Re: [OSL | CCIE_Voice] physical lab before exam?

[Darby Weaver]

You should be fine.

But if you feel you must touch the gear - find a local Cisco Office and see
if they don't have the gear you need to work with.




On Thu, Apr 23, 2009 at 11:19 PM, junk cisco  wrote:

> hi,
>
> I need your advice before taking the lab in US
>
> I have studied 8 months with  Remote lab from Internetworkexpert , do i
> need to rent real lab (physical lab)  before sitting the lab exam ?   I am
> planning to take the lab in May.
> Need suggestion please?  If yes, where can rent for good price ?   I am in
> west of Canada.
>
> Or anybody pass just using the remote lab ?  Pls share your experience if
> do not have equipment in office
>
>
> Thks
>

[OSL | CCIE_Voice] physical lab before exam?

[junk cisco]

hi,

I need your advice before taking the lab in US

I have studied 8 months with  Remote lab from Internetworkexpert , do i need
to rent real lab (physical lab)  before sitting the lab exam ?   I am
planning to take the lab in May.
Need suggestion please?  If yes, where can rent for good price ?   I am in
west of Canada.

Or anybody pass just using the remote lab ?  Pls share your experience if do
not have equipment in office


Thks

Re: [OSL | CCIE_Voice] SIP Trunk Issue

[Larry Hadrava]

Norma:
Would you like to share with the group what the issue was. I'm sure that you
are not the first or last to have this happen :-)

Thanks
Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com


On Thu, Apr 23, 2009 at 7:03 PM, Norma Exel  wrote:

> nevermind.  figured it out.  i apologize as this is too far out of the
> scope of the lab study.  thanks anyway.  -N
>
> - Original Message -
> From: "Norma Exel"
> To: "Norma Exel" , ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] SIP Trunk Issue
> Date: Wed, 1 Apr 2009 23:30:48 -0500
>
> Here's the ASCII drawing of the call flow...
>
> BR2 phone-SIP Voip Dial-peer-UCM TrunkUCMIP Phone
> (sccp)Call Transfer(no consult)(aka blind transfer)UCMIP Phone
> (sccp)
>
> - Original Message -
> From: "Norma Exel"
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] SIP Trunk Issue
> Date: Wed, 1 Apr 2009 23:16:14 -0500
>
> Here's a good one!  When I make a call over a sip trunk and a sccp phone
> picks up and performs a blind transfer to another sccp phone, there is no
> ringback heard on the calling phone.  no MTP or annunciator is configured on
> the UCM cluster.  Any idea where to start?  I suspect UCM is not providing
> the SIP leg appropriate "transferring" message when the sccp phone signals
> for a transfer.  My scenario doesn't affect H323 (inbound PRI) since
> ringback can be heard on a blind transfer when a call comes in on a PRI.
> One would believe that UCM should be able to provide the appropriate
> signaling back through either the H323 or SIP trunk to reflect a call
> transfer so that ringback would be heard with either trunk.  Anybody with
> insight to this would be greatly appreciated.
>
> Norma
>
> -- Be Yourself @ mail.com!
> Choose From 200+ Email Addresses
> Get a *Free* Account at www.mail.com !
>
>
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Re: [OSL | CCIE_Voice] SIP Trunk Issue

[Norma Exel]

nevermind.  figured it out.  i apologize as this is too far out of the
scope of the lab study.  thanks anyway.  -N

  - Original Message -
  From: "Norma Exel"
  To: "Norma Exel" , ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] SIP Trunk Issue
  Date: Wed, 1 Apr 2009 23:30:48 -0500

  Here's the ASCII drawing of the call flow... BR2 phone-SIP Voip
  Dial-peer-UCM TrunkUCMIP Phone (sccp)Call Transfer(no
  consult)(aka blind transfer)UCMIP Phone (sccp) 

- Original Message -
From: "Norma Exel"
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP Trunk Issue
Date: Wed, 1 Apr 2009 23:16:14 -0500

Here's a good one!  When I make a call over a sip trunk and a
sccp phone picks up and performs a blind transfer to another sccp
phone, there is no ringback heard on the calling phone.  no MTP
or annunciator is configured on the UCM cluster.  Any idea where
to start?  I suspect UCM is not providing the SIP leg appropriate
"transferring" message when the sccp phone signals for a
transfer.  My scenario doesn't affect H323 (inbound PRI) since
ringback can be heard on a blind transfer when a call comes in on
a PRI.  One would believe that UCM should be able to provide the
appropriate signaling back through either the H323 or SIP trunk
to reflect a call transfer so that ringback would be heard with
either trunk.  Anybody with insight to this would be greatly
appreciated. Norma
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  Get a Free Account at www.mail.com!

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Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked orNot

[Sergio Polizer]

Cliff, You are right!

If I press hold during a call to BR1 two sessions of hw-xcoder at HQ is 
allocated.

But, The MTP checkbox is UNchecked. I do not understanding how it are beeing 
requested.

I just have one trunk, so there is no option to be other trunk. I restarted the 
services and all the calls still be working with flash/transfer...that's crazy.

From: cl...@mcglamry.net
To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot
Date: Thu, 23 Apr 2009 16:52:51 -0400










I don't see how that could work without an MTP.  
Suggest you reset CCM services and retry.  I think you have it right, but 
you need the MTP for supplementary services.
 

  - Original Message - 
  From: 
  Sergio 
  Polizer 
  To: ccie_voice@onlinestudylist.com 
  ; cl...@mcglamry.net 
  Sent: Thursday, April 23, 2009 4:13 
  PM
  Subject: FW: [OSL | CCIE_Voice] ICT-GK 
  controlled - Should have MTP checked orNot
  
I found one solution:

For G729 calls between CME to HQ 
  and Br1 phones through a GK WITH supplementary services, I used:

- H225 
  GK Controlled (MTP not checked)
- With DP Br2 (G729 with all other regions) 
  and with HW xcoder  at HQ DP.



  
  From: spoli...@hotmail.com
To: cl...@mcglamry.net; ccie_voice@onlinestudylist.com
Subject: 
  RE: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
  orNot
Date: Thu, 23 Apr 2009 16:18:01 -0300


  

  Cliff, Thanks for your input. See the answers bellow. 
  

If you're getting a fast busy on the transfer, you probably have an 
  issue with your Region set up.  Things to look at:
A) No, It is when I 
  answer the call at BR1 phone.

   
  What codec is defined between the GKTrunk and the BR1 
  region?  If it's defined as G711, but you have limited bandwidth, the 
  call will fail.  
  A) G729, and the location for br1 is 24Kbps. 


  Do you have a hardware transcoder available to the Trunk 
  MRGL?  If you need that, and don't have it in the MRGL, the call will 
  fail.  
  I'd also put the hardware resources ABOVE the software 
  resources.  Then it should work with transfer and all.
A) Yes, Is the 
  HW transcoding from HQ. 

How should be the regions 
  configured for ICT_GK and the HW xcoder at HQ?

Sergio.

  
  From: cl...@mcglamry.net
To: spoli...@hotmail.com; 
  ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ICT-GK 
  controlled - Should have MTP checked orNot
Date: Thu, 23 Apr 2009 14:23:37 
  -0400


  

  If you're getting a fast busy on the transfer, you 
  probably have an issue with your Region set up.  Things to look 
  at:
   
  What codec is defined between the GKTrunk and the BR1 
  region?  If it's defined as G711, but you have limited bandwidth, the 
  call will fail.  
   
  Do you have a hardware transcoder available to the Trunk 
  MRGL?  If you need that, and don't have it in the MRGL, the call will 
  fail.  
  I'd also put the hardware resources ABOVE the software 
  resources.  Then it should work with transfer and all.
   
   
  
- Original Message - 
From: 
Sergio 
Polizer 
To: ccie_voice@onlinestudylist.com 

Sent: Thursday, April 23, 2009 2:00 
PM
Subject: [OSL | CCIE_Voice] ICT-GK 
controlled - Should have MTP checked orNot

HI,

I have a call from BR2 to HQ/BR1 in a ICT_GK with 
MTP checked to provide supplementary services like hold/transfer.

My 
ICT have a BR2 DP that speak G729 with all others and a HW transcoding that 
have HQ DP.

When I call from CME to HQ the call goes to HQ transcoder 
at G729 and connect a HQ phone at G711.

When I call to BR1, I have 
fast busy after answer de call. I also have a HW transcoding at Br1 with DP 
BR1. 

I do not undersdant why this call does not work.  What am 
I missing?


IF I uncheck MTP all the calls works but the transfer 
does not work. I wonder IF a good practice at real lab is always UNCHECK 
the 
MTP unless be required by the question.

Someone may clarify this to 
me?

Thanks in advance.

Sergio.



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Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked orNot

[Cliff McGlamry]

I don't see how that could work without an MTP.  Suggest you reset CCM services 
and retry.  I think you have it right, but you need the MTP for supplementary 
services.

  - Original Message - 
  From: Sergio Polizer 
  To: ccie_voice@onlinestudylist.com ; cl...@mcglamry.net 
  Sent: Thursday, April 23, 2009 4:13 PM
  Subject: FW: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot


  I found one solution:

  For G729 calls between CME to HQ and Br1 phones through a GK WITH 
supplementary services, I used:

  - H225 GK Controlled (MTP not checked)
  - With DP Br2 (G729 with all other regions) and with HW xcoder  at HQ DP.



--
  From: spoli...@hotmail.com
  To: cl...@mcglamry.net; ccie_voice@onlinestudylist.com
  Subject: RE: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot
  Date: Thu, 23 Apr 2009 16:18:01 -0300


  Cliff, Thanks for your input. See the answers bellow. 

  If you're getting a fast busy on the transfer, you probably have an issue 
with your Region set up.  Things to look at:
  A) No, It is when I answer the call at BR1 phone.


  What codec is defined between the GKTrunk and the BR1 region?  If it's 
defined as G711, but you have limited bandwidth, the call will fail.  
  A) G729, and the location for br1 is 24Kbps. 


  Do you have a hardware transcoder available to the Trunk MRGL?  If you need 
that, and don't have it in the MRGL, the call will fail.  
  I'd also put the hardware resources ABOVE the software resources.  Then it 
should work with transfer and all.
  A) Yes, Is the HW transcoding from HQ. 


  How should be the regions configured for ICT_GK and the HW xcoder at HQ?

  Sergio.

--
  From: cl...@mcglamry.net
  To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot
  Date: Thu, 23 Apr 2009 14:23:37 -0400


  If you're getting a fast busy on the transfer, you probably have an issue 
with your Region set up.  Things to look at:

  What codec is defined between the GKTrunk and the BR1 region?  If it's 
defined as G711, but you have limited bandwidth, the call will fail.  

  Do you have a hardware transcoder available to the Trunk MRGL?  If you need 
that, and don't have it in the MRGL, the call will fail.  
  I'd also put the hardware resources ABOVE the software resources.  Then it 
should work with transfer and all.


- Original Message - 
From: Sergio Polizer 
To: ccie_voice@onlinestudylist.com 
Sent: Thursday, April 23, 2009 2:00 PM
Subject: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot


HI,

I have a call from BR2 to HQ/BR1 in a ICT_GK with MTP checked to provide 
supplementary services like hold/transfer.

My ICT have a BR2 DP that speak G729 with all others and a HW transcoding 
that have HQ DP.

When I call from CME to HQ the call goes to HQ transcoder at G729 and 
connect a HQ phone at G711.

When I call to BR1, I have fast busy after answer de call. I also have a HW 
transcoding at Br1 with DP BR1. 

I do not undersdant why this call does not work.  What am I missing?


IF I uncheck MTP all the calls works but the transfer does not work. I 
wonder IF a good practice at real lab is always UNCHECK the MTP unless be 
required by the question.

Someone may clarify this to me?

Thanks in advance.

Sergio.



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agora. É grátis! 


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[OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ

[Jiahong - tobeccie Fang]

Two match dial-peers is right, we 1st one will be used if its status is up and 
remote end able to receive it, otherwise 2nd one kick in.

After dial-peer matched, codec/dtmf-relay come in. 

James

> Message: 1
> Date: Thu, 23 Apr 2009 10:24:22 -0700
> From: Nara Shikamaru 
> Subject: Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ
>   router
> To: Rick Grimes 
> Cc: "ccie_voice@onlinestudylist.com" ,
>   Jiahong - tobeccie Fang 
> Message-ID:
>   
> Content-Type: text/plain; charset="iso-8859-1"
> 
> It's set to 4 digits on the trunk.  That's not the problem here.
> 
> On Thu, Apr 23, 2009 at 10:14 AM, Rick Grimes 
> wrote:
> 
> >  Change your destination-pattern to 4 digits or make sure that significant
> > digits is set to 4 in your sip trunk.
> >
> >
> >
> >
> >
> >
> >
> > *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Nara Shikamaru
> > *Sent:* Thursday, April 23, 2009 11:26 AM
> > *To:* Jiahong - tobeccie Fang
> > *Cc:* ccie_voice@onlinestudylist.com
> > *Subject:* Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ
> > router
> >
> >
> >
> > James,
> >
> >  If there are two dial peers that are identical except for the
> > preference setting, is it normal for the debug to reflect matching on both?
> > I'm wondering if the problem is occuring before the dial-peer has an
> > oppurtunity to do anything with the call.  Codec and dtmf-relay settings are
> > relevant, I believe, in the final steps of the call staging set up, no?
> >
> >
> >
> >  When I configured my PSTN switch and had to work out dial peers to
> > forward traffic to my PSTN phones on another router (FXS ports), traffic
> > would not be forwarded until the dial peer match was resolved.  I will add
> > the syntax to the dial peers and let you know, just seems unusual based on
> > my understanding.
> >
> > 2009/4/23 Jiahong - tobeccie Fang 
> >
> > OK you match the outbound dial-peer properly. You may have problem with
> > dtmf-relay  or codec negotiation.
> >
> > Copy/paste your 'voice class codec 1' config. Also add 'dtmf-relay
> > sip-notify rtp-nte' in both dial-peers
> >
> > James F.
> >
> > > Message: 1
> > > Date: Wed, 22 Apr 2009 22:48:04 -0700
> > > From: Nara Shikamaru 
> > > Subject: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router
> > > To: "ccie_voice@onlinestudylist.com" 
> > > Message-ID:
> > > 
> > > Content-Type: text/plain; charset="iso-8859-1"
> >
> >
> > >
> > > Hello,
> > > I am working on inbound calling to the HQ site for HQ (internal ext
> > > 500X). Calling inbound from PSTN phone, the dial-peer debug traffic shows
> > > that both redundant dial-peers are being matched (5000 and 5001).
> > However,
> > > my impression is that the "preference" syntax should be the deciding
> > factor
> > > in which dial peer is used. As far as I can tell the inbound call is not
> > > being passed to CUCM's SIP trunk. From the dial-peer debug;
> > >
> > > *P1-HQ-VG#
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > > Calling Number=0113432141891, Called Number=2123945002,
> > > Voice-Interface=0x0,
> > > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > > Type=PEER_TYPE_VOICE,
> > > Peer Info Type=DIALPEER_INFO_SPEECH
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > > Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > > Calling Number=0113432141891, Called Number=2123945002,
> > > Voice-Interface=0x0,
> > > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > > Type=PEER_TYPE_VOICE,
> > > Peer Info Type=DIALPEER_INFO_SPEECH
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > > Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > > Calling Number=, Called Number=2123945002, Peer Info
> > > Type=DIALPEER_INFO_SPEECH
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > > Result=Success(0) after DP_MATCH_DEST
> > > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersMoreArg:
> > > Result=SUCCESS(0)
> > > List of Matched Outgoing Dial-peer(s):
> > > 1: Dial-peer Tag=5000
> > > 2: Dial-peer Tag=5001
> > > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > > Calling Number=2123945002, Called Number=2123945002, Peer Info
> > > Type=DIALPEER_INFO_SPEECH
> > > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > > Result=Success(0) after DP_MATCH_DEST
> > > Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> > > Result=SUCCESS(0)
> > > List of Matched Outgoing Dial-peer(s):

[OSL | CCIE_Voice] FW: ICT-GK controlled - Should have MTP checked orNot

[Sergio Polizer]

I found one solution:

For G729 calls between CME to HQ and Br1 phones through a GK WITH supplementary 
services, I used:

- H225 GK Controlled (MTP not checked)
- With DP Br2 (G729 with all other regions) and with HW xcoder  at HQ DP.


From: spoli...@hotmail.com
To: cl...@mcglamry.net; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot
Date: Thu, 23 Apr 2009 16:18:01 -0300








Cliff, Thanks for your input. See the answers bellow. 

If you're getting a fast busy on the transfer, you 
probably have an issue with your Region set up.  Things to look 
at:
A) No, It is when I answer the call at BR1 phone.

 
What codec is defined between the GKTrunk and the BR1 
region?  If it's defined as G711, but you have limited bandwidth, the call 
will fail.  
A) G729, and the location for br1 is 24Kbps. 


Do you have a hardware transcoder available to the Trunk 
MRGL?  If you need that, and don't have it in the MRGL, the call will 
fail.  
I'd also put the hardware resources ABOVE the software 
resources.  Then it should work with transfer and all.
A) Yes, Is the HW transcoding from HQ. 

How should be the regions configured for ICT_GK and the HW xcoder at HQ?

Sergio.
From: cl...@mcglamry.net
To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot
Date: Thu, 23 Apr 2009 14:23:37 -0400










If you're getting a fast busy on the transfer, you 
probably have an issue with your Region set up.  Things to look 
at:
 
What codec is defined between the GKTrunk and the BR1 
region?  If it's defined as G711, but you have limited bandwidth, the call 
will fail.  
 
Do you have a hardware transcoder available to the Trunk 
MRGL?  If you need that, and don't have it in the MRGL, the call will 
fail.  
I'd also put the hardware resources ABOVE the software 
resources.  Then it should work with transfer and all.
 
 

  - Original Message - 
  From: 
  Sergio 
  Polizer 
  To: ccie_voice@onlinestudylist.com 
  
  Sent: Thursday, April 23, 2009 2:00 
  PM
  Subject: [OSL | CCIE_Voice] ICT-GK 
  controlled - Should have MTP checked orNot
  
HI,

I have a call from BR2 to HQ/BR1 in a ICT_GK with 
  MTP checked to provide supplementary services like hold/transfer.

My 
  ICT have a BR2 DP that speak G729 with all others and a HW transcoding that 
  have HQ DP.

When I call from CME to HQ the call goes to HQ transcoder 
  at G729 and connect a HQ phone at G711.

When I call to BR1, I have fast 
  busy after answer de call. I also have a HW transcoding at Br1 with DP BR1. 
  

I do not undersdant why this call does not work.  What am I 
  missing?


IF I uncheck MTP all the calls works but the transfer does 
  not work. I wonder IF a good practice at real lab is always UNCHECK the MTP 
  unless be required by the question.

Someone may clarify this to 
  me?

Thanks in advance.

Sergio.


  
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Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked orNot

[Sergio Polizer]

Cliff, Thanks for your input. See the answers bellow. 

If you're getting a fast busy on the transfer, you 
probably have an issue with your Region set up.  Things to look 
at:
A) No, It is when I answer the call at BR1 phone.

 
What codec is defined between the GKTrunk and the BR1 
region?  If it's defined as G711, but you have limited bandwidth, the call 
will fail.  
A) G729, and the location for br1 is 24Kbps. 


Do you have a hardware transcoder available to the Trunk 
MRGL?  If you need that, and don't have it in the MRGL, the call will 
fail.  
I'd also put the hardware resources ABOVE the software 
resources.  Then it should work with transfer and all.
A) Yes, Is the HW transcoding from HQ. 

How should be the regions configured for ICT_GK and the HW xcoder at HQ?

Sergio.
From: cl...@mcglamry.net
To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked 
orNot
Date: Thu, 23 Apr 2009 14:23:37 -0400










If you're getting a fast busy on the transfer, you 
probably have an issue with your Region set up.  Things to look 
at:
 
What codec is defined between the GKTrunk and the BR1 
region?  If it's defined as G711, but you have limited bandwidth, the call 
will fail.  
 
Do you have a hardware transcoder available to the Trunk 
MRGL?  If you need that, and don't have it in the MRGL, the call will 
fail.  
I'd also put the hardware resources ABOVE the software 
resources.  Then it should work with transfer and all.
 
 

  - Original Message - 
  From: 
  Sergio 
  Polizer 
  To: ccie_voice@onlinestudylist.com 
  
  Sent: Thursday, April 23, 2009 2:00 
  PM
  Subject: [OSL | CCIE_Voice] ICT-GK 
  controlled - Should have MTP checked orNot
  
HI,

I have a call from BR2 to HQ/BR1 in a ICT_GK with 
  MTP checked to provide supplementary services like hold/transfer.

My 
  ICT have a BR2 DP that speak G729 with all others and a HW transcoding that 
  have HQ DP.

When I call from CME to HQ the call goes to HQ transcoder 
  at G729 and connect a HQ phone at G711.

When I call to BR1, I have fast 
  busy after answer de call. I also have a HW transcoding at Br1 with DP BR1. 
  

I do not undersdant why this call does not work.  What am I 
  missing?


IF I uncheck MTP all the calls works but the transfer does 
  not work. I wonder IF a good practice at real lab is always UNCHECK the MTP 
  unless be required by the question.

Someone may clarify this to 
  me?

Thanks in advance.

Sergio.


  
  Quer deixar seu Messenger turbinado de emoticons? Clique aqui e baixe agora. 
É grátis! 
_
Emoticons e Winks super diferentes para o Messenger. Baixe agora, é grátis!
http://specials.br.msn.com/ilovemessenger/pacotes.aspx

Re: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked orNot

[Cliff McGlamry]

If you're getting a fast busy on the transfer, you probably have an issue with 
your Region set up.  Things to look at:

What codec is defined between the GKTrunk and the BR1 region?  If it's defined 
as G711, but you have limited bandwidth, the call will fail.  

Do you have a hardware transcoder available to the Trunk MRGL?  If you need 
that, and don't have it in the MRGL, the call will fail.  
I'd also put the hardware resources ABOVE the software resources.  Then it 
should work with transfer and all.


  - Original Message - 
  From: Sergio Polizer 
  To: ccie_voice@onlinestudylist.com 
  Sent: Thursday, April 23, 2009 2:00 PM
  Subject: [OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked orNot


  HI,

  I have a call from BR2 to HQ/BR1 in a ICT_GK with MTP checked to provide 
supplementary services like hold/transfer.

  My ICT have a BR2 DP that speak G729 with all others and a HW transcoding 
that have HQ DP.

  When I call from CME to HQ the call goes to HQ transcoder at G729 and connect 
a HQ phone at G711.

  When I call to BR1, I have fast busy after answer de call. I also have a HW 
transcoding at Br1 with DP BR1. 

  I do not undersdant why this call does not work.  What am I missing?


  IF I uncheck MTP all the calls works but the transfer does not work. I wonder 
IF a good practice at real lab is always UNCHECK the MTP unless be required by 
the question.

  Someone may clarify this to me?

  Thanks in advance.

  Sergio.


--
  Quer deixar seu Messenger turbinado de emoticons? Clique aqui e baixe agora. 
É grátis! 

Re: [OSL | CCIE_Voice] Version 3 Lab equipment...

[Nara Shikamaru]

Why would you want to install CUC on a DL 380?  You can install all of the
platforms on VMware (Intel only right now, not AMD yet.)  I have CUCM, CRS,
CUC, and CUPS on a dell desktop with 4 gb ram.

On Tue, Apr 21, 2009 at 11:28 PM, Jonathan Charles wrote:

> They are cheap on Ebay...
>
> I have a crapload of phones, most are 41s and 61s... got a couple of 65s..
>
> I am just annoyed that I can't install CUC on a DL 380 G3 or G4...
>
>
> J
>
>
> On Tue, Apr 21, 2009 at 10:01 PM, Michael Ciarfello <
> mciarfe...@iplogic.com> wrote:
>
>> Hi everyone,
>>
>> Could also be x2 phones for ilbc and g722 codec.
>>
>> They could keep a couple x0 phones for basic SIP functionality and how to
>> deal with that (sip dial rules, etc.)
>>
>> I'm jealous you (Johnathan) has a 3750.  hehe
>>
>> 
>> From: ccie_voice-boun...@onlinestudylist.com [
>> ccie_voice-boun...@onlinestudylist.com] On Behalf Of WorkerBee [
>> cisco...@gmail.com]
>> Sent: Tuesday, April 21, 2009 8:00 PM
>> To: Chris Parker
>> Cc: ccie_voice@onlinestudylist.com
>> Subject: Re: [OSL | CCIE_Voice] Version 3 Lab equipment...
>>
>> Probably phones maybe upgraded to 7961 series "Type B" that support
>> more presence features compare to "Type A" phones such as 7940, 7960.
>>
>>
>>
>>
>> On Wed, Apr 22, 2009 at 1:50 AM, Chris Parker  wrote:
>> > I would say another HWIC-4ESW, and seven 79XX - phones 3 for HQ, 2 for
>> BR1
>> > and 2 for BR2
>> >
>> > Jonathan Charles wrote:
>> >>
>> >> Well, I have an NME-CUE, two 2821s, and a 2811, a 3750, HWIC-4ESW and a
>> >> pile of DL 380s... I am just curious what I am missing...
>> >>
>> >> On Mon, Apr 20, 2009 at 7:56 PM, Chris Parker > >> > wrote:
>> >>
>> >>
>> >>I think BR2 will also be an NM-ESW type device. Suppose it could
>> >>also be another 3750. I'm using HWIC-ESW for BR1/2 in my lab and a
>> >>3560 at HQ. Not sure why they are using 3800's per the blue print.
>> >>I think you can do everything with 2801/11. In my lab BR1/2 are
>> >>2801 and HQ is 2811. The only trick with using 2801 is memory. You
>> >>need at least 256MB for the 12.4(24)T IOS.
>> >>
>> >>Chris
>> >>
>> >>
>> >>Jonathan Charles wrote:
>> >>
>> >>OK, I have abandoned my attempts at the V2 lab (just not
>> >>enough time, and a little too retro for me...)
>> >>
>> >>So, I am curious about the topology...
>> >>
>> >>3750 at HQ, makes sense...
>> >>
>> >>4 port HWIC at BR1... ok...
>> >>
>> >>What is happening at BR2, switch-wise?
>> >>
>> >>Also... why are the routers so beefy at HQ and BR1? Are there
>> >>other modules in there we should be aware of?
>> >>
>> >>What am I missing? I have about 20 phones here..
>> >>
>> >>Also, when is IPExpert going to update their doco? I
>> >>downloaded some spartan labs... but there is not a lot of info
>> >>there... when the upgrade comes, how much? My lab is currently
>> >>scheduled for 27 JULY
>> >>
>> >>
>> >>
>> >>Jonathan
>> >>
>> >>
>> >>
>> >>
>> >
>> >
>>
>
>


-- 
-Shikamaru

[OSL | CCIE_Voice] ICT-GK controlled - Should have MTP checked or Not

[Sergio Polizer]

HI,

I have a call from BR2 to HQ/BR1 in a ICT_GK with MTP checked to provide 
supplementary services like hold/transfer.

My ICT have a BR2 DP that speak G729 with all others and a HW transcoding that 
have HQ DP.

When I call from CME to HQ the call goes to HQ transcoder at G729 and connect a 
HQ phone at G711.

When I call to BR1, I have fast busy after answer de call. I also have a HW 
transcoding at Br1 with DP BR1. 

I do not undersdant why this call does not work.  What am I missing?


IF I uncheck MTP all the calls works but the transfer does not work. I wonder 
IF a good practice at real lab is always UNCHECK the MTP unless be required by 
the question.

Someone may clarify this to me?

Thanks in advance.

Sergio.

_
Emoticons e Winks super diferentes para o Messenger. Baixe agora, é grátis!
http://specials.br.msn.com/ilovemessenger/pacotes.aspx

Re: [OSL | CCIE_Voice] VPIM problem

[Cliff McGlamry]

>From the trace, it would appear that the most likely issue is that the host 
>you have defined in the MX record doesn't exist.  

Within an MX record (in Microsoft DNS), there is a spot where you put in the 
host name.  This should be the fully qualified domain name of the host, and 
MUST be DNS resolvable (which means an A record for the host must exist).  

You can test this by opening a command prompt and type nslookup and hit return. 
 That will give you a > prompt.  You can type server followed by the ip address 
of the DNS server you want to query to point it at the DNS server you want to 
test.  

>
> server 74.7.1.66
Default Server:  [74.7.1.66]
Address:  74.7.1.66

Then set the type to MX so you can look for that specific record type:

> set type=mx

It will just return back to a > prompt.  Type in the name of the domain you 
want the MX record for:

> mcglamry.net
Server:  [74.7.1.66]
Address:  74.7.1.66

mcglamry.netMX preference = 10, mail exchanger = rocketmail.mcglamry.net
mcglamry.netMX preference = 15, mail exchanger = mail.mcglamry.net
rocketmail.mcglamry.net internet address = 74.7.1.67

If you look at the end of the first line, you'll see it returned the host name 
for the MX record with preference 10 as rocketmail.mcglamry.net.  It went a 
little farther and did a lookup for the A record and gives me the address in 
the last line.  But I could have done it manually too:

To get the A record: 

> set type=a

You'll just get a > prompt back.

> rocketmail.mcglamry.net
Server:  [74.7.1.66]
Address:  74.7.1.66

Name:rocketmail.mcglamry.net
Address:  74.7.1.67

Most likely, you fat fingered the host name in the MX record.  But this is how 
you can walk through and check it.

Cliff


  - Original Message - 
  From: jeremy co 
  To: ccie_voice@onlinestudylist.com 
  Sent: Thursday, April 23, 2009 12:56 PM
  Subject: [OSL | CCIE_Voice] VPIM problem


  Hi,

  I configured vpim, but seems cue cannot connect to unity

  com.cisco.aesop.smtp.SmtpService : findLocation: ret 1
  5410 01/20 21:36:29.167 netw dbug 1 com.cisco.aesop.smtp.SmtpService : 
findLocation: ret 1
  5410 01/20 21:36:29.168 netw dbug 1 com.cisco.aesop.smtp.SmtpService : 
remotesLocations size=2
  5410 01/20 21:36:29.169 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread : 
localDomain: cue.ccievoice.com
  5410 01/20 21:36:29.169 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread : 
First recipient: 1002002
  5410 01/20 21:36:29.171 netw dbug 1 com.cisco.aesop.smtp.SmtpService : 
getRemoteLocation: phone: null, domain: null, address: 1002002
  5410 01/20 21:36:29.172 netw dbug 1 com.cisco.aesop.smtp.SmtpService : 
getRemoteLocation: start 0, end 1
  5410 01/20 21:36:29.173 netw dbug 1 com.cisco.aesop.smtp.SmtpService : 
getRemoteLocation: matched rn.getLocationId(): 100
  5410 01/20 21:36:29.173 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread : 
Getting mail server addresses for: unity.ccie
  5410 01/20 21:36:29.175 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread : 
domain unity.ccie
  5410 01/20 21:36:29.175 netw dns 1 unity.ccie
  5410 01/20 21:36:29.176 netw dbug 1 com.cisco.aesop.smtp.DnsAgent : DnsAgent: 
Resolving unity.ccie
  5410 01/20 21:36:29.188 netw dns 2 MX:  priority: 10
  5410 01/20 21:36:29.189 netw dbug 1 com.cisco.aesop.smtp.DnsAgent : Found MX 
record:  priority: 10
  5410 01/20 21:36:29.190 netw dns 1 
  5410 01/20 21:36:29.190 netw dbug 1 com.cisco.aesop.smtp.DnsAgent : DnsAgent: 
Resolving 
  5410 01/20 21:36:29.197 netw dbug 1 com.cisco.aesop.smtp.DnsAgent : Unable to 
resolve



  network location id 100
   email domain unity.ccie
   name "unity"
   end location

  network location id 852
   email domain cue.ccievoice.com
   name "cue"
   end location

  network local location id 852


  I confirmed, smtp to unity.ccie MX port 25and I connected to it . so this 
address is resolvable, why cue cannot resolve it?


  Jeremy

Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router

[Nara Shikamaru]

It's set to 4 digits on the trunk.  That's not the problem here.

On Thu, Apr 23, 2009 at 10:14 AM, Rick Grimes wrote:

>  Change your destination-pattern to 4 digits or make sure that significant
> digits is set to 4 in your sip trunk.
>
>
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Nara Shikamaru
> *Sent:* Thursday, April 23, 2009 11:26 AM
> *To:* Jiahong - tobeccie Fang
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ
> router
>
>
>
> James,
>
>  If there are two dial peers that are identical except for the
> preference setting, is it normal for the debug to reflect matching on both?
> I'm wondering if the problem is occuring before the dial-peer has an
> oppurtunity to do anything with the call.  Codec and dtmf-relay settings are
> relevant, I believe, in the final steps of the call staging set up, no?
>
>
>
>  When I configured my PSTN switch and had to work out dial peers to
> forward traffic to my PSTN phones on another router (FXS ports), traffic
> would not be forwarded until the dial peer match was resolved.  I will add
> the syntax to the dial peers and let you know, just seems unusual based on
> my understanding.
>
> 2009/4/23 Jiahong - tobeccie Fang 
>
> OK you match the outbound dial-peer properly. You may have problem with
> dtmf-relay  or codec negotiation.
>
> Copy/paste your 'voice class codec 1' config. Also add 'dtmf-relay
> sip-notify rtp-nte' in both dial-peers
>
> James F.
>
> > Message: 1
> > Date: Wed, 22 Apr 2009 22:48:04 -0700
> > From: Nara Shikamaru 
> > Subject: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router
> > To: "ccie_voice@onlinestudylist.com" 
> > Message-ID:
> > 
> > Content-Type: text/plain; charset="iso-8859-1"
>
>
> >
> > Hello,
> > I am working on inbound calling to the HQ site for HQ (internal ext
> > 500X). Calling inbound from PSTN phone, the dial-peer debug traffic shows
> > that both redundant dial-peers are being matched (5000 and 5001).
> However,
> > my impression is that the "preference" syntax should be the deciding
> factor
> > in which dial peer is used. As far as I can tell the inbound call is not
> > being passed to CUCM's SIP trunk. From the dial-peer debug;
> >
> > *P1-HQ-VG#
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Calling Number=0113432141891, Called Number=2123945002,
> > Voice-Interface=0x0,
> > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > Type=PEER_TYPE_VOICE,
> > Peer Info Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Calling Number=0113432141891, Called Number=2123945002,
> > Voice-Interface=0x0,
> > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > Type=PEER_TYPE_VOICE,
> > Peer Info Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > Calling Number=, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > Result=Success(0) after DP_MATCH_DEST
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersMoreArg:
> > Result=SUCCESS(0)
> > List of Matched Outgoing Dial-peer(s):
> > 1: Dial-peer Tag=5000
> > 2: Dial-peer Tag=5001
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Calling Number=2123945002, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Result=Success(0) after DP_MATCH_DEST
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> > Result=SUCCESS(0)
> > List of Matched Outgoing Dial-peer(s):
> > 1: Dial-peer Tag=5000
> > 2: Dial-peer Tag=5001
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Calling Number=2123945002, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Result=Success(0) after DP_MATCH_DEST
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> > Result=SUCCESS(0)
> > List of Matched Outgoing Dial-peer(s):
> > 1: Dial-peer Tag=5000
> > 2: Dial-peer Tag=5001
> > Apr 22 22:37:14: //-1//DPM/dpAssociateIncomingPeerCore:
> > Calling Number=2123945002, Called Number=, Voice-Interface=0x0,
>

Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router

[Rick Grimes]

Change your destination-pattern to 4 digits or make sure that significant 
digits is set to 4 in your sip trunk.



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru
Sent: Thursday, April 23, 2009 11:26 AM
To: Jiahong - tobeccie Fang
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router

James,
 If there are two dial peers that are identical except for the preference 
setting, is it normal for the debug to reflect matching on both?  I'm wondering 
if the problem is occuring before the dial-peer has an oppurtunity to do 
anything with the call.  Codec and dtmf-relay settings are relevant, I believe, 
in the final steps of the call staging set up, no?

 When I configured my PSTN switch and had to work out dial peers to forward 
traffic to my PSTN phones on another router (FXS ports), traffic would not be 
forwarded until the dial peer match was resolved.  I will add the syntax to the 
dial peers and let you know, just seems unusual based on my understanding.
2009/4/23 Jiahong - tobeccie Fang mailto:mo...@hotmail.com>>
OK you match the outbound dial-peer properly. You may have problem with 
dtmf-relay  or codec negotiation.

Copy/paste your 'voice class codec 1' config. Also add 'dtmf-relay sip-notify 
rtp-nte' in both dial-peers

James F.

> Message: 1
> Date: Wed, 22 Apr 2009 22:48:04 -0700
> From: Nara Shikamaru mailto:shikam...@kagadis.com>>
> Subject: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router
> To: "ccie_voice@onlinestudylist.com" 
> mailto:ccie_voice@onlinestudylist.com>>
> Message-ID:
> mailto:a3c82292090448s47ecbe51r46047aaeb98d5...@mail.gmail.com>>
> Content-Type: text/plain; charset="iso-8859-1"

>
> Hello,
> I am working on inbound calling to the HQ site for HQ (internal ext
> 500X). Calling inbound from PSTN phone, the dial-peer debug traffic shows
> that both redundant dial-peers are being matched (5000 and 5001). However,
> my impression is that the "preference" syntax should be the deciding factor
> in which dial peer is used. As far as I can tell the inbound call is not
> being passed to CUCM's SIP trunk. From the dial-peer debug;
>
> *P1-HQ-VG#
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> Calling Number=0113432141891, Called Number=2123945002,
> Voice-Interface=0x0,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> Calling Number=0113432141891, Called Number=2123945002,
> Voice-Interface=0x0,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> Calling Number=, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=5000
> 2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> Calling Number=2123945002, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=5000
> 2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> Calling Number=2123945002, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=5000
> 2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1//DPM/dpAssociateIncomingPeerCore:
> Calling Number=2123945002, Called Number=, Voice-Interface=0x0,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/x

[OSL | CCIE_Voice] VPIM problem

[jeremy co]

Hi,

I configured vpim, but seems cue cannot connect to unity

com.cisco.aesop.smtp.SmtpService : findLocation: ret 1
5410 01/20 21:36:29.167 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
findLocation: ret 1
5410 01/20 21:36:29.168 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
remotesLocations size=2
5410 01/20 21:36:29.169 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread :
localDomain: cue.ccievoice.com
5410 01/20 21:36:29.169 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread :
First recipient: 1002002
5410 01/20 21:36:29.171 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
getRemoteLocation: phone: null, domain: null, address: 1002002
5410 01/20 21:36:29.172 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
getRemoteLocation: start 0, end 1
5410 01/20 21:36:29.173 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
getRemoteLocation: matched rn.getLocationId(): 100
5410 01/20 21:36:29.173 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread :
Getting mail server addresses for: unity.ccie
5410 01/20 21:36:29.175 netw dbug 1 com.cisco.aesop.smtp.SmtpSenderThread :
domain unity.ccie
5410 01/20 21:36:29.175 netw dns 1 unity.ccie
5410 01/20 21:36:29.176 netw dbug 1 com.cisco.aesop.smtp.DnsAgent :
DnsAgent: Resolving unity.ccie
5410 01/20 21:36:29.188 netw dns 2 MX:  priority: 10
5410 01/20 21:36:29.189 netw dbug 1 com.cisco.aesop.smtp.DnsAgent : Found MX
record:  priority: 10
5410 01/20 21:36:29.190 netw dns 1
5410 01/20 21:36:29.190 netw dbug 1 com.cisco.aesop.smtp.DnsAgent :
DnsAgent: Resolving
5410 01/20 21:36:29.197 netw dbug 1 com.cisco.aesop.smtp.DnsAgent : Unable
to resolve



network location id 100
 email domain unity.ccie
 name "unity"
 end location

network location id 852
 email domain cue.ccievoice.com
 name "cue"
 end location

network local location id 852


I confirmed, smtp to unity.ccie MX port 25and I connected to it . so this
address is resolvable, why cue cannot resolve it?


Jeremy

Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router

[Nara Shikamaru]

James,
 If there are two dial peers that are identical except for the
preference setting, is it normal for the debug to reflect matching on both?
I'm wondering if the problem is occuring before the dial-peer has an
oppurtunity to do anything with the call.  Codec and dtmf-relay settings are
relevant, I believe, in the final steps of the call staging set up, no?

 When I configured my PSTN switch and had to work out dial peers to
forward traffic to my PSTN phones on another router (FXS ports), traffic
would not be forwarded until the dial peer match was resolved.  I will add
the syntax to the dial peers and let you know, just seems unusual based on
my understanding.

2009/4/23 Jiahong - tobeccie Fang 

> OK you match the outbound dial-peer properly. You may have problem with
> dtmf-relay  or codec negotiation.
>
> Copy/paste your 'voice class codec 1' config. Also add 'dtmf-relay
> sip-notify rtp-nte' in both dial-peers
>
> James F.
>
> > Message: 1
> > Date: Wed, 22 Apr 2009 22:48:04 -0700
> > From: Nara Shikamaru 
> > Subject: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router
> > To: "ccie_voice@onlinestudylist.com" 
> > Message-ID:
> > 
> > Content-Type: text/plain; charset="iso-8859-1"
>
> >
> > Hello,
> > I am working on inbound calling to the HQ site for HQ (internal ext
> > 500X). Calling inbound from PSTN phone, the dial-peer debug traffic shows
> > that both redundant dial-peers are being matched (5000 and 5001).
> However,
> > my impression is that the "preference" syntax should be the deciding
> factor
> > in which dial peer is used. As far as I can tell the inbound call is not
> > being passed to CUCM's SIP trunk. From the dial-peer debug;
> >
> > *P1-HQ-VG#
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Calling Number=0113432141891, Called Number=2123945002,
> > Voice-Interface=0x0,
> > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > Type=PEER_TYPE_VOICE,
> > Peer Info Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Calling Number=0113432141891, Called Number=2123945002,
> > Voice-Interface=0x0,
> > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > Type=PEER_TYPE_VOICE,
> > Peer Info Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
> > Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > Calling Number=, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
> > Result=Success(0) after DP_MATCH_DEST
> > Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersMoreArg:
> > Result=SUCCESS(0)
> > List of Matched Outgoing Dial-peer(s):
> > 1: Dial-peer Tag=5000
> > 2: Dial-peer Tag=5001
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Calling Number=2123945002, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Result=Success(0) after DP_MATCH_DEST
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> > Result=SUCCESS(0)
> > List of Matched Outgoing Dial-peer(s):
> > 1: Dial-peer Tag=5000
> > 2: Dial-peer Tag=5001
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Calling Number=2123945002, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Result=Success(0) after DP_MATCH_DEST
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
> > Result=SUCCESS(0)
> > List of Matched Outgoing Dial-peer(s):
> > 1: Dial-peer Tag=5000
> > 2: Dial-peer Tag=5001
> > Apr 22 22:37:14: //-1//DPM/dpAssociateIncomingPeerCore:
> > Calling Number=2123945002, Called Number=, Voice-Interface=0x0,
> > Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> > Type=PEER_TYPE_VOICE,
> > Peer Info Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1//DPM/dpAssociateIncomingPeerCore:
> > Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=5000
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Calling Number=, Called Number=2123945002, Peer Info
> > Type=DIALPEER_INFO_SPEECH
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Match Rule=DP_MATCH_DEST; Called Number=2123945002
> > Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
> > Result=Success(0) afte

[OSL | CCIE_Voice] Digit Manipulation in Cisco UNITY

[Tech Guy]

BlankI have been googling this for sometime now, but no luck yet. I know that 
there is a digit manipulation file in UNITY server just like the num-exp in 
CME. The only difference is that, it is an editable flat file residing in one 
of the directories in UNITY server. 

For some reasons I cannot locate that file lately.

Tech Guy


<>

Re: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router

[Jiahong - tobeccie Fang]

OK you match the outbound dial-peer properly. You may have problem with 
dtmf-relay  or codec negotiation.

Copy/paste your 'voice class codec 1' config. Also add 'dtmf-relay sip-notify 
rtp-nte' in both dial-peers

James F.

> Message: 1
> Date: Wed, 22 Apr 2009 22:48:04 -0700
> From: Nara Shikamaru 
> Subject: [OSL | CCIE_Voice] v3 lab : SIP trunk/dial peers on HQ router
> To: "ccie_voice@onlinestudylist.com" 
> Message-ID:
>   
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hello,
>  I am working on inbound calling to the HQ site for HQ (internal ext
> 500X).  Calling inbound from PSTN phone, the dial-peer debug traffic shows
> that both redundant dial-peers are being matched (5000 and 5001).  However,
> my impression is that the "preference" syntax should be the deciding factor
> in which dial peer is used.  As far as I can tell the inbound call is not
> being passed to CUCM's SIP trunk.  From the dial-peer debug;
> 
> *P1-HQ-VG#
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
>Calling Number=0113432141891, Called Number=2123945002,
> Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
>Calling Number=0113432141891, Called Number=2123945002,
> Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=5000
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
>Calling Number=, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=5000
>  2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Calling Number=2123945002, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=5000
>  2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Calling Number=2123945002, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=5000
>  2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1//DPM/dpAssociateIncomingPeerCore:
>Calling Number=2123945002, Called Number=, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1//DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=5000
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Calling Number=, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1//DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=5000
>  2: Dial-peer Tag=5001
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
>Calling Number=, Called Number=2123945002, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=2123945002
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Apr 22 22:37:14: //-1/A018D0158081/DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=5000
>  2: Dial-peer Tag=5001*
> 
> 
> This 

Re: [OSL | CCIE_Voice] Question on VPIM Addressing in Unity

[Cliff McGlamry]

No, there isn't.

This is where VPIM and the exchange voice connector comes in.  There should be 
a user in Unity defined with the primary extension of 1001.  But no user named 
1001.

Cliff

  - Original Message - 
  From: Scott ODonnell 
  To: OSL Group 
  Sent: Thursday, April 23, 2009 9:03 AM
  Subject: [OSL | CCIE_Voice] Question on VPIM Addressing in Unity


  Just reading through some threads on VPIM problems.
  I wanted to confirm something.


  If I create a subscriber on UNITY , called H Phone1 with extension 1001, the 
underlying domain account is HPhone1 with a smtp address of hpho...@voip.lab 
(or whatever domain is configured).


  I just want to make sure this is right.


  Should there be some entry in Exchange for 1...@voip.lab?


  As I remember, the voice connector is suppose to manage the alias of 
extension to Exchange user .


  - Scott

Re: [OSL | CCIE_Voice] is there a phone remote control to test moh ios multicast br1 via vracks?

[zamuel del Toro]

 

Hi Vik, 

are there a reason to not allow a phone remote control(voip integrations, free 
on internet).?, that could be great.

you can try by your own. on my ccm work fine. do you know if is a port 
restriccion or something like that?. can you  suggest it to ip-expert?. with a 
phone remote control we can test everything  like we were in real lab.

thanks and reggards
 


Date: Wed, 22 Apr 2009 18:57:39 -0700
Subject: Re: [OSL | CCIE_Voice] is there a phone remote control to test moh ios 
multicast br1 via vracks?
From: vma...@ipexpert.com
To: saralilin2...@yahoo.co.jp; sdelto...@hotmail.com; 
ccie_voice@onlinestudylist.com

You could call in from the pstn and put the pstn caller on hold. The originator 
of the call could be an HQ or BR1 phone that calls via PSTN to BR2.

-- 
Vik Malhi – CCIE #13890, CCSI #31584 
Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444 
Fax: +1.810.454.0130 
Mailto: vma...@ipexpert.com


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








From: 
Date: Thu, 23 Apr 2009 09:01:37 +0900 (JST)
To: zamuel del Toro , OSL Group 

Subject: Re: [OSL | CCIE_Voice] is there a phone remote control to test moh ios 
multicast br1 via vracks?

i think you can just web into the phone ip, see if there is mcat traffic?

_
Color coding for safety: Windows Live Hotmail alerts you to suspicious email.
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[OSL | CCIE_Voice] Question on VPIM Addressing in Unity

[Scott ODonnell]

Just reading through some threads on VPIM problems.I wanted to confirm
something.

If I create a subscriber on UNITY , called H Phone1 with extension 1001, the
underlying domain account is HPhone1 with a smtp address of
hpho...@voip.lab(or whatever domain is configured).

I just want to make sure this is right.

Should there be some entry in Exchange for 1...@voip.lab?

As I remember, the voice connector is suppose to manage the alias of
extension to Exchange user .

- Scott