> > > situation. The phone that places the call does not have any information
> > > other than the 4 digit number but the phone receiving the call on the CCM
> > > side has caller name and number. I have everything set to allow in my ICT
> > > and SIP trunk.
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] Alerting and connected name through IPIPgateway.

[Vik Malhi]

You are trying to use Connected Party Name Presentation (CONP). This is only
supported on internal and QSIG calls.
 

Vik Malhi
CCIE Voice Instructor / Developer - IPexpert, Inc.
CCIE Voice #13890 CCSI #31584
URL: http://www.IPexpert.com  
Toll Free: +1.866.225.8064
International: +1.810.326.1444


 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voice Noob
Sent: Friday, December 07, 2007 10:11 AM
To: senthil natarajan; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Alerting and connected name through
IPIPgateway.


Label does not have anything to do with calling name. It is for cosmetic
purposes only.
Yes I did put a name command under the ephone-dn. You could also see that
the CME side is setup correctly by they second paragraph. I do not have my
CME config anymore. I plan on setting it back up next week. 
 
If I reverse this and place the call from the CME side it is the same
situation. The phone that places the call does not have any information
other than the 4 digit number but the phone receiving the call on the CCM
side has caller name and number. I have everything set to allow in my ICT
and SIP trunk. 


On Dec 7, 2007 10:08 AM, Voice Noob <[EMAIL PROTECTED]> wrote:


I never saw my original e-mail when I asked this question. When you send the
first e-mail to this group do you not see your own e-mail? 


On Dec 7, 2007 8:38 AM, senthil natarajan <[EMAIL PROTECTED]> wrote:


Configs of CME please? Did you try configuring the label and the caller mask
on the CME side? 


On Dec 6, 2007 1:54 PM, Voice Noob <[EMAIL PROTECTED]> wrote:


Sending this again. 


On Dec 5, 2007 8:48 AM, Voice Noob <[EMAIL PROTECTED]> wrote:


  

I have my CCM site connected to my CME site by and IPIPGW. On the CCM side I
am sending the calls via H.323 to an ICT which is the IPIPGW. The calls are
being received by the CME side as SIP translated by the IPIPGW. When I place
a call from the HQ phone to my CME phone I do not get any caller Id info on
the HQ phone. I do get the calling name and number on the CME side, the side
that was called. Even after we are connected I still do not get any
information on the phone that placed the call. 




If I reverse this and place the call from the CME side it is the same
situation. The phone that places the call does not have any information
other than the 4 digit number but the phone receiving the call on the CCM
side has caller name and number. I have everything set to allow in my ICT
and SIP trunk. 

Re: [OSL | CCIE_Voice] CUE, call forward busy is not working via PSTN Line.

[Mark Snow]

So putting a Voice TR on the Voice port itself would in fact be a good  
idea if we wanted to do away with the dialplan pattern command  
altogether (which is what we recommend doing).
And the config could (you can do this many ways - this is only one)  
look like this (note: this config is specific to the dialplan in our  
WB's for the BR2 CME site) :

!
voice translation-rule 10
 rule 1 /617…\(2…\)/ /\1/
!
voice translation-rule 20
 rule 1 /\(2...\)/ /617527\1/
!
voice translation-profile ANI
 translate calling 20
!
voice translation-profile DNIS
 translate called 10
!
voice-port 0/0/0:23
 translation-profile incoming DNIS
 translation-profile outgoing ANI
!
end
!


But in the case that Bala already has the dialplan pattern command in  
use - and the call has already come from the Voice-Port - it is not  
then again hitting the POTS Voice-Port before going into CUE - so we  
would not be able to affect the digit manipulation by putting a voice  
translation rule on the Voice port - it would have to go on the VoIP  
Dial-Peer as I mentioned.



Cheers,

Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: [EMAIL PROTECTED]

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On  
Demand and Audio Certification Training Tools for the Cisco CCIE R&S  
Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
CCIE Storage Lab Certifications.



On Dec 5, 2007, at 8:39 PM, Jasan Mundur wrote:


Apologies - Corrected config here.
isdn switch-type primary-ni
network-clock-participate wic 1
network-clock-select 5 T1 0/1/0
!
controller T1 0/1/0
framing esf
linecode b8zs
pri-group timeslots 1-2,24

interface Serial0/1/0:23
 isdn outgoing display-ie

voice translation-profile VPIN
 translate called 1

voice translation-rule 1
  rule 1 /^408555/ //

voice-port 0/1/0:23
translation-profile in VPIN


 Jasan Mundur




On 12/5/07, Jasan Mundur <[EMAIL PROTECTED]> wrote:
Hi
My 2 cents.
Expanding on the Mark's method of using trans-profilehow about  
applying the TP on the voice-port itself. Sort of correcting at the  
source itself. ? ?  But ,correct me if I am wrong.


So based on our assumed config that Bala has 4085551003 it would  
look like this. voice-port 0/n/0:nn is created as soon as the time  
slots are defined at the controller T1.Just like the s 0/n/0:nn gets  
created.


!
!
isdn switch-type primary-ni
network-clock-participate wic 1
network-clock-select 5 T1 0/1/0
!
controller T1 0/1/0
framing esf
linecode b8zs
pri-group timeslots 1-2,24
interface Serial0/1/0:23

 isdn outgoing display-ie

voice translation-rule VPIN

  rule 1 /^408555/ //

voice-port 0/1/0:23

translation-profile in VPIN

## End of Message.

Jasan Mundur







On 12/5/07, Mark Snow <[EMAIL PROTECTED]> wrote:
Bala,


It will hit CUE using 10 digits as the RDNIS (redirected dnis) to  
which CUE will look for a mailbox with the number 4085551003 for  
phone 3 - but will not find one - since phone 3 has an extension in  
CUE of 1003 only.



This is due to the ANI or RDNIS being spoofed by the 'destination- 
pattern' command on the auto-generated Dial-Peer that was created by  
the router for phone 1003 (and every other ephone in that 1...  
range) the moment the 'dialplan-pattern' command was entered.

The dial-peer will look something like this:


!
dial-peer voice 20006 pots
 destination-pattern 4085551003
 huntstop
 progress_ind setup enable 3
 port 50/0/3
!


Do a 'sh telephony-service dial-peer' to see these dial-peers that  
do not show up by default in the 'sh run' config.



In order to correct this - you either:
1) Don't use the 'dialplan-pattern' command in telephony service and  
handle incoming 10 digit manipulation down to 4 and outgoing 4 digit  
manipulation up to 10 manually with Voice Translation Rules (this is  
my preference)

or
2) Perform the following voice-translation rules to overcome your  
problems you are having (I modified these to fit what it "looks"  
like your dialplan is set to with the 408-555-1xxx pattern - but YMMV)



!
voice translation-rule 1
  rule 1 /^408555\(1...\)/ /\1/
!
voice translation-profile VM
  translate calling 1
  translate called 1
  translate redirect-called 1
!
dial-peer voice 3600 voip
  translation-profile outgoing VM
  destination-pattern 1600
  session protocol sipv2
  dtmf-relay sip-notify
  session target ipv4:10.1.202.2
  codec g711ulaw
  no vad
!
!
dial-peer voice 3601 voip
  translation-profile outgoing VM
  destination-pattern 4085551600
  session protocol sipv2
  dtmf-relay sip-notify
  session target ipv4:10.1.202.2
  codec g711ulaw
  no vad
!




HTH Some,

Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: [EMAIL PROT

[OSL | CCIE_Voice] Documentation of CAS-Custom

[Jose Linero Welcker]

Hi:

I am doing the lab section 4 guide, I am configuring the R2 variations it 
asks, however it has been difficult for me to find the documentation in the 
documentation CD regarding the CAS-custom configuration specific for the 
12.4 IOS version for ISR. I have an idea to change the configuration based 
on CAS-custom, but due during the exam the documentation CD is the only help 
I will have I want to be sure where to find it. Please let me know if any of 
you knows how to get there.


Regards,

Jose

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.click-url.com/go/onm00200636ave/direct/01/

Re: [OSL | CCIE_Voice] Alerting and connected name through IPIP gateway.

[Voice Noob]

Label does not have anything to do with calling name. It is for cosmetic
purposes only.
Yes I did put a name command under the ephone-dn. You could also see that
the CME side is setup correctly by they second paragraph. I do not have my
CME config anymore. I plan on setting it back up next week.

If I reverse this and place the call from the CME side it is the same
situation. The phone that places the call does not have any information
other than the 4 digit number but the phone receiving the call on the CCM
side has caller name and number. I have everything set to allow in my ICT
and SIP trunk.

On Dec 7, 2007 10:08 AM, Voice Noob <[EMAIL PROTECTED]> wrote:

> I never saw my original e-mail when I asked this question. When you send
> the first e-mail to this group do you not see your own e-mail?
>
>
> On Dec 7, 2007 8:38 AM, senthil natarajan <[EMAIL PROTECTED]> wrote:
>
> > Configs of CME please? Did you try configuring the label and the caller
> > mask on the CME side?
> >
> >
> > On Dec 6, 2007 1:54 PM, Voice Noob <[EMAIL PROTECTED]> wrote:
> >
> > > Sending this again.
> > >
> > >
> > > On Dec 5, 2007 8:48 AM, Voice Noob <[EMAIL PROTECTED]> wrote:
> > >
> > > >
> > > >
> > > > I have my CCM site connected to my CME site by and IPIPGW. On the
> > > > CCM side I am sending the calls via H.323 to an ICT which is the
> > > > IPIPGW. The calls are being received by the CME side as SIP translated 
> > > > by
> > > > the IPIPGW. When I place a call from the HQ phone to my CME phone I do 
> > > > not
> > > > get any caller Id info on the HQ phone. I do get the calling name and 
> > > > number
> > > > on the CME side, the side that was called. Even after we are connected I
> > > > still do not get any information on the phone that placed the call.
> > > >
> > > >
> > > > If I reverse this and place the call from the CME side it is the
> > > > same situation. The phone that places the call does not have any 
> > > > information
> > > > other than the 4 digit number but the phone receiving the call on the 
> > > > CCM
> > > > side has caller name and number. I have everything set to allow in my 
> > > > ICT
> > > > and SIP trunk.
> > > >
> > >
> > >
> >
>

[OSL | CCIE_Voice] SIP trunk between CME and CCM 4.1

[Balamurugan Singaram]

Hi,
   
  I am trying a topology sip trunk, with ccm 4.1 and CME, CME source ip address 
- 172.16.2.100, from CME IPPhone I am able to reach CCM IPPhone, but from CCM 
IPPhone to CME I am not able to reach CME IPPhone.
   
  I am having route pattern in CCM to CME as siptrunk as gateway, 
   
  In SIP Trunk is configured as follows in CCM:
   
  Sip trunk destination address - 172.16.2.100, 
  Enable the "media termination point required", 
  but media resource group list is "[none]", in ccm side.  I am not configured 
any transcoder in CCM side. Region is g711ulaw.
   
   In CME side transcoder is up.
   
   
  Could please let me know what I missing from CCM to CME call routing via SIP 
trunk.
   
  Thank you,
  Bala
   

 Send instant messages to your online friends http://uk.messenger.yahoo.com 

Re: [OSL | CCIE_Voice] MOH with Mutlicast streaming from BR1 section21.24

[Vik Malhi]

I don't have a router handy right now...to speed things up can you let me
know if are you trying to source from the flash of the BR1 router or simply
multicast from the MOH server on CallManager ? 


Vik Malhi
CCIE Voice Instructor / Developer - IPexpert, Inc. 
CCIE Voice #13890 CCSI #31584
URL: http://www.IPexpert.com 
Toll Free: +1.866.225.8064
International: +1.810.326.1444 

-Original Message-
From: Ahmed Nawar [mailto:[EMAIL PROTECTED] 
Sent: Friday, December 07, 2007 3:15 PM
To: [EMAIL PROTECTED]
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MOH with Mutlicast streaming from BR1
section21.24

Vik, I realy can't make it to work. Could u pls post all the configuration
needed for this task to be accomplished with succses.

With Regards


Ahmed Mohammed Nawar
Networking Specialist
IBM Integrated Communications Services
Cisco IP Communications Support Specialist Cisco IP Telephony Operations
Specialist CCNA , CCNP,  CCVP , CCIE Voice Written

IBM-WTC  ,  Egypt Branch
Building  C - 10
Pyramids Heights Office Park,
KM.22 -Cairo - Alex. Desert Road,
P.O. Box 166 El-Ahram
Giza, Egypt.

Mobile:(20-10) 1552657
Tel.:  (202) 3536 2 536 Ext. 1120
DID:  (202) 3536 1120
Fax:  (202) 3536 2 505
Email: [EMAIL PROTECTED]


   
 "Vik Malhi"   
 <[EMAIL PROTECTED] 
 com>   To 
   Ahmed Nawar/Egypt/[EMAIL PROTECTED], 
   
 03/12/2007 10:27  
 ص  cc 
   
   Subject 
 Please respond to RE: [OSL | CCIE_Voice] MOH with 
 <[EMAIL PROTECTED] Mutlicast streaming from BR1
   com>section21.24
   
   
   
   
   
   




You must test by holding a call coming into the BR1 gateway since you cannot
listen to phones connected to the BR1 switch. Do not forget the "ccm-manager
music-on-hold" command (followed by mgcp/no mgcp).


Vik Malhi
CCIE Voice Instructor / Developer - IPexpert, Inc.
CCIE Voice #13890 CCSI #31584
URL: http://www.IPexpert.com
Toll Free: +1.866.225.8064
International: +1.810.326.1444

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ahmed Nawar
Sent: Saturday, December 01, 2007 3:46 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MOH with Mutlicast streaming from BR1
section21.24


I did all it said in the book but i didnt succeed to hear any moh. is there
any mode details that not written on that section answer.

[OSL | CCIE_Voice] DNIS-Digits

[Jose Linero Welcker]

Hi:

If we need to make the setup of the call 3 seconds quicker than default in 
R2 we can use the cas-custom command dnis-digits according to its 
explanation:


If the AS5x00 doesn’t know the number of DNIS digits beforehand, it has to 
rely on a timeout mechanism (3 seconds) to detect the end of DNIS.


Configuring max will speed up the call   set-up time by 3 seconds

Due the carrier instructs that they will send us 10 digits, I did the 
configuration dnis-digits min 1 max 10, but in the proctor guide I found the 
answer is dnis-digits min 3 max 11, I don´t understand why 3 as minimum and 
11 as maximum, does the values 1 as minimum and 10 as maximum accomplish the 
task required?.


Thanks in advance.

Regards,

Jose

_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

Re: [OSL | CCIE_Voice] DNIS-Digits

[Voice Noob]

I had problems understanding the same thing. I setup it up back in my lab
and had my telco router send me 10 digits. If I use max 11 it does not speed
anything up but if I set it to max 10 it speeds the call setup time.

On Dec 7, 2007 10:47 AM, Jose Linero Welcker <
[EMAIL PROTECTED]> wrote:

> Hi:
>
> If we need to make the setup of the call 3 seconds quicker than default in
> R2 we can use the cas-custom command dnis-digits according to its
> explanation:
>
> If the AS5x00 doesn't know the number of DNIS digits beforehand, it has to
> rely on a timeout mechanism (3 seconds) to detect the end of DNIS.
>
> Configuring max will speed up the call   set-up time by 3 seconds
>
> Due the carrier instructs that they will send us 10 digits, I did the
> configuration dnis-digits min 1 max 10, but in the proctor guide I found
> the
> answer is dnis-digits min 3 max 11, I don´t understand why 3 as minimum
> and
> 11 as maximum, does the values 1 as minimum and 10 as maximum accomplish
> the
> task required?.
>
> Thanks in advance.
>
> Regards,
>
> Jose
>
> _
> Express yourself instantly with MSN Messenger! Download today it's FREE!
> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
>
>

Re: [OSL | CCIE_Voice] MOH with Mutlicast streaming from BR1 section21.24

[Ahmed Nawar]

Vik, I realy can't make it to work. Could u pls post all the configuration
needed for this task to be accomplished with succses.

With Regards


Ahmed Mohammed Nawar
Networking Specialist
IBM Integrated Communications Services
Cisco IP Communications Support Specialist
Cisco IP Telephony Operations Specialist
CCNA , CCNP,  CCVP , CCIE Voice Written

IBM-WTC  ,  Egypt Branch
Building  C - 10
Pyramids Heights Office Park,
KM.22 -Cairo - Alex. Desert Road,
P.O. Box 166 El-Ahram
Giza, Egypt.

Mobile:(20-10) 1552657
Tel.:  (202) 3536 2 536 Ext. 1120
DID:  (202) 3536 1120
Fax:  (202) 3536 2 505
Email: [EMAIL PROTECTED]


   
 "Vik Malhi"   
 <[EMAIL PROTECTED] 
 com>   To 
   Ahmed Nawar/Egypt/[EMAIL PROTECTED], 
   
 03/12/2007 10:27  
 ص  cc 
   
   Subject 
 Please respond to RE: [OSL | CCIE_Voice] MOH with 
 <[EMAIL PROTECTED] Mutlicast streaming from BR1
   com>section21.24
   
   
   
   
   
   




You must test by holding a call coming into the BR1 gateway since you
cannot
listen to phones connected to the BR1 switch. Do not forget the
"ccm-manager
music-on-hold" command (followed by mgcp/no mgcp).


Vik Malhi
CCIE Voice Instructor / Developer - IPexpert, Inc.
CCIE Voice #13890 CCSI #31584
URL: http://www.IPexpert.com
Toll Free: +1.866.225.8064
International: +1.810.326.1444

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ahmed Nawar
Sent: Saturday, December 01, 2007 3:46 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MOH with Mutlicast streaming from BR1
section21.24


I did all it said in the book but i didnt succeed to hear any moh. is there
any mode details that not written on that section answer.

Re: [OSL | CCIE_Voice] DNIS-Digits

[Vik Malhi]

The DNIS-DIGITS command speeds up the setup time for inbound calls FROM the
PSTN. Given that proctorlabs are sending 10 digits all of the time min 10
max 10 would be good too. This does not affect outbound calls.
 

Vik Malhi
CCIE Voice Instructor / Developer - IPexpert, Inc.
CCIE Voice #13890 CCSI #31584
URL: http://www.IPexpert.com  
Toll Free: +1.866.225.8064
International: +1.810.326.1444


 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voice Noob
Sent: Friday, December 07, 2007 1:55 PM
To: Jose Linero Welcker
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] DNIS-Digits


I had problems understanding the same thing. I setup it up back in my lab
and had my telco router send me 10 digits. If I use max 11 it does not speed
anything up but if I set it to max 10 it speeds the call setup time. 


On Dec 7, 2007 10:47 AM, Jose Linero Welcker
<[EMAIL PROTECTED]> wrote:


Hi:

If we need to make the setup of the call 3 seconds quicker than default in
R2 we can use the cas-custom command dnis-digits according to its 
explanation:

If the AS5x00 doesn't know the number of DNIS digits beforehand, it has to
rely on a timeout mechanism (3 seconds) to detect the end of DNIS.

Configuring max will speed up the call   set-up time by 3 seconds 

Due the carrier instructs that they will send us 10 digits, I did the
configuration dnis-digits min 1 max 10, but in the proctor guide I found the
answer is dnis-digits min 3 max 11, I don´t understand why 3 as minimum and 
11 as maximum, does the values 1 as minimum and 10 as maximum accomplish the
task required?.

Thanks in advance.

Regards,

Jose


_
Express yourself instantly with MSN Messenger! Download today it's FREE!
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/