Re: [OSL | CCIE_Voice] Troubleshoot Multicast MOH

[jason sung]

i think you might be missing "voice class codec" command under your
dial-peer.




On Sat, May 10, 2008 at 8:27 AM, Jane Ryer (jryer) <[EMAIL PROTECTED]> wrote:

>  Base multicast address of 239.1.1.3 is for a G711 stream, then it goes up
> by one for the other types of codecs.  Since G729 is the third possibility
> of the four, the address you want is 239.1.1.5 in your configuration.
>
>
>
> A more typical configuration would be to set .1 as the base multicast
> address, then your router config (.3 for G729) would be correct.
>
>
>
> Also on the router, did you specify the IP addresses for which interfaces
> to send the multicast stream out?  You would want the Voice VLAN interface
> address and the Loopback address (so that calls from the PSTN would hear
> it).
>
>
>
> Good luck!
>
> Jane
>
>
>  --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *ccievoice1
> *Sent:* Friday, May 09, 2008 9:14 PM
> *To:* CCIE Voice Maillist
> *Subject:* [OSL | CCIE_Voice] Troubleshoot Multicast MOH
>
>
>
> Hi all,
>
> Not able to get Multicast MOH to work for remote site using g729. Instead,
> hearing Tone on Hold only. No problem with HQ site Multicast MOH.
>
> I configured:
> 1. ip multicast routing and ip pim dense-mode in router
> 2. ccm-manager music-on-hold | moh music-on-hold.au | multicast moh
> 239.1.1.3 port 16384 configured in router
> 3. Enable Multicast Audio Sources on this MOH Server
> 4. Base Multicast IP Address = 239.1.1.3
> 5. Base Multicast Port Number = 16384
> 6. Increment Multicast on = IP Address
> 7. Max Hops = 5
> 8. MOH Audio Source:  SampleAudioSource (1) = Allow Multicasting
> 9. In Media Resource Group =  Use Multicast for MOH Audio (requires at
> least one multicast MOH resource)
> 10. Cisco IP Voice Media Streaming App = 711 mulaw and 729 Annex A
> selected.
> 11. Media Resource Group List configured in Device Pool level with
> appropriate MRG.
> 12.MOHMulticastResourceActive counter got incremented when put the call
> On-Hold.
> 13. Reboot all the devices
> Perhaps, anyone know some good steps in troubleshooting multicast moh
> issue?
>
> Thanks.
>

Re: [OSL | CCIE_Voice] CME B-ACD

[jason sung]

I have not tried what I am about to say but technically it should work.

Under service queue, assign a hunt pilot for voicemail. For example as
follows

application
service queue
param aa-hunt5 23215

ephone-dn 5
number 23215
call-forward all 3600 (3600 is your VM pilot).

Don't forget to assign the translation that converts 23215 to 3215...

So now if a caller wants to go to voicemail press "5"
On Sat, May 10, 2008 at 1:46 PM, Mike O <[EMAIL PROTECTED]> wrote:

> I setup ACD with CME and it seems to be working pretty good. I was
> wondering if thier is away to have a user while on hold in a call queue,
> press a button and be dropped in voice mail instead of staying on hold.
>
> Is this possible?
>
> Thanks,
>
> Mike
>

Re: [OSL | CCIE_Voice] block caller-id

[jason sung]

May be it is my setup that is sending the extension number out.

thanks for your reply.

On Sat, May 10, 2008 at 6:08 PM, Kelly Love <[EMAIL PROTECTED]> wrote:

> Jason,
>
> I believe if you have a route pattern and restrict the calling line id
> presentation and calling-name presentation on the route pattern this should
> prevent anything from being sent out. I have done this before in my lab and
> have it block so that it only shows up as private on the phone I am calling.
>
>
>   On Fri, May 9, 2008 at 9:13 PM, jason sung <[EMAIL PROTECTED]> wrote:
>
> > How can I block caller-id on PSTN calls from call manager?
> >
> > I use route pattern to hide caller-id and name but the extension number
> > still goes out.
> >
> > I do not even want to send a 4-digit extension number.
> >
> > TIA
> >
>
>

[OSL | CCIE_Voice] block caller-id

[jason sung]

How can I block caller-id on PSTN calls from call manager?

I use route pattern to hide caller-id and name but the extension number
still goes out.

I do not even want to send a 4-digit extension number.

TIA

[OSL | CCIE_Voice] VG248

[jason sung]

When configuring VG248 Product specific configuration like Call Control
mode, caller-id or MWI, why don't my changes on the VG248 automatically get
updated on the callmanager device configuration?

Re: [OSL | CCIE_Voice] Gatekeeper Registration

[jason sung]

this is dependent on you callmanager group configuration.

On Mon, Apr 28, 2008 at 8:25 PM, boonchin .ng <[EMAIL PROTECTED]> wrote:

> It is actually based on the CCM Server's CTI ID.
>
> Login to your ccmadmin, goto System --> Cisco CallManager and click on
> your CCM Server. On the top of the page, you will see CTI ID: 1
>
> And, you can change the ID by editing a field in the SQL database.
>
> HTH
>
>
> On Tue, Apr 29, 2008 at 4:17 AM, Santry, Ryan <[EMAIL PROTECTED]>
> wrote:
>
> > Is there any way to change this or does it just work like that?
> >
> > Thanks
> >
> > Ryan Santry
> > Business Communications
> > Senior Technical Support
> > Sentinel Technologies, Inc
> > 2550 Warrenville Road
> > Downers Grove, IL,60515
> > SNR: 630-769-4394
> >
> >  -Original Message-
> > From: Jacob Owen [mailto:[EMAIL PROTECTED]
> > Sent: Monday, April 28, 2008 3:17 PM
> > To: Santry, Ryan; CCIE Maillist
> > Subject: Re: [OSL | CCIE_Voice] Gatekeeper Registration
> >
> > Pub would be _1
> > Sub would be _2
> >
> > --- "Santry, Ryan" <[EMAIL PROTECTED]> wrote:
> >
> > > When you register CCM with a PUB and SUB  to a
> > > gatekeeper what
> > > determines who has the _1 and _2?
> > >
> > >
> > >
> > > For example:
> > >
> > >
> > >
> > > 10.1.200.20 51156 10.1.200.20 49727 HQ-RTR
> > >  VOIP-GW
> > >
> > > H323-ID: HQ-RTR_1
> > >
> > > Voice Capacity Max.=  Avail.=  Current.= 0
> > >
> > > 10.1.200.21 52828 10.1.200.21 52756 HQ-RTR
> > >  VOIP-GW
> > >
> > > H323-ID: HQ-RTR_2
> > >
> > > Voice Capacity Max.=  Avail.=  Current.= 0
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > Thanks
> > >
> > >
> > >
> > > Ryan Santry
> > >
> > > Business Communications
> > >
> > > Senior Technical Support
> > >
> > > Sentinel Technologies, Inc
> > >
> > > 2550 Warrenville Road
> > >
> > > Downers Grove, IL,60515
> > >
> > > SNR: 630-769-4394
> > >
> > >
> > >
> > >
> > >
> > > This email may contain proprietary and confidential
> > > information for the sole use of the intended
> > > recipient.
> > > Any review, retransmission, dissemination, or other
> > > use of this information by persons or entities other
> > > than
> > > the intended recipient is prohibited. If you are not
> > > the intended recipient, please contact the sender
> > > and
> > > delete all copies. To the extent that opinions are
> > > expressed in this message, they are not necessarily
> > > the
> > > opinions of Sentinel Technologies or any of its
> > > affiliates, employees, directors, officers or
> > > shareholders.
> > >
> >
> >
> > Jacob Owen
> > CCIE #14063 (R&S, Service Provider), CCVP, CCDP
> >
> >
> >
> > 
> > 
> > Be a better friend, newshound, and
> > know-it-all with Yahoo! Mobile.  Try it now.
> > http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
> >
> >
> > This email may contain proprietary and confidential information for the
> > sole use of the intended recipient.
> > Any review, retransmission, dissemination, or other use of this
> > information by persons or entities other than
> > the intended recipient is prohibited. If you are not the intended
> > recipient, please contact the sender and
> > delete all copies. To the extent that opinions are expressed in this
> > message, they are not necessarily the
> > opinions of Sentinel Technologies or any of its affiliates, employees,
> > directors, officers or shareholders.
> >
> >
>

Re: [OSL | CCIE_Voice] RTP Header Compression Issue

[jason sung]

I used real phones.

On Fri, Apr 18, 2008 at 10:44 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> Thanks Vik and Jason.  Yes, i just confirmed it myself too.  It worked for
> me today.  I was using IP Blue Soft phone and for some reasons that didn't
> work with the compression.  I used Cisco IP Communicator today to test it
> out and it worked.
>
> Did you use IP Blue or IP Communicator to test for your compression?
>
> JD
>
>
>  --
> Date: Fri, 18 Apr 2008 15:28:22 -0500
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> Subject: Re: [OSL | CCIE_Voice] RTP Header Compression Issue
> CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
>
>
> JD, Vik,  Compression works for me.
>
> Same configs as Vik's...
>
>   Service-policy output: br1llq
> Class-map: br1rtp (match-any)
>   1134 packets, 72576 bytes
>   5 minute offered rate 0 bps, drop rate 0 bps
>   Match: ip dscp ef (46)
> 1134 packets, 72576 bytes
> 5 minute rate 0 bps
>   Queueing
> Strict Priority
> Output Queue: Conversation 40
> Bandwidth 96 (kbps) Burst 2400 (Bytes)
> (pkts matched/bytes matched) 1134/29588
> (total drops/bytes drops) 0/0
>   compress:
>   header ip rtp
>   UDP/RTP (compression on, Cisco, RTP)
> Sent:1134 total, 1133 compressed,
>  42988 bytes saved, 25052 bytes sent
>  2.71 efficiency improvement factor
>  99% hit ratio, five minute miss rate 0 misses/sec, 0
> max
>  rate 0 bps
>
>
>
> On Fri, Apr 18, 2008 at 7:29 AM, Devildoc <[EMAIL PROTECTED]> wrote:
>
> Hi Vik,
>
> I believe you, and I would never think that you would hoax me in anyway.
> :)
>
> I just don't know why my configuration didn't work, so i'll retry it
> tonight with the router reloaded and report back to you.  Thanks for all
> your help.
>
> JD
>
>
>  --
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] RTP Header Compression Issue
> Date: Thu, 17 Apr 2008 10:26:19 -0700
>
>
> JD- I assure you I have it working, my cut n paste excluded the last line
> which is "  service-policy WAN-EDGE-BR2" - I'm not trying to hoax you:-)
>
> I have it working with the config, try again and report back but the only
> thing I did was reload. I will hazard a guess that you will not have any
> problem seeing the compression working.
>
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
> 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:* Devildoc [mailto:[EMAIL PROTECTED]
> *Sent:* Thursday, April 17, 2008 5:40 AM
> *To:* [EMAIL PROTECTED]; 'CCIE Voice Online Study List'
> *Subject:* RE: [OSL | CCIE_Voice] RTP Header Compression Issue
>
> Vik,
>
> Thanks for the info.  My configuration is pretty much like yours, except
> that according to your configuration below, i don't see how you had your
> compression working since you didn't even associate your CBWFQ policy 
> WAN-EDGE-BR2
> to your class-based traffic shaping policy shape-br2.
>
> In my case, after I applied my CBWFQ policy to the class-based traffic
> shaping policy and attached my shaping policy to the FR map class (without
> reloading the router), I did a show policy-map interface while making a call
> from HQ to BR1 (in my case it is between HQ and BR1) and saw that the number
> of packets for RTP and signal traffics incremented while my compression
> packets remained at zero.
>
> However, if i applied my CBWFQ policy to the virtual interface using MLP
> and used generic traffic shaping under the FR map class, then compression
> worked.  I saw that the number of packets for RTP, signal, and
> compression all incremented.
>
> I'll give it another try next time but with the router reloaded to see if
> that makes any difference.  Thanks for your info.
>
> JD
>
>
>
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] RTP Header Compression Issue
> Date: Wed, 16 Apr 2008 14:40:05 -0700
>
>
>  I got it working with class-based shaping. Make sure you really do have a
> call traversing the WAN and that you have matches in your RTP class within
> the policy-map. I reloaded the router (for a different reason) so I am not
> 100% sure if this is required. If your config looks good maybe worth a try.
> I have the policy-map I used shown below.
>
>
> class-map match-all RTP

Re: [OSL | CCIE_Voice] RTP Header Compression Issue

[jason sung]

JD, Vik,  Compression works for me.

Same configs as Vik's...

  Service-policy output: br1llq
Class-map: br1rtp (match-any)
  1134 packets, 72576 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: ip dscp ef (46)
1134 packets, 72576 bytes
5 minute rate 0 bps
  Queueing
Strict Priority
Output Queue: Conversation 40
Bandwidth 96 (kbps) Burst 2400 (Bytes)
(pkts matched/bytes matched) 1134/29588
(total drops/bytes drops) 0/0
  compress:
  header ip rtp
  UDP/RTP (compression on, Cisco, RTP)
Sent:1134 total, 1133 compressed,
 42988 bytes saved, 25052 bytes sent
 2.71 efficiency improvement factor
 99% hit ratio, five minute miss rate 0 misses/sec, 0
max
 rate 0 bps



On Fri, Apr 18, 2008 at 7:29 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> Hi Vik,
>
> I believe you, and I would never think that you would hoax me in anyway.
> :)
>
> I just don't know why my configuration didn't work, so i'll retry it
> tonight with the router reloaded and report back to you.  Thanks for all
> your help.
>
> JD
>
>
>  --
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] RTP Header Compression Issue
> Date: Thu, 17 Apr 2008 10:26:19 -0700
>
>
> JD- I assure you I have it working, my cut n paste excluded the last line
> which is "  service-policy WAN-EDGE-BR2" - I'm not trying to hoax you:-)
>
> I have it working with the config, try again and report back but the only
> thing I did was reload. I will hazard a guess that you will not have any
> problem seeing the compression working.
>
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
> 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:* Devildoc [mailto:[EMAIL PROTECTED]
> *Sent:* Thursday, April 17, 2008 5:40 AM
> *To:* [EMAIL PROTECTED]; 'CCIE Voice Online Study List'
> *Subject:* RE: [OSL | CCIE_Voice] RTP Header Compression Issue
>
> Vik,
>
> Thanks for the info.  My configuration is pretty much like yours, except
> that according to your configuration below, i don't see how you had your
> compression working since you didn't even associate your CBWFQ policy 
> WAN-EDGE-BR2
> to your class-based traffic shaping policy shape-br2.
>
> In my case, after I applied my CBWFQ policy to the class-based traffic
> shaping policy and attached my shaping policy to the FR map class (without
> reloading the router), I did a show policy-map interface while making a call
> from HQ to BR1 (in my case it is between HQ and BR1) and saw that the number
> of packets for RTP and signal traffics incremented while my compression
> packets remained at zero.
>
> However, if i applied my CBWFQ policy to the virtual interface using MLP
> and used generic traffic shaping under the FR map class, then compression
> worked.  I saw that the number of packets for RTP, signal, and
> compression all incremented.
>
> I'll give it another try next time but with the router reloaded to see if
> that makes any difference.  Thanks for your info.
>
> JD
>
>
>
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] RTP Header Compression Issue
> Date: Wed, 16 Apr 2008 14:40:05 -0700
>
>
>  I got it working with class-based shaping. Make sure you really do have a
> call traversing the WAN and that you have matches in your RTP class within
> the policy-map. I reloaded the router (for a different reason) so I am not
> 100% sure if this is required. If your config looks good maybe worth a try.
> I have the policy-map I used shown below.
>
>
> class-map match-all RTP
>  match ip dscp ef
> class-map match-all SIG
>  match ip dscp cs3
> !
> !
> policy-map WAN-EDGE-BR2
>  class RTP
>   priority 120
>compress header ip rtp
>  class SIG
>   bandwidth 32
>  class class-default
>   fair-queue
> policy-map shape-br2
>  class class-default
>   shape average 729600 7296 0
> !
> !
> interface Serial0/0/0:0.2 point-to-point
>  ...
>  frame-relay interface-dlci 202
>   class br2
> !
> !
> map-class frame-relay br2
>  frame-relay cir 729600
>  frame-relay bc 7296
>  frame-relay be 0
>  frame-relay mincir 729600
>  service-policy output shape-br2
>
> Here is a snippet of the output of sh policy-map interface
>
> Service-policy : WAN-EDGE-BR2
>
>   Class-map: RTP (match-all)
> 1711 packets,

Re: [OSL | CCIE_Voice] RTP Header Compression Issue

[jason sung]

i am interested in this problem.

I will try this in my lab later today and post my findings.

On Thu, Apr 17, 2008 at 7:40 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> Vik,
>
> Thanks for the info.  My configuration is pretty much like yours, except
> that according to your configuration below, i don't see how you had your
> compression working since you didn't even associate your CBWFQ policy 
> WAN-EDGE-BR2
> to your class-based traffic shaping policy shape-br2.
>
> In my case, after I applied my CBWFQ policy to the class-based traffic
> shaping policy and attached my shaping policy to the FR map class (without
> reloading the router), I did a show policy-map interface while making a call
> from HQ to BR1 (in my case it is between HQ and BR1) and saw that the number
> of packets for RTP and signal traffics incremented while my compression
> packets remained at zero.
>
> However, if i applied my CBWFQ policy to the virtual interface using MLP
> and used generic traffic shaping under the FR map class, then compression
> worked.  I saw that the number of packets for RTP, signal, and
> compression all incremented.
>
> I'll give it another try next time but with the router reloaded to see if
> that makes any difference.  Thanks for your info.
>
> JD
>
>
>
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Subject: RE: [OSL | CCIE_Voice] RTP Header Compression Issue
> Date: Wed, 16 Apr 2008 14:40:05 -0700
>
>
>
>  I got it working with class-based shaping. Make sure you really do have a
> call traversing the WAN and that you have matches in your RTP class within
> the policy-map. I reloaded the router (for a different reason) so I am not
> 100% sure if this is required. If your config looks good maybe worth a try.
> I have the policy-map I used shown below.
>
>
> class-map match-all RTP
>  match ip dscp ef
> class-map match-all SIG
>  match ip dscp cs3
> !
> !
> policy-map WAN-EDGE-BR2
>  class RTP
>   priority 120
>compress header ip rtp
>  class SIG
>   bandwidth 32
>  class class-default
>   fair-queue
> policy-map shape-br2
>  class class-default
>   shape average 729600 7296 0
> !
> !
> interface Serial0/0/0:0.2 point-to-point
>  ...
>  frame-relay interface-dlci 202
>   class br2
> !
> !
> map-class frame-relay br2
>  frame-relay cir 729600
>  frame-relay bc 7296
>  frame-relay be 0
>  frame-relay mincir 729600
>  service-policy output shape-br2
>
> Here is a snippet of the output of sh policy-map interface
>
> Service-policy : WAN-EDGE-BR2
>
>   Class-map: RTP (match-all)
> 1711 packets, 109504 bytes
> 5 minute offered rate 8000 bps, drop rate 0 bps
> Match: ip dscp ef (46)
> Queueing
>   Strict Priority
>   Output Queue: Conversation 72
>   Bandwidth 120 (kbps) Burst 3000 (Bytes)
>   (pkts matched/bytes matched) 0/0
>   (total drops/bytes drops) 0/0
> compress:
> header ip rtp
> UDP/RTP (compression on, Cisco, RTP)
>   Sent:1827 total, 1826 compressed,
>
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
> 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:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Devildoc
> *Sent:* Wednesday, April 16, 2008 12:57 PM
> *To:* CCIE Voice Online Study List
> *Subject:* [OSL | CCIE_Voice] RTP Header Compression Issue
>
> Hello,
>
> Has anyone gotten RTP header compression working with LLQ?  I have tried
> different methods (legacy FRTS and class-based FRTS) and none of them
> worked.  I could only get it to work if I used MLP and attached the LLQ
> policy to the virtual interface.  That seemed to work fine.  Short of MLP,
> nothing worked for me.
>
> Here is the compression section of the show policy-map interface output:
>
>   compress:
>   header ip rtp
>   UDP/RTP (compression on, Cisco, RTP)
> Sent:0 total, 0 compressed,
>  0 bytes saved, 0 bytes sent
>  rate 0 bps
>
> According to the output, compression is on and it is using Cisco format.
> In the MLP case, compression was using IPHC format.
>
> So what are the differences between the 2 formats and why did MLP work and
> other FRTS methods didn't work?  Anyone has any idea?  Thanks.
>
> JD
>
>
>
> --
> Use video conversation to talk face-to-face with Windows Live Messenger. Get
> started!
>
>
> --

Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail...

[jason sung]

Two Methods.
#1. Vik's special (Easier to configure)


(1) Ensure that the HQ site has some spare DID numbers- e.g. in PL we route
21222x1...to the HQ gateway.



(2) Use the alias command in call-manager-fallback to route each extension
to a unique DID number. E.g.



Call-manager-fallback

voicemail 912122211600

alias 1 2001 to 2001 cfw 912122211661 timeout 12

alias 2 2002 to 2002 cfw 912122211662 timeout 12

alias 3 2003 to 2003 cfw 912122211663 timeout 12



(3) On the CallManager create a Translation Pattern as shown below:



DN = 166X / pt-internal

CSS = css-internal

Called # Mask = 200x



(4) When the CCM receives the call it tries to ring 200X which is not
registered. It will then use the call fwd no answer setting (which should be
send to VM).



(5) Add the Alternate Extension on Unity so that direct calls are routed to
subscriber sign-in.

#2 Vm-integration since PRI probably will not support RDNIS



> CONFIGURATION

> telephony-setup or

> call-manager-fallback

>  voicemail 913335551234

>  call-forward busy 913335551234

>  call-forward noan 913335551234 timeout 12

>  timeout interdigit 7

>

> vm-integration

>  pattern direct # CGN (setup alternate extension in Unity that

> matches full 10-digit number)  <--- Don't put in this step --

>  pattern trunk-to-ext busy # FDN

>  pattern trunk-to-ext noan # FDN

>  pattern ext-to-ext busy # FDN

>  pattern ext-to-ext noan # FDN

>

> dial-peer voice 86 pots

>  destination-pattern 913335551234T (make sure you add the T so it

> will append the # FDN)

>  port 0/0/0:23

>  prefix 13335551234

On Wed, Apr 16, 2008 at 1:03 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> OK, so AAR kicks in and I forward the call to the PSTN, the user I was
> dialing does not answer and the call should forward to vmail... how
> would we make this work? voicemail under srst config?
>
>
>
> Jonathan
>

Re: [OSL | CCIE_Voice] "ATA Not Found"

[jason sung]

Me too.

On Sat, Apr 12, 2008 at 11:28 AM, boonchin .ng <[EMAIL PROTECTED]>
wrote:

> Hi Maark,
>
> I followed your steps exactly and I got this loaded in the ATA186
> (attached screen shoot)
>
> VOICE-6500-1> (enable) sh cdp nei 2/27 de
> Port (Our Port): 2/27
> Device-ID: SEP000FF7C054DB
> Device Addresses:
>   IP Address: 10.5.200.112
> Holdtime: 140 sec
> Capabilities: HOST
> Version:
>   ATA030102H323040927A
> Platform: Cisco ATA 186
> Port-ID (Port on Neighbors's Device): Port 1
> VTP Management Domain: unknown
> Native VLAN: unknown
> Duplex: half
> System Name: unknown
> System Object ID: unknown
> Management Addresses: unknown
> Physical Location: unknown
> VOICE-6500-1> (enable)
>
> Thanks
>
> On Thu, Feb 14, 2008 at 11:47 AM, Mark Snow <[EMAIL PROTECTED]> wrote:
>
> > Hey Anil - I cannot get to the ATA currently without an IP address - and
> > someone else is on that pod.
> >
> > What I do suggest for you and anyone else in this situation is as
> > follows:
> > - I have scripted a way in which the ATA's can all "Auto-Update" from
> > H323 to SCCP image (I can accommodate the opposite effect - but it requires
> > some intervention and currently they it is set to upgrade to SCCP).
> > - What you need to do is get the ATA to obtain an IP address (be patient
> > - just like IP Phones they are a little finicky and take a while to get an
> > IP and actually respond).
> > - Web to the device with a  http://10.xx.200.x/dev  and change the 2nd
> > two fields to set them exactly as follows (screenshot included):
> >UseTFTP=1
> >TFTPURL=192.20.200.30
> > - Then hit apply and give it some time while it reboots and gets its new
> > config. If for some reason after you have hit apply, and the ATA allows you
> > to refresh the web page - but still shows an H323 load - then power the
> > device off for about 30 seconds and power it back on (you can do this from
> > your ProctorLabs.com Voice vRack web page).
> >
> > - Once you can get to the device again - you will now see it in SCCP
> > mode - change your TFTPURL to your UCM Server for your Pod and you should be
> > done (screenshot for that included as well).
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Hope that helps a few folks!
> >
> > (Do not do this if you still want to practice in H323 mode)
> >
> >
> > 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 Feb 12, 2008, at 7:22 PM, anil batra wrote:
> >
> > Apprciate your kind response and willingness to
> > > upgrade. But was wondering if I will be able to do
> > > myslef during my next session. If not can you please
> > > do it for me on Pod14 I will be using on Feb 16th from
> > > 1PM to 9PM EST.
> > >
> > > Regards,
> > > Anil
> > >
> > > --- Mark Snow <[EMAIL PROTECTED]> wrote:
> > >
> > > If you send me the IP Address of it - I can change
> > > > it to SCCP for you
> > > > if you like ...
> > > >
> > > >
> > > > 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 Feb 12, 2008, at 3:24 PM, anil batra wrote:
> > > >
> > > > never mid it was set to H323 firmware.
> > > > >
> > > > > --- anil batra <[EMAIL PROTECTED]> wrote:
> > > > >
> > > > > Finally ATA got IP from DHCP but once I added it
> > > > > >
> > > > > to
> > > >
> > > > > CCM and it shows not found. I double checked MAC
> > > > > > address enter is correct. Any clue please ???
> > > > > >
> > > > > > Regards,
> > > > > > Anil
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > >
> > > >
> > > 
> > >
> > > >  Be a better friend, newshound, and
> > > > > > know-it-all with Yahoo! Mobile.  Try it now.
> > > > > >
> > > > > >
> > > > >
> > > > http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
> > >
> > > >
> > > > > >
> > > > > >
> > > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > >
> > > ___

Re: [OSL | CCIE_Voice] VOICE Passed !!!!!!!!!!!!

[jason sung]

Dude,

A monkey can pass the test given the questions.

Please keep your tips and ideas to yourself.

I am sure you will pass your next CCIE using cciecert.net

On Fri, Apr 11, 2008 at 7:26 PM, ccie2007 <[EMAIL PROTECTED]> wrote:

> I just passed yesterday on Tokyo
>
> I am really pleasure with this achievement
>
>
> First my recommendation for all guys to understand all topic of the blue
> print
> from Cisco site and documentation CD as a main resource
>
> Second I use Internetwork Expert's as practical Labs which contain a lot
> of the real LAB concepts, great explanation for various topics and cover
> almost all topics in the blue print. thanks Brain
>
> Also i really recommand that you go to cciecert.net then you will get a
> real ccie LAB information
> from this site
>
> My advice to all to go through this certificate because I have now a lot
> of understanding of network technology
>
> My next attempt may be CCIE SP
>
>  Regards
>  #14867 CCIE Security, R/S, VOICE
>  Hiroyasu Kato
>
>

Re: [OSL | CCIE_Voice] Multicast MOh

[jason sung]

reboot. Everything started working as expected.

I have this very nasty habbit of not rebooting things

On Fri, Apr 11, 2008 at 12:40 PM, Scott Monasmith <[EMAIL PROTECTED]>
wrote:

> I don't know why the stats are displaying under show call-manager-fallback
> since you don't have 'moh music-on-hold.au' configured. I don't know if
> these stats reflect multicast group announcements. I would venture to guess
> that since the phone was configured to use multicast moh (via the mrgl)
> using the ip of 239.1.1.1 and your router is configured to multicast moh
> traffic with 239.1.1.1 that it sends out multicast packets without regard
> to whether it contains music or not. BUT THAT IS JUST A GUESS.
>
>   On Fri, Apr 11, 2008 at 12:27 PM, jason sung <[EMAIL PROTECTED]> wrote:
>
> > Thanks Scott,
> >
> > That is what I wanted to know. I thought so too, but your confirmation
> > adds to my confidence.
> >
> > I do see the ccm-manager music-on-hold stats, but I have not configured
> > any moh source under call-manager-fallback so why do i see the stats?
> >
> > Is the router acting up and picking up old configs (for which I did have
> > moh configured), may be it needs a reboot???
> >
> >   On Fri, Apr 11, 2008 at 12:09 PM, Scott Monasmith <[EMAIL PROTECTED]>
> > wrote:
> >
> > > Correct. I was just answering his initial question regarding the
> > > perfmon counters.
> > >
> > > you can also run 'debug ephone moh'
> > >
> > >   On Fri, Apr 11, 2008 at 12:06 PM, Jonathan Charles <
> > > [EMAIL PROTECTED]> wrote:
> > >
> > > > Right, but that does not prove it is working, the show ccm music
> > > > will
> > > > prove it tho...
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On Fri, Apr 11, 2008 at 12:05 PM, Scott Monasmith <
> > > > [EMAIL PROTECTED]> wrote:
> > > > > Jason,
> > > > >
> > > > > If you configure your branch router to source Multicast MoH via
> > > > SRST, you
> > > > > should still see the MoH perfmon counters increase even though the
> > > > MoH is
> > > > > not traversing the WAN.
> > > > >
> > > > >
> > > > >
> > > > > On Fri, Apr 11, 2008 at 10:53 AM, jason sung <[EMAIL PROTECTED]>
> > > > wrote:
> > > > >
> > > > > >
> > > > > > Should I see my perf mon counters increase when i am using
> > > > remote site
> > > > > router as the moh source.
> > > > > >
> > > > > > Here is how I am trying to isolate.
> > > > > >
> > > > > > I have multicast-routing disabled on the wan interfaces (no ip
> > > > pim
> > > > > dense-mode)
> > > > > >
> > > > > > My MOH server source hop count is set to 1.
> > > > > >
> > > > > > Remote Site MRG is set to use Multicast MOH on callmanager
> > > > configuration.
> > > > > >
> > > > > > I have no moh configured on the remote site router, but I do see
> > > > my
> > > > > Perfmon counters increase.
> > > > > >
> > > > > > should the end result be no MOH??
> > > > >
> > > > >
> > > > >
> > > > > --
> > > > > "There are only 10 types of people in the world: Those who
> > > > understand
> > > > > binary, and those who don't"
> > > >
> > >
> > >
> > >
> > > --
> > >  "There are only 10 types of people in the world: Those who understand
> > > binary, and those who don't"
> > >
> >
> >
>
>
> --
>  "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"
>

Re: [OSL | CCIE_Voice] Multicast MOh

[jason sung]

Thanks Scott,

That is what I wanted to know. I thought so too, but your confirmation adds
to my confidence.

I do see the ccm-manager music-on-hold stats, but I have not configured any
moh source under call-manager-fallback so why do i see the stats?

Is the router acting up and picking up old configs (for which I did have moh
configured), may be it needs a reboot???

On Fri, Apr 11, 2008 at 12:09 PM, Scott Monasmith <[EMAIL PROTECTED]>
wrote:

> Correct. I was just answering his initial question regarding the perfmon
> counters.
>
> you can also run 'debug ephone moh'
>
>   On Fri, Apr 11, 2008 at 12:06 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
>
> > Right, but that does not prove it is working, the show ccm music will
> > prove it tho...
> >
> >
> > Jonathan
> >
> > On Fri, Apr 11, 2008 at 12:05 PM, Scott Monasmith <[EMAIL PROTECTED]>
> > wrote:
> > > Jason,
> > >
> > > If you configure your branch router to source Multicast MoH via SRST,
> > you
> > > should still see the MoH perfmon counters increase even though the MoH
> > is
> > > not traversing the WAN.
> > >
> > >
> > >
> > > On Fri, Apr 11, 2008 at 10:53 AM, jason sung <[EMAIL PROTECTED]>
> > wrote:
> > >
> > > >
> > > > Should I see my perf mon counters increase when i am using remote
> > site
> > > router as the moh source.
> > > >
> > > > Here is how I am trying to isolate.
> > > >
> > > > I have multicast-routing disabled on the wan interfaces (no ip pim
> > > dense-mode)
> > > >
> > > > My MOH server source hop count is set to 1.
> > > >
> > > > Remote Site MRG is set to use Multicast MOH on callmanager
> > configuration.
> > > >
> > > > I have no moh configured on the remote site router, but I do see my
> > > Perfmon counters increase.
> > > >
> > > > should the end result be no MOH??
> > >
> > >
> > >
> > > --
> > > "There are only 10 types of people in the world: Those who understand
> > > binary, and those who don't"
> >
>
>
>
> --
>  "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"
>

[OSL | CCIE_Voice] Multicast MOh

[jason sung]

Should I see my perf mon counters increase when i am using remote site
router as the moh source.

Here is how I am trying to isolate.

I have multicast-routing disabled on the wan interfaces (no ip pim
dense-mode)

My MOH server source hop count is set to 1.

Remote Site MRG is set to use Multicast MOH on callmanager configuration.

I have no moh configured on the remote site router, but I do see my Perfmon
counters increase.

should the end result be no MOH??

Re: [OSL | CCIE_Voice] Calling number manipulation on MGCP gateways

[jason sung]

Try service parameters.

callmanager service parameter: "NATIONAL NUMBER PREFIX"

I have done this for years. This puts 91 to all the incoming calls, missed
calls(on the phone)

On Wed, Apr 9, 2008 at 3:46 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:

> Yeah. That is what I m trying to achieve. I have not really been an MGCP
> man in the past. Having always opted for h323 gateways in the past for all
> of my customers and having that flexibility I have always been able to
> translate the calling number on inbound calls at the voice port on the
> gateway. I was hoping to find an equivilent on MGCP. I found the
> International prefix parametre but if you say that only works with numbering
> plan then its no good for me. It is a pain for incoming numbers. If anyone
> has any other ideas i would appreciate your thoughts.
>
> Thanks again guys for an excellent forum.
>
>
> Paul
>
>
> On Thu, Apr 10, 2008 at 6:29 AM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
>
> > I know what you are trying to do; you want the callback number in the
> > directories to be ready to dial, so they don't have to EditDial and
> > add the 91 in front
> >
> > This has been a pain for years... not sure if there is a reasonable
> > solution tho...
> >
> >
> >
> > Jonathan
> >
> > On Wed, Apr 9, 2008 at 2:00 AM, Paul and Bobs <[EMAIL PROTECTED]>
> > wrote:
> > > Guys
> > >
> > > Trying to manipulate the calling number on MDCP gateway. I can do it n
> > h323
> > > with voice translation patterns but wandering if there was similar for
> > all
> > > incoming on MGCP. I know its controlled by ccm and the config will
> > have to
> > > be here but i cant think of a generic translation pattern that will
> > capture
> > > all incoming calls and maipulate the calling number to say add a 91 to
> > the
> > > front.
> > >
> > > Cheers
> > >
> > > Paul
> > >
> >
>
>

Re: [OSL | CCIE_Voice] GK Technology Prefix

[jason sung]

Like Jonathan mentioned # is only for ease of reading debugs.

Gatekeeper work just fine without the #.

On Wed, Apr 9, 2008 at 3:33 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> Does anyone know if it is a standard practice to include a # sign with a
> tech-prefix?  Let say... if you are asked to configured a tech prefix of 2
> for one gateway and another tech prefix of 3 for another gateway, is it
> assumed that the configured tech prefix should be 2# and 3# respectively?  I
> know you don't have to include the # with the tech prefix and it works just
> fine, but i was just wondering if it's a standard or good practice to
> include the # sign.  I couldn't find any document on the Internet that
> references the tech prefix without using the # sign.
>
> JD
>
> --
> Get in touch in an instant. Get Windows Live Messenger 
> now.
>

Re: [OSL | CCIE_Voice] dial-peer patterns

[jason sung]

Yep, T does the Trick.


thank you all who replied.

On Wed, Apr 9, 2008 at 12:50 PM, Erin Mikelson <[EMAIL PROTECTED]>
wrote:

> The only way to get this to work is to add the T at the end of the local
> pattern.
>
>
> > From: [EMAIL PROTECTED]
> > Subject: CCIE_Voice Digest, Vol 26, Issue 83
> > To: ccie_voice@onlinestudylist.com
> > Date: Wed, 9 Apr 2008 12:00:02 -0400
> >
> > Send CCIE_Voice mailing list submissions to
> > ccie_voice@onlinestudylist.com
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > http://onlinestudylist.com/mailman/listinfo/ccie_voice
> > or, via email, send a message with subject or body 'help' to
> > [EMAIL PROTECTED]
> >
> > You can reach the person managing the list at
> > [EMAIL PROTECTED]
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of CCIE_Voice digest..."
> >
> >
> > Today's Topics:
> >
> > 1. dial-peer patterns (jason sung)
> > 2. Re: dial-peer patterns (ccievoice1)
> > 3. Re: dial-peer patterns (Matthew Cody)
> >
> >
> > --
> >
> > Message: 1
> > Date: Wed, 9 Apr 2008 10:09:02 -0500
> > From: "jason sung" <[EMAIL PROTECTED]>
> > Subject: [OSL | CCIE_Voice] dial-peer patterns
> > To: "CCIE Voice Online Study List" 
> > Message-ID:
> > <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Has anyone been able to get long distance working using the following
> two
> > dial-peer patterns
> > 1.Local 9[1-9]..
> > 2. LD 91[1-9]..[1-9]..
> >
> > I tweaked it all the possible ways but every time I dialed the LD number
> the
> > router snapped it and sent it out as local.
> >
> > I do know that both patterns are possible matches when you start dialing
> but
> > shouldn't the router wait for rest of the digits to come in and than
> make a
> > decision?
> >
> > I know this is most likely not going to happen in lab but I just wanted
> to
> > get this working. It is bugging me.
> > -- next part --
> > An HTML attachment was scrubbed...
> > URL:
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080409/b51bee9e/attachment-0001.html
> >
> > --
> >
> > Message: 2
> > Date: Wed, 9 Apr 2008 23:11:02 +0800
> > From: ccievoice1 <[EMAIL PROTECTED]>
> > Subject: Re: [OSL | CCIE_Voice] dial-peer patterns
> > To: "jason sung" <[EMAIL PROTECTED]>
> > Cc: CCIE Voice Online Study List 
> > Message-ID:
> > <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Try adding a T at the end of your local dial-peer ?
> >
> > HTH
> >
> > On Wed, Apr 9, 2008 at 11:09 PM, jason sung <[EMAIL PROTECTED]> wrote:
> >
> > > Has anyone been able to get long distance working using the following
> two
> > > dial-peer patterns
> > > 1.Local 9[1-9]..
> > > 2. LD 91[1-9]..[1-9]..
> > >
> > > I tweaked it all the possible ways but every time I dialed the LD
> number
> > > the router snapped it and sent it out as local.
> > >
> > > I do know that both patterns are possible matches when you start
> > > dialing but shouldn't the router wait for rest of the digits to come
> in and
> > > than make a decision?
> > >
> > > I know this is most likely not going to happen in lab but I just
> wanted to
> > > get this working. It is bugging me.
> > >
> > -- next part --
> > An HTML attachment was scrubbed...
> > URL:
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080409/e4284c85/attachment-0001.html
> >
> > --
> >
> > Message: 3
> > Date: Wed, 9 Apr 2008 12:50:14 -0400
> > From: "Matthew Cody" <[EMAIL PROTECTED]>
> > Subject: Re: [OSL | CCIE_Voice] dial-peer patterns
> > To: "'ccievoice1'" <[EMAIL PROTECTED]>, "'jason sung'"
> > <[EMAIL PROTECTED]>
> > Cc: 'CCIE Voice Online Study List' 
> > Message-ID: <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="us-ascii"
> >
> > If this is a US number (NANP), then it should be [2-9] not [1-9].
> >
> >

[OSL | CCIE_Voice] dial-peer patterns

[jason sung]

Has anyone been able to get long distance working using the following two
dial-peer patterns
1.Local 9[1-9]..
2. LD 91[1-9]..[1-9]..

I tweaked it all the possible ways but every time I dialed the LD number the
router snapped it and sent it out as local.

I do know that both patterns are possible matches when you start dialing but
shouldn't the router wait for rest of the digits to come in and than make a
decision?

I know this is most likely not going to happen in lab but I just wanted to
get this working. It is bugging me.

Re: [OSL | CCIE_Voice] Gatekeeper tech prefix

[jason sung]

Yes.

On Mon, Apr 7, 2008 at 5:20 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:

> Thanks adam
>
> So what happens to GK if I have two or more hosts registered with th same
> tech prefix. Does it then look at the zone prefix.
>
> Paul
>
> On Mon, Apr 7, 2008 at 10:53 PM, <[EMAIL PROTECTED]> wrote:
>
> >
> > The tech prefix helps GK steer the call to the correct registered
> > gateway.  i.e., 1# could be CCM, 2# could be CME, etc.
> >
> >
> >
> >
> > Adam W. Moore
> > CCIE-Voice #18462
> > CCVP, CCNP, CIPTSS, UCSE
> > CIPTOS, CIPCCXS, CCNA, CCDA
> > MCSE(2K/2K3), MCSA(2K/2K3), MCP
> > Verizon Network Integration
> > 500 Technology Drive
> > South Charleston, WV 25309
> > 304-746-1011(W), 304-545-5261(C)
> >
> >
> >*"Paul and Bobs" <[EMAIL PROTECTED]>*
> >
> > 04/06/2008 10:16 PM
> >To
> > Adam W. Moore/EMPL/WV/[EMAIL PROTECTED]  cc
> > "Matthew Cody" <[EMAIL PROTECTED]>, ccie_voice@onlinestudylist.com,
> > "Jonathan Charles" <[EMAIL PROTECTED]>  Subject
> > Gatekeeper tech prefix
> >
> >
> >
> >
> > I am getting myself confused with this tech prefix. If I register my CCM
> > H225 trunk to Gatekeeper and configure tech-prefix on the CCM trunk to be 1#
> > does that let the gatekeeper know to send all calls it recieves beginning
> > with 1 to itself. Or do you have to be dialing 1#.
> >
> > I have it working I am just trying to understand how its working. I have
> > it configured with prefix of 1 and no default tech-prefix on the gatekeeper.
> >
> > What I am ultimately trying to achieve it call routing through the
> > Gatekeeper without any manipulation of the gatekeeper config.
> >
> > ie .
> >
> > gatekeeper
> > zone local HQ-RTR *ipexpert.com* 
> > no shut
> >
> > I have the CME registering with the Gatekeeper and it forwards its call
> > routing table to the gatekeeper. But trying to do the same thing with CCM.
> >
> > Not sure if there is a default route that I can use to say to the
> > gatekeeper that if you dont know about a call then forward it to the CCM
> > h225 trunk.
> >
> > Thanks
> >
> > Paul
> >
>
>

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[jason sung]

Here is an excerpt from the report card.

"Your CCIE Certification Lab exam was scored based on grading policies that
are adhered to uniformly by our proctors worldwide. Marking was based on
whether the answer you provided works. Candidates are not required to use a
set methodology in achieving a correct result. The imperative is that the
solution provided produces the outcome requested. "
I do believe that they actually grade you correctly but again there are
chances of proctors messing up as well.

Like Scott mentioned, I would just focus on things I can control and forget
the rest... better luck next time buddy...



On Sat, Apr 5, 2008 at 1:09 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> They told me the reason why they don't hold the voice exams is because
> they would take up a lot more storage (hard drives/images of 3 servers,
> router configs, etc.) vs. just backing up a few config files from a
> router/switch to a text file for the R&S lab.
>
> Sorry.
>
>
>   On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> >
> > What?!?!?
> >
> >
> > Jonathan
> >
> > On Sat, Apr 5, 2008 at 1:07 PM, Scott Monasmith <[EMAIL PROTECTED]>
> > wrote:
> > > Correct. BUT not available for voice though, Jonathon.
> > >
> > >
> > >
> > >
> > > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > > Yeah, as I thought:
> > > >
> > > > "Reevaluation of Results
> > > >
> > > > If you are concerned your results are in error, you may request a
> > > > "reread" until 14 days after your lab date via an email to
> > > > [EMAIL PROTECTED] Each reread costs $250.00 USD and consists of a
> > > > proctor loading your configurations into a rack to recreate the test
> > > > and re-score the entire exam. This process may take up to three
> > weeks
> > > > after receipt of payment"
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On Sat, Apr 5, 2008 at 1:04 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > > wrote:
> > > > > There is a rescore option... isn't there? There was for the R&S...
> > > > >
> > > > >
> > > > >  Jonathan
> > > > >
> > > > >
> > > > >
> > > > >  On Sat, Apr 5, 2008 at 12:57 PM, Scott Monasmith <
> > [EMAIL PROTECTED]>
> > > wrote:
> > > > >  > Devildoc, I do feel your frustration. I finished my last
> > attempt in a
> > > little
> > > > >  > over 5 hours and spent the next 3 hours verifying my work. I
> > walked
> > > out of
> > > > >  > the exam feeling like I nailed it. However, based on my score
> > report,
> > > it
> > > > >  > reflected a score of about 74-76 points. There were 3 sections
> > where
> > > the
> > > > >  > score came out and I was left scratching my head thinking "how
> > can
> > > this be?"
> > > > >  > - Talk about deflating. I had plenty of time to verify
> > everthing and
> > > I felt
> > > > >  > very good about my chances. And to this day I still have no
> > idea how
> > > I could
> > > > >  > have missed points on those sections.
> > > > >  >
> > > > >  > To me, there are 2 things we can do:
> > > > >  > 1. study harder
> > > > >  > 2. after each failed attempt, continue to stress to cisco (via
> > the
> > > critique
> > > > >  > in your score report) that a re-score option needs to be
> > established
> > > for the
> > > > >  > exam.
> > > > >  >
> > > > >  > If I'm spending $2,000 (exam + travel) for each attempt, the
> > least
> > > they can
> > > > >  > do is reassure us that they are doing everything possible to
> > ensure
> > > that
> > > > >  > there are no errors in the grading.
> > > > >  >
> > > > >  > BTW, a proctor told me that voice is the most challenging to
> > grade
> > > since
> > > > >  > there is more than one way to achieve the desired results
> > > > >  >
> > > > >  >
> > > > >  >
> > > > >  >
> > > > >  > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > > >  > > I ran into the same problem with the R&S lab... there might
> > be 3 or
> > > 4
> > > > >  > > ways to do something, but only one of them gets you points...
> > not
> > > sure
> > > > >  > > if this is the same thing on the CCIE Voice lab... but I
> > would bet
> > > it
> > > > >  > > is.
> > > > >  > >
> > > > >  > > No idea on how they grade the test.
> > > > >  > >
> > > > >  > > But I think a lot has to do with how you use the proctor...
> > so, if
> > > > >  > > there are 2 ways to do something, that means to go to the
> > proctor
> > > and
> > > > >  > > say, 'hey, I have way A and way B... which one is preferred?'
> > > > >  > >
> > > > >  > > Now, I would bet that one of those ways doesn't meet the
> > > > >  > > requirements... which is why this test is as difficult as it
> > is...
> > > > >  > > because you are going to have to know why 'way B' doesn't
> > work...
> > > > >  > > which means a deep understanding of not just how to configure
> > > > >  > > something, but in the way it works at a protocol level.
> > > > >  > >
> > > > >  > > For example... let's say I wanted you to set up CAC for a
> > specific
> > > > >  > > location. Now, no big, right, just set up 

Re: [OSL | CCIE_Voice] Primer on voice translation rules...

[jason sung]

check out document id:61083.

On Sat, Apr 5, 2008 at 1:13 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> I found that if I remove the \ it makes it easier to read...
>
> So, this /^9\(555\)\(.\)77\(6\)/  /1800\1\3\66\2/
>
> Becomes
>
> /^9(555)(.)77(6)/  /1800\1\3\66\2/
>
> So, now it is clear that starting with a 9, then 555, a dot, then 77 and a
> 6
>
> We prefix 1800, bring over the 555, bring over the 6, then 66 and
> bring over the .
>
> And get... 1800555666.
>
> Got it...
>
> Of course, reading them isn't the hard part... it is making them the
> first go round...
>
>
> Jonathan
>
> On Fri, Apr 4, 2008 at 8:35 PM, Edward French <[EMAIL PROTECTED]>
> wrote:
> >
> > Jonathan,
> >
> > The Cisco documentation on Voice translation sucks. Here is my
> explaination:
> >
> > first make a voice translation-rule
> > then apply the rule to a voice translation-profile
> > Then apply it to your dial-peer, voice-port.
> >
> > Ok so that is the part most people get here comes the tough part
> >
> > in its most basic form a translation rule converts a set of digits to
> > something else so
> >
> > /123/ /456/ would change 123 to 456 pretty simple so lets make it toughe
> >
> > /\(123/)/ /9\1/  this take set 1 and puts a 9 in front of it so the
> result
> > is 9123. To give a bit more detail anytime you have a set of \( \) it
> > creates a numbered "set" the first set of \( \) is set 1. the second set
> of
> > \( \) is set and so on. If something is inside of \( \) and is passed to
> the
> > translated pattern it is passed without modification. So to pass a set
> to
> > the translated number you have to call it the way you call it is \set
> > number. Anything not in a called set will be discarded.
> >
> > So a little tougher string
> >
> > /\(2\)\(4\)/ /\2\1/ would be 42  as you can see from this 2 is "set 1"
> and 4
> > is "set 2" in the result string we are requesting "set 2" then "set 1"
> >
> > So lets do something usefull
> >
> > /\(3...\)/ /331322\1/ converts a dn of 3002 to a full number of
> 3313223002
> > this is the translation rule needed for CME to CF to CUE with dial-plan
> in
> > use.
> >
> > Now for a ridiculous on:
> >
> > /^9\(555\)\(.\)77\(6\)/  /1800\1\3\66\2/
> > and we dial 95558775
> > the result is 18005556668 or
> > first we matched our pattern so start replacing and the we see the first
> > thing is to place 1800, then "set 1" 555, then "set 3" or 6, then 66,
> then
> > "set 2" or . which was input to the string as 8 when we put it all
> together
> > in order we get 18005556668.
> >
> > Hope this helps
> >
> > Ed
> >
> >
> >
> >
> >
> > - Original Message 
> > From: Jonathan Charles <[EMAIL PROTECTED]>
> > To: CCIE Voice Online Study List 
> > Sent: Friday, April 4, 2008 8:55:25 PM
> > Subject: [OSL | CCIE_Voice] Primer on voice translation rules...
> >
> >  OK, I think I am just really dumb here... I have gone thru the audio
> > and the video, I took copious notes, checked the Cisco site for these,
> > and I am still not getting it... I understand how the example works,
> > but I tried doing some stuff out of my head...
> >
> > Anyone have more resources for this?
> >
> >
> >
> > Jonathan
> >
> >
>

Re: [OSL | CCIE_Voice] MLP serialization delay

[jason sung]

we are going to need an expert opinion on this.

SRND says not to exceed 20 ms delay. but what if the the difference between
port speed and CIR is big that the delay value according to the calculations
go higher than 20 ms???

On Fri, Apr 4, 2008 at 11:13 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> What Jonathan wrote below about the fragment delay for 768Kbps is 10ms and
> for 384Kbps is 5ms are only true if those delays were based on the port
> speeds.
>
> However, I think Jason was asking for the fragment delay for a CIR of
> 384kbps and the port speed of 768kbps.  In this case, Jason's calculation of
> 20ms for fragment delay is correct.  Fragment delay is calculated based on
> port speed and NOT based on CIR.
>
> JD
>
>
> > Date: Thu, 3 Apr 2008 00:42:16 -0500
> > From: [EMAIL PROTECTED]
> > To: [EMAIL PROTECTED]
> > CC: ccie_voice@onlinestudylist.com
> > Subject: Re: [OSL | CCIE_Voice] MLP serialization delay
>
> >
> > Serialization delay is a function of how long it will take to get X
> > bits onto the wire at speed Y
> >
> > And is simply X/Y
> >
> > So, a 1500-byte ping will take 214ms to be put on the wire at a line
> > speed of 56 bps
> >
> > (1500*8)/56000 = .214
> >
> > So, you list the speed of the circuit 768000
> >
> > The fragment size ... 960
> >
> > So, (960*8)/768000 = 10 ms
> >
> > So, 384k would equal 5ms
> >
> >
> >
> > Jonathan
> >
> > On Wed, Apr 2, 2008 at 8:29 PM, jason sung <[EMAIL PROTECTED]> wrote:
> > > Can someone confirm my calculations? I am trying to calculate delay
> based on
> > > port speed and CIR.
> > >
> > > Port speed: 768
> > > CIR: 384
> > >
> > > fragment size = 768 *10/8 = 960
> > >
> > > delay = fragment size *8/ CIR
> > > = 960*8/384
> > > = 20 ms
> > >
> > > 20 ms sounds too high though
> > >
> > > TIA
>
>
> --
> Pack up or back up–use SkyDrive to transfer files or keep extra copies. Learn
> how.
>

Re: [OSL | CCIE_Voice] CUE primary extension to mailbox access

[jason sung]

I would avoid using dialplan pattern command for CME in the lab.

Instead just use translation profiles on the voice-port for ANI and DNIS.




On Thu, Apr 3, 2008 at 1:55 AM, Paul and Bobs <[EMAIL PROTECTED]> wrote:

> I am hitting the bug as explained above. I removed the dial-plan and it
> worked. I added the dialplan and changed the extension in CUE to OTHER
> (e.164 number) and it worked.
>
> Thanks guys
>
>
> On Thu, Apr 3, 2008 at 4:27 PM, Erick Bergquist <[EMAIL PROTECTED]>
> wrote:
>
> > Does the user in CUE have the right extension configured that the call
> > is coming from?  I've seen that be the issue before and changing it
> > and/or also setting the e.164 number to where it is coming from gets
> > it going. You could also briefly get rid of the dialplan pattern
> > command if it is doing something unexpected with manipulating the
> > number to get the user enrolled then put tit back in.
> >
> > On Wed, Apr 2, 2008 at 10:17 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> > >
> > http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_field_notice09186a008023cfe2.shtml
> > >
> > >  I don't think it applies...
> > >
> > >  But it might...
> > >
> > >
> > >
> > >  Jonathan
> > >
> > >
> > >
> > >
> > >
> > >
> > >  On Wed, Apr 2, 2008 at 9:36 PM, Paul and Bobs <[EMAIL PROTECTED]>
> > wrote:
> > >  > self enroll
> > >  >
> > >  > I see from another reply it could be a bug with the dial plan
> > >  >
> > >  > I will look into that
> > >  >
> > >  > Thanks
> > >  >
> > >  >
> > >  >
> > >  > On Thu, Apr 3, 2008 at 11:26 AM, Jonathan Charles <
> > [EMAIL PROTECTED]> wrote:
> > >  >
> > >  > > What do you mean by 'access the mailbox'? Do you mean try to
> > leave a
> > >  > > message or try to self-enroll?
> > >  > >
> > >  > >
> > >  > >
> > >  > > Jonathan
> > >  > >
> > >  > >
> > >  > >
> > >  > >
> > >  > > On undefined, Paul and Bobs <[EMAIL PROTECTED]> wrote:
> > >  > > > Hi
> > >  > > >
> > >  > > > I am having an issue with my CUE. Whenever I setup a new
> > mailbox for a
> > >  > user
> > >  > > > and try to access the mailbox it is saying that I can only
> > access it
> > >  > from
> > >  > > > the primary extension for security reasons thens hangs up.
> > >  > > >
> > >  > > > This happens on all new mailboxes.
> > >  > > >
> > >  > > > Paul
> > >  > > >
> > >  > >
> > >  >
> > >  >
> > >
> >
>
>

Re: [OSL | CCIE_Voice] IPMA AAR

[jason sung]

I can get IPMA to  work during AAR but that intercept key just does not
work. I look at the CCM traces but it shows nothing.

On Thu, Apr 3, 2008 at 8:24 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> That doesn't necessarily mean we might not be asked to do it...
> remember, test of worst practices...
>
> : )
>
> On Wed, Apr 2, 2008 at 12:41 PM, Mark Snow <[EMAIL PROTECTED]> wrote:
> > AAR and IPMA are inherently incompatible in CUCM 4.1 - as is AAR with
> ANY
> > CTI Route Points.
> >
> >  --
> >  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]
> >  --
> >  Join our free online support and peer group communities:
> > http://www.IPexpert.com/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.
> >  --
> >
> >
> >
> >  On Mar 31, 2008, at 10:29 PM, jason sung wrote:
> >
> >
> > > Has anyone been able to get the INTERCEPT key to work durring AAR on
> > IPMA??
> > >
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] MLP serialization delay

[jason sung]

When calculating delay don't you divide it by the CIR rather than port
speed.

Petr Lapukhov, CCIE #16379, says:

We need to calculate the fragment size for MLPPP. Since physical port speed
is 512Kpbs, and required serialization delay should not exceed 10ms
(remember, fragment size is based on physical port speed!!!), the fragment
size must be set to 512000/8*0.01=640 bytes. How is the fragment size
configured with MLPPP? By using command ppp multilink fragment delay -
however, IOS CLI takes this delay value (in milliseconds) and multiplies it
by configured interface (virtual-template) bandwidth (in our case 384Kbps).
We can actually change the virtual-template bandwidth to match the physical
interface speed, but this would affect the CBWFQ weights! Therefore, we take
the virtual-template bandwidth (384Kpbs) and adjust the delay to make sure
the fragment size matches the physical interace rate is 512Kbps. This way,
the "effective" delay value would be set to "640*8/*384* = 13ms"
(Fragment_Size/*CIR**8) to accomodate the physical and logical bandwidth
discrepancy. (This may be unimportant if our physical port speed does not
differ much from PVC CIR. However, if you have say PVC CIR=384Kbps and port
speed 768Kbps you may want to pay attention to this issue).




On Thu, Apr 3, 2008 at 12:42 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Serialization delay is a function of how long it will take to get X
> bits onto the wire at speed Y
>
> And is simply X/Y
>
> So, a 1500-byte ping will take 214ms to be put on the wire at a line
> speed of 56 bps
>
> (1500*8)/56000 = .214
>
> So, you list the speed of the circuit 768000
>
> The fragment size ... 960
>
> So, (960*8)/768000 = 10 ms
>
> So, 384k would equal 5ms
>
>
>
> Jonathan
>
> On Wed, Apr 2, 2008 at 8:29 PM, jason sung <[EMAIL PROTECTED]> wrote:
>  > Can someone confirm my calculations? I am trying to calculate delay
> based on
> > port speed and CIR.
> >
> > Port speed: 768
> > CIR: 384
> >
> > fragment size = 768 *10/8 = 960
> >
> > delay = fragment size *8/ CIR
> > = 960*8/384
> > = 20 ms
> >
> > 20 ms sounds too high though
> >
> > TIA
>

Re: [OSL | CCIE_Voice] MOH H323

[jason sung]

I did and the proof is that I was able to see multicast for g729.

On Wed, Apr 2, 2008 at 10:04 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

>  Did you have the command 'dcm-manager music-on-hold' ?
>
> You still need that one.
>
> Mark Snow Sr Technical Instructor
> IPexpert, Inc.
>
> Sent from my iPhone
>
> On Apr 2, 2008, at 9:23 PM, "jason sung" <[EMAIL PROTECTED]> wrote:
>
>   Not SRST, I meant an actual H323 gateway.
>
> Yes your assumption is correct, this is MoH to the PSTN PRI trunk.
>
> Here is what I did.
>
> Since it worked with MGCP, i configured siteB router as MGCP and pulled
> CCM trace.
>
> Next I wiped out MGCP and configured as H323 gateway to compare CCM
> traces, but now it works.
> My router configs were saved, I did not manually enter even a single
> command except NO SHUT the interfaces
>
>
>
>
> On Wed, Apr 2, 2008 at 8:05 PM, Mark Snow <[EMAIL PROTECTED]> wrote:
>
> > It wouldn't work - Multicast doesn't allow transcoders.
> > Are you in SRST fallback when you are stating that you are in H323?
> > Also I assume this is MoH to the PSTN PRI Trunk - since IP phones speak
> > SCCP not MGCP or H323 - am I correct on that assumption?
> >
> > Anyway - next test you would need to do is to run PerfMon on the
> > (correct) UCM (so most likely MoH server is registered to the Sub CPE if you
> > are using the Sub) and check to see when the music starts - does the MoH
> > PerfMon counter show that the music is coming from the Sub or Pub as either
> > a Multicast or Unicast stream ... if not - you have other things afoot!
> >
> > 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]
> > --
> > Join our free online support and peer group communities:
> > http://www.IPexpert.com/communities<http://www.ipexpert.com/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.
> > --
> >
> > On Apr 2, 2008, at 7:59 PM, jason sung wrote:
> >
> > Ok, I eliminated the 1st option by only allowing g711. (removed g729
> > under ip streaming services)
> >
> > 2nd option you mention. How would that happen?
> >
> >
> >
> > On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> >
> > > And we know from Mark Snow that the region setting only says 'this is
> > > the MAXIMUM bandwidth' we can use, so if it set to G.711, then G.729
> > > is still available for use.
> > >
> > > If you want it to use G.711 for MoH disable G.729 under the IP Voice
> > > Media Streaming App Service Parameter
> > >
> > > Or.
> > >
> > > Your H.323 destination is forcing G.729 so it is being transcoded...
> > > one of those...
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > > > I have a small problem.
> > > >
> > > > My multicast MOH works fine at the Remote site using MGCP, but fails
> > > to use
> > > > g711 at remote site when I use H323.
> > > >
> > > > CallManager configurations stay the same. When I switch from H323 to
> > > MGCP
> > > > everything works fine, if I switch back to H323, MOH works only for
> > > g729 and
> > > > not g711.
> > > >
> > > > For g711 I created MOH server device pool such that it will talk
> > > only g711
> > > > with remote site.
> > > >
> > > > any id
> > >
> >

Re: [OSL | CCIE_Voice] Shaping for FRF.12

[jason sung]

One reason you might want to use FRF.12 over legacy FRTS is because you can
fragment the packets.



On Wed, Apr 2, 2008 at 8:28 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> Does it matter if you use FRTS or legacy FRTS with FRF.12? Is there really
> any difference? Why would you use one over the other?
>
> --
> "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"

[OSL | CCIE_Voice] MLP serialization delay

[jason sung]

Can someone confirm my calculations? I am trying to calculate delay based on
port speed and CIR.

Port speed: 768
CIR: 384

fragment size = 768 *10/8 = 960

delay = fragment size *8/ CIR
= 960*8/384
= 20 ms

20 ms sounds too high though

TIA

Re: [OSL | CCIE_Voice] MOH H323

[jason sung]

Not SRST, I meant an actual H323 gateway.

Yes your assumption is correct, this is MoH to the PSTN PRI trunk.

Here is what I did.

Since it worked with MGCP, i configured siteB router as MGCP and pulled CCM
trace.

Next I wiped out MGCP and configured as H323 gateway to compare CCM traces,
but now it works.
My router configs were saved, I did not manually enter even a single command
except NO SHUT the interfaces




On Wed, Apr 2, 2008 at 8:05 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

> It wouldn't work - Multicast doesn't allow transcoders.
> Are you in SRST fallback when you are stating that you are in H323?
> Also I assume this is MoH to the PSTN PRI Trunk - since IP phones speak
> SCCP not MGCP or H323 - am I correct on that assumption?
>
> Anyway - next test you would need to do is to run PerfMon on the (correct)
> UCM (so most likely MoH server is registered to the Sub CPE if you are using
> the Sub) and check to see when the music starts - does the MoH PerfMon
> counter show that the music is coming from the Sub or Pub as either a
> Multicast or Unicast stream ... if not - you have other things afoot!
>
> 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]
> --
> Join our free online support and peer group communities:
> http://www.IPexpert.com/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.
> --
>
> On Apr 2, 2008, at 7:59 PM, jason sung wrote:
>
> Ok, I eliminated the 1st option by only allowing g711. (removed g729 under
> ip streaming services)
>
> 2nd option you mention. How would that happen?
>
>
>
> On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
>
> > And we know from Mark Snow that the region setting only says 'this is
> > the MAXIMUM bandwidth' we can use, so if it set to G.711, then G.729
> > is still available for use.
> >
> > If you want it to use G.711 for MoH disable G.729 under the IP Voice
> > Media Streaming App Service Parameter
> >
> > Or.
> >
> > Your H.323 destination is forcing G.729 so it is being transcoded...
> > one of those...
> >
> >
> >
> >
> > Jonathan
> >
> > On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > > I have a small problem.
> > >
> > > My multicast MOH works fine at the Remote site using MGCP, but fails
> > to use
> > > g711 at remote site when I use H323.
> > >
> > > CallManager configurations stay the same. When I switch from H323 to
> > MGCP
> > > everything works fine, if I switch back to H323, MOH works only for
> > g729 and
> > > not g711.
> > >
> > > For g711 I created MOH server device pool such that it will talk only
> > g711
> > > with remote site.
> > >
> > > any ideas??
> >
>
>
>

Re: [OSL | CCIE_Voice] CUE primary extension to mailbox access

[jason sung]

dialplan pattern bug. have you verified that???

On Wed, Apr 2, 2008 at 7:23 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:

> Hi
>
> I am having an issue with my CUE. Whenever I setup a new mailbox for a
> user and try to access the mailbox it is saying that I can only access it
> from the primary extension for security reasons thens hangs up.
>
> This happens on all new mailboxes.
>
> Paul
>

Re: [OSL | CCIE_Voice] MOH H323

[jason sung]

Region is set to 711,

streaming service is set to 711 and 729.

729 works no 711.

If i change the streaming service to 711 only, N0 MOH

On Wed, Apr 2, 2008 at 7:25 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Well, wait
>
> The region is set to 711-only, the MOH is set to 711-only, the
> voice-class codec is set to 711 as first preference, but 729 works...
>
> What is the region for the base voice stream?
>
>
>
> Jonathan
>
> On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > This is weird, I set the voice class to use 711 as 1st preference and
> both
> > codecs work
> >
> > I set it back to the way I had it g729 1st and 711 stops working.
> > I thought codec would be negotiated on voice-class codec. no?
> >
> > I am going to erase and do everything over.
> >
> >
> >
> > On Wed, Apr 2, 2008 at 7:02 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > A voice-class codec on the dial peer would do it, with no G.711 on
> > > it... What are the codec preferences?
> > >
> > > What is preference 1? Is it G.729a?
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > > > Ok, I eliminated the 1st option by only allowing g711. (removed g729
> > under
> > > > ip streaming services)
> > > >
> > > > 2nd option you mention. How would that happen?
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> > > >
> > > > > And we know from Mark Snow that the region setting only says 'this
> is
> > > > > the MAXIMUM bandwidth' we can use, so if it set to G.711, then
> G.729
> > > > > is still available for use.
> > > > >
> > > > > If you want it to use G.711 for MoH disable G.729 under the IP
> Voice
> > > > > Media Streaming App Service Parameter
> > > > >
> > > > > Or.
> > > > >
> > > > > Your H.323 destination is forcing G.729 so it is being
> transcoded...
> > > > > one of those...
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > > > > > I have a small problem.
> > > > > >
> > > > > > My multicast MOH works fine at the Remote site using MGCP, but
> fails
> > to
> > > > use
> > > > > > g711 at remote site when I use H323.
> > > > > >
> > > > > > CallManager configurations stay the same. When I switch from
> H323 to
> > > > MGCP
> > > > > > everything works fine, if I switch back to H323, MOH works only
> for
> > g729
> > > > and
> > > > > > not g711.
> > > > > >
> > > > > > For g711 I created MOH server device pool such that it will talk
> > only
> > > > g711
> > > > > > with remote site.
> > > > > >
> > > > > > any ideas??
> > > > >
> > > >
> > > >
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] MOH H323

[jason sung]

This is weird, I set the voice class to use 711 as 1st preference and both
codecs work

I set it back to the way I had it g729 1st and 711 stops working.
I thought codec would be negotiated on voice-class codec. no?

I am going to erase and do everything over.

On Wed, Apr 2, 2008 at 7:02 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> A voice-class codec on the dial peer would do it, with no G.711 on
> it... What are the codec preferences?
>
> What is preference 1? Is it G.729a?
>
>
>
>
>
> Jonathan
>
> On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > Ok, I eliminated the 1st option by only allowing g711. (removed g729
> under
> > ip streaming services)
> >
> > 2nd option you mention. How would that happen?
> >
> >
> >
> >
> >
> > On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > And we know from Mark Snow that the region setting only says 'this is
> > > the MAXIMUM bandwidth' we can use, so if it set to G.711, then G.729
> > > is still available for use.
> > >
> > > If you want it to use G.711 for MoH disable G.729 under the IP Voice
> > > Media Streaming App Service Parameter
> > >
> > > Or.
> > >
> > > Your H.323 destination is forcing G.729 so it is being transcoded...
> > > one of those...
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > > On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > > > I have a small problem.
> > > >
> > > > My multicast MOH works fine at the Remote site using MGCP, but fails
> to
> > use
> > > > g711 at remote site when I use H323.
> > > >
> > > > CallManager configurations stay the same. When I switch from H323 to
> > MGCP
> > > > everything works fine, if I switch back to H323, MOH works only for
> g729
> > and
> > > > not g711.
> > > >
> > > > For g711 I created MOH server device pool such that it will talk
> only
> > g711
> > > > with remote site.
> > > >
> > > > any ideas??
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] MOH H323

[jason sung]

yes preference 1 is g729

On Wed, Apr 2, 2008 at 7:02 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> A voice-class codec on the dial peer would do it, with no G.711 on
> it... What are the codec preferences?
>
> What is preference 1? Is it G.729a?
>
>
>
>
>
> Jonathan
>
> On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > Ok, I eliminated the 1st option by only allowing g711. (removed g729
> under
> > ip streaming services)
> >
> > 2nd option you mention. How would that happen?
> >
> >
> >
> >
> >
> > On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > And we know from Mark Snow that the region setting only says 'this is
> > > the MAXIMUM bandwidth' we can use, so if it set to G.711, then G.729
> > > is still available for use.
> > >
> > > If you want it to use G.711 for MoH disable G.729 under the IP Voice
> > > Media Streaming App Service Parameter
> > >
> > > Or.
> > >
> > > Your H.323 destination is forcing G.729 so it is being transcoded...
> > > one of those...
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > > On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > > > I have a small problem.
> > > >
> > > > My multicast MOH works fine at the Remote site using MGCP, but fails
> to
> > use
> > > > g711 at remote site when I use H323.
> > > >
> > > > CallManager configurations stay the same. When I switch from H323 to
> > MGCP
> > > > everything works fine, if I switch back to H323, MOH works only for
> g729
> > and
> > > > not g711.
> > > >
> > > > For g711 I created MOH server device pool such that it will talk
> only
> > g711
> > > > with remote site.
> > > >
> > > > any ideas??
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] MOH H323

[jason sung]

Ok, I eliminated the 1st option by only allowing g711. (removed g729 under
ip streaming services)

2nd option you mention. How would that happen?



On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> And we know from Mark Snow that the region setting only says 'this is
> the MAXIMUM bandwidth' we can use, so if it set to G.711, then G.729
> is still available for use.
>
> If you want it to use G.711 for MoH disable G.729 under the IP Voice
> Media Streaming App Service Parameter
>
> Or.
>
> Your H.323 destination is forcing G.729 so it is being transcoded...
> one of those...
>
>
>
>
> Jonathan
>
> On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> > I have a small problem.
> >
> > My multicast MOH works fine at the Remote site using MGCP, but fails to
> use
> > g711 at remote site when I use H323.
> >
> > CallManager configurations stay the same. When I switch from H323 to
> MGCP
> > everything works fine, if I switch back to H323, MOH works only for g729
> and
> > not g711.
> >
> > For g711 I created MOH server device pool such that it will talk only
> g711
> > with remote site.
> >
> > any ideas??
>

[OSL | CCIE_Voice] MOH H323

[jason sung]

I have a small problem.

My multicast MOH works fine at the Remote site using MGCP, but fails to use
g711 at remote site when I use H323.

CallManager configurations stay the same. When I switch from H323 to MGCP
everything works fine, if I switch back to H323, MOH works only for g729 and
not g711.

For g711 I created MOH server device pool such that it will talk only g711
with remote site.

any ideas??

Re: [OSL | CCIE_Voice] Lab restrictions

[jason sung]

Jonathan,

>From my one experience, I can say that there are no restrictions
universally, if there are any than it will be noted in the question.

On Tue, Apr 1, 2008 at 8:24 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> On the R&S lab you can't use static routes, etc.
>
> What restrictions are on the CCIE Voice lab? Can you not use the web
> interface to configure CallManager?
>
> No, seriously.
>
> Can you use ccm-manager config?
>
> That kinda stuff?
>
>
>
>
> Jonathan
>

[OSL | CCIE_Voice] IPMA AAR

[jason sung]

Has anyone been able to get the INTERCEPT key to work durring AAR on IPMA??

Re: [OSL | CCIE_Voice] Lab questions about RTP

[jason sung]

This is NDA. you are disclosing information :)

good to know. Thanks.

On Sun, Mar 30, 2008 at 3:06 PM, Jacob Owen <[EMAIL PROTECTED]> wrote:

> Edward,
> I probably know more about RTP than most people on
> this mailing list, my next voice attempt on 4/21 will
> be #9 for me in RTP (this is for all IE's not just
> Voice :) ).
>
> I believe you get 2 blank pieces of paper which you
> will have to return at the end of the day, i am sure
> you can request more if you need more.
>
> PC's seem to be plenty fast all the times I have been
> there, I think for voice the server response is more
> of an issue than PC would be.  I believe the screen
> resolution is at least 1280X1024 and if not that
> pretty close.  Only 1 monitor is provided.
>
> Their are 2-3 very, very large coolers in the break
> room about 150 feet from the lab that is stocked with
> just about every kind of soda (diet and regular) you
> could imagine, the bathrooms are about 1/2 to the
> break room on the right.  To leave the lab you need to
> grab a visitor badge hanging on the door and if
> someone else is already out you can still leave and
> just ring the doorbell to get back into the lab.
>
> The lunch is provided to you and you have to take it
> in a small conference room off of the main lab room.
> As for your low carb diet I am not sure, they really
> only offer 1 choice and it often contains meat and
> cheese (actually always I think) which I know some
> people won't eat for personal or religous reasons.  I
> am sure they will have something, and if you would
> like you could write [EMAIL PROTECTED] with ?'s and see
> if you can get a response.  I did see something once
> about if you had special requests you could ask, but I
> am not sure where that is at.  LIke I said, I could
> walk that room blindfolded so let me know if you or
> anyone else on this forum has any other questions.
>
>
> --- Edward French <[EMAIL PROTECTED]> wrote:
>
> > As I have said before I am making my first attempt
> > on April 18th. So I have a couple of questions about
> > the lab environment.
> >
> > Is there adequate space to notes? (I know on Cisco
> > provided sheets with Cisco proveded pens)
> >
> > Are the PC's reasonably fast?
> > Do the monitors have at least 1280 X 1024
> > resolution?
> > Do they have dual monitors?
> >
> > Is there access to beverages?
> >
> > Are the bathrooms close?
> >
> > Do you need money for lunch?
> > Is lunch cafeteria style? or a boxed lunch or what?
> > I am on a low carb diet, will I be able to get meat
> > and cheese?
> >
> >
> > Thanks
> >
> > Ed
> >
> >
> >
>
>
> Jacob Owen
> CCIE #14063 (R&S, Service Provider), CCVP, CCDP
>
>
>
>  
> 
> OMG, Sweet deal for Yahoo! users/friends:Get A Month of Blockbuster Total
> Access, No Cost. W00t
> http://tc.deals.yahoo.com/tc/blockbuster/text2.com
>

[OSL | CCIE_Voice] Fwd: bandwidth usage

[jason sung]

Mark,

can you please shed some light on this question.

Either I am asking someting so stupid nobody wants to answer OR I am asking
something impossible?

Basically I am trying to send few g711 calls and check the bandwidth and
than compare it with few g729 calls.

-- Forwarded message --
From: jason sung <[EMAIL PROTECTED]>
Date: Thu, Mar 27, 2008 at 9:25 PM
Subject: bandwidth usage
To: CCIE Maillist 


I have been trying different commands, but none of them give me a definative
answer on HOW TO CHECK BANDWIDTH USAGE on the router?


Does anybody have any ideas? I tried the show policy-map interface command
but that does not show me what I want.

Re: [OSL | CCIE_Voice] QOS marking on the router

[jason sung]

Thanks JD.

I think you are correct. I keep forgetting this basic logic. I was jotting
down small tips to read during the flight.


CUE works the same.
On Fri, Mar 28, 2008 at 3:37 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> If you mark the control traffic comming from the LAN (i.e. Cat6500
> switch), then you'll need to apply the policy on the FastEthernet trunk in
> the inbound direction on the router.  When you apply a policy to an
> interface, it is from the perspective of the router that the policy is being
> serviced.  In this case, the router sees the traffics comming into it, so it
> must be applied on the inbound.
>
> JD
>
>
>  --
> Date: Thu, 27 Mar 2008 21:50:12 -0500
> From: [EMAIL PROTECTED]
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] QOS marking on the router
>
> If I am trying to mark my control traffic on the router. Do I apply the
> policymap  to inbound or oubound side of the fastEthernet.
>
> I think outbound but I have heard people say inbound.
>
>
> --
> Watch "Cause Effect," a show about real people making a real difference. Learn
> more. 
>

[OSL | CCIE_Voice] QOS marking on the router

[jason sung]

If I am trying to mark my control traffic on the router. Do I apply the
policymap  to inbound or oubound side of the fastEthernet.

I think outbound but I have heard people say inbound.

[OSL | CCIE_Voice] bandwidth usage

[jason sung]

I have been trying different commands, but none of them give me a definative
answer on HOW TO CHECK BANDWIDTH USAGE on the router?


Does anybody have any ideas? I tried the show policy-map interface command
but that does not show me what I want.

Re: [OSL | CCIE_Voice] exchange server and AD cleanup for Unity

[jason sung]

Run the exchange clean up agent and then purge the mailboxes.

On Wed, Mar 26, 2008 at 11:51 AM, Juan Lopez Hernandez -X (jlopezhe - IBM -
INS at Cisco) <[EMAIL PROTECTED]> wrote:

>  Is there a nice way to clean up mailbox leftovers on the exchange system?
> I used the 'Bulk Subscriber Delete' from the Unity toolbox, but this only
> deletes Unity subscribers' mailboxes that are defined in the Unity DB. Is
> there a way to clean up the message store completely, as I get a 'system
> error' every time I want to leave a message and I think it's related to
> adding new users on Unity, whose settings conflict with leftover mailbox
> information.
>
> What tool could I use to delete all mailboxes on exchange and all (unity)
> accounts in AD?
>
> cheers,
> Juan
>
>
>
>

Re: [OSL | CCIE_Voice] bandwidth usage

[jason sung]

I have an example here, but I can not confirm what field tells me the
bandwidth usage?

BR1#show policy-map interface ser0/0.65  output
 Serial0/0.20: DLCI 201 -
  Service-policy output: vats
Class-map: class-default (match-any)
  5382 packets, 345196 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: any
  Traffic Shaping
   Target/Average   Byte   Sustain   ExcessInterval  Increment
 Rate   Limit  bits/int  bits/int  (ms)  (bytes)
   729600/7296004563648  0 5 456
Adapt  Queue Packets   Bytes Packets   Bytes Shaping
Active Depth Delayed   Delayed   Active
BECN   0 5382  3451960 0 no
Voice Adaptive Shaping inactive
  Service-policy : llq
Class-map: rtp (match-any)
  4496 packets, 287744 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: ip dscp ef (46)
4496 packets, 287744 bytes
5 minute rate 0 bps
  Queueing
Strict Priority
Output Queue: Conversation 40
Bandwidth 33 (%)
Bandwidth 120 (kbps) Burst 3000 (Bytes)
(pkts matched/bytes matched) 4496/287744
(total drops/bytes drops) 0/0
Class-map: sig (match-any)
  346 packets, 19032 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: ip dscp cs3 (24)
346 packets, 19032 bytes
5 minute rate 0 bps
  Queueing
Output Queue: Conversation 41
Bandwidth 2 (%)
Bandwidth 7 (kbps) Max Threshold 64 (packets)
(pkts matched/bytes matched) 346/19032
(depth/total drops/no-buffer drops) 0/0/0

Class-map: class-default (match-any)
  540 packets, 38420 bytes
  5 minute offered rate 0 bps, drop rate 0 bps
  Match: any
  Queueing
Flow Based Fair Queueing
Maximum Number of Hashed Queues 32
(total queued/total drops/no-buffer drops) 0/0/0

On Wed, Mar 26, 2008 at 10:00 AM, Onur Tufekci <[EMAIL PROTECTED]>
wrote:

> show policy-map interface "fast0/0" should can you confirm?
>
>
> On Wed, Mar 26, 2008 at 10:55 AM, jason sung <[EMAIL PROTECTED]> wrote:
>
> >
> > How can I check bandwidth usage on a policy map?
> >
> > For example I want to send a regular g729 call and then I want to send a
> > compressed g729 call.
> >
> > Basically I want to compare the bandwidth usage between the two.
> >
> > TIA.
> >
>
>

[OSL | CCIE_Voice] bandwidth usage

[jason sung]

How can I check bandwidth usage on a policy map?

For example I want to send a regular g729 call and then I want to send a
compressed g729 call.

Basically I want to compare the bandwidth usage between the two.

TIA.

[OSL | CCIE_Voice] checksum question

[jason sung]

I know both TCP and UDP communication layer provide checksum count and
verification as one of their service. So if we compress RTP is it safe to
assume that the header size is reduced to 4 bytes?

I think so but a verification stamp from someone would be good.

Thanks.

[OSL | CCIE_Voice] WAN QOS question

[jason sung]

I am trying to understand QOS over WAN and want to make sure my math is
correct as well.

I am trying to calculate 4 g711/g729 calls over a WAN link with layer2
overhead.

  g711 g729 PPP 84x4=336 37x4=148 MLP 86x4=344 30x4=120 Frame w/FRF.12
84x4=336 28x4=112

Can anyone comment on this?

Thanks in advance.

Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - movedfromUnivercd ??

[jason sung]

SJ.
I have heard good things about RTP as well.

On Fri, Mar 14, 2008 at 1:26 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Were you in SJ or RTP?
>
>
> J
>
> On Fri, Mar 14, 2008 at 1:26 PM, jason sung <[EMAIL PROTECTED]> wrote:
> > CCO was fine, server responses were fine too.
> >
> > Nothing wrong with the lab. All rumors...
> >
> >
> >
> >
> >
> >
> > On Fri, Mar 14, 2008 at 1:21 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > Oh, also... got a question about the lab (no NDA stuff...)
> > >
> > > I have heard that the CCM response is incredibly slow, taking minutes
> > > to load a page... is this true, and does this constitute a hardware
> > > failure?
> > >
> > > I have CCM/IPCC running on a VM and it is very fast... what is the
> > > problem on the lab?
> > >
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On Fri, Mar 14, 2008 at 11:05 AM, Jane Ryer (jryer) <[EMAIL PROTECTED]>
> > wrote:
> > >
> > >
> > >
> > > > I sat the lab for the first time in RTP this past Tuesday, March
> 11th.
> > > >  (I did not pass - not an unexpected outcome, but still
> disappointing.)
> > > >
> > > >  In addition to the CCM and QoS SRND's on the desktop, there is now
> also
> > > >  a .PDF version of the Cisco CAD Installation Guide, CAD 6.1 for IP
> > > >  Contact Center Express Edition Release 4.0.  I am pretty sure that
> it
> > > >  was the file found at this url:
> > > >
> > http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_cente
> > > >  r/crs/express_4_0/installation/for_cad/cad611ig.pdf
> > > >
> > > >  It does include the url for the IP Phone Agent service.
> > > >
> > > >  The redirect links worked fine for everything else I tried.
> > > >
> > > >  Jane Ryer
> > > >  R&S CCIE # 
> > > >  Network Consulting Engineer
> > > >  Cisco Systems Advanced Services team
> > > >
> > > >
> > > >
> > > >  -Original Message-
> > > >  From: [EMAIL PROTECTED]
> > > >
> > > >
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of Mark
> Snow
> > > >  Sent: Tuesday, February 26, 2008 7:10 PM
> > > >  To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)
> > > >  Cc: ccie_voice@onlinestudylist.com; Mike Prestidge
> > > >  Subject: Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL -
> > > >  movedfromUnivercd ??
> > > >
> > > >  CME is still here:
> > > >  http://www.cisco.com/univercd/cc/td/doc/product/voice/its/index.htm
> > > >  Just have to poke around to find it.
> > > >
> > > >  But the UCCX - That is disturbing.
> > > >  I am looking into it.
> > > >
> > > >
> > > >  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 Feb 26, 2008, at 5:29 PM, Juan Lopez Hernandez -X (jlopezhe -
> IBM -
> > > >  INS at Cisco) wrote:
> > > >
> > > >  > The same goes for all CME related docs :-( , since last night...
> > > >  >
> > > >  > Juan
> > > >  >
> > > >  > -Original Message-
> > > >  > From: [EMAIL PROTECTED]
> > > >  > [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> > > >  > Prestidge
> > > >  > Sent: Tuesday, February 26, 2008 11:23 PM
> > > >  > To: ccie_voice@onlinestudylist.com
> > > >  > Subject: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved
> > > >  > fromUnivercd ??
> > > >  >
> > > >  > It seems that the (annoying) people moving the documents off
> Univercd
> > > >  > have now also moved the documents with the URL for IP Phone
> agents!!
> > > >  >
> > > >  > I used to be able to find this by browsing via the following:
> > > >  >
> > > >  > Univercd > Customer Contact Software > IPCC Express and IP IVR >
> CRS
> > > >  > 5.0(x) > English > Documentation for Cisco IP Agents > Cisco CAD
> > > >  > Installation Guide 6.4
> > > >  >
> > > >  > Now this documentation has also been moved to a link that is not
> > > >  > available in the lab.  Does anyone know an alternative location
> to
> > > >  > find
> > > >  > the URL within Univercd?
> > > >  >
> > > >  > Mike
> > > >  >
> > > >  > This communication, including any attachments, is confidential.
> If
> > you
> > > >  > are not the intended recipient, you should not read it - please
> > > >  > contact
> > > >  > me immediately, destroy it, and do not copy or use any part of
> this
> > > >  > communication or disclose anything about it. Thank you. Please
> note
> > > >  > that
> > > >  > this communication does not designate an information system for
> the
> > > >  > purposes of the Electronic Transactions Act 2002.
> > > >  >
> > > >
> > > >
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - movedfromUnivercd ??

[jason sung]

CCO was fine, server responses were fine too.

Nothing wrong with the lab. All rumors...




On Fri, Mar 14, 2008 at 1:21 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Oh, also... got a question about the lab (no NDA stuff...)
>
> I have heard that the CCM response is incredibly slow, taking minutes
> to load a page... is this true, and does this constitute a hardware
> failure?
>
> I have CCM/IPCC running on a VM and it is very fast... what is the
> problem on the lab?
>
>
>
>
> Jonathan
>
> On Fri, Mar 14, 2008 at 11:05 AM, Jane Ryer (jryer) <[EMAIL PROTECTED]>
> wrote:
>  > I sat the lab for the first time in RTP this past Tuesday, March 11th.
> >  (I did not pass - not an unexpected outcome, but still disappointing.)
> >
> >  In addition to the CCM and QoS SRND's on the desktop, there is now also
> >  a .PDF version of the Cisco CAD Installation Guide, CAD 6.1 for IP
> >  Contact Center Express Edition Release 4.0.  I am pretty sure that it
> >  was the file found at this url:
> >
> http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_cente
> >  r/crs/express_4_0/installation/for_cad/cad611ig.pdf
> >
> >  It does include the url for the IP Phone Agent service.
> >
> >  The redirect links worked fine for everything else I tried.
> >
> >  Jane Ryer
> >  R&S CCIE # 
> >  Network Consulting Engineer
> >  Cisco Systems Advanced Services team
> >
> >
> >
> >  -Original Message-
> >  From: [EMAIL PROTECTED]
> >
> >
> > [mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow
> >  Sent: Tuesday, February 26, 2008 7:10 PM
> >  To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)
> >  Cc: ccie_voice@onlinestudylist.com; Mike Prestidge
> >  Subject: Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL -
> >  movedfromUnivercd ??
> >
> >  CME is still here:
> >  http://www.cisco.com/univercd/cc/td/doc/product/voice/its/index.htm
> >  Just have to poke around to find it.
> >
> >  But the UCCX - That is disturbing.
> >  I am looking into it.
> >
> >
> >  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 Feb 26, 2008, at 5:29 PM, Juan Lopez Hernandez -X (jlopezhe - IBM -
> >  INS at Cisco) wrote:
> >
> >  > The same goes for all CME related docs :-( , since last night...
> >  >
> >  > Juan
> >  >
> >  > -Original Message-
> >  > From: [EMAIL PROTECTED]
> >  > [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> >  > Prestidge
> >  > Sent: Tuesday, February 26, 2008 11:23 PM
> >  > To: ccie_voice@onlinestudylist.com
> >  > Subject: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved
> >  > fromUnivercd ??
> >  >
> >  > It seems that the (annoying) people moving the documents off Univercd
> >  > have now also moved the documents with the URL for IP Phone agents!!
> >  >
> >  > I used to be able to find this by browsing via the following:
> >  >
> >  > Univercd > Customer Contact Software > IPCC Express and IP IVR > CRS
> >  > 5.0(x) > English > Documentation for Cisco IP Agents > Cisco CAD
> >  > Installation Guide 6.4
> >  >
> >  > Now this documentation has also been moved to a link that is not
> >  > available in the lab.  Does anyone know an alternative location to
> >  > find
> >  > the URL within Univercd?
> >  >
> >  > Mike
> >  >
> >  > This communication, including any attachments, is confidential. If
> you
> >  > are not the intended recipient, you should not read it - please
> >  > contact
> >  > me immediately, destroy it, and do not copy or use any part of this
> >  > communication or disclose anything about it. Thank you. Please note
> >  > that
> >  > this communication does not designate an information system for the
> >  > purposes of the Electronic Transactions Act 2002.
> >  >
> >
> >
>

Re: [OSL | CCIE_Voice] Dialplan pattern

[jason sung]

Thanks Mark,

I was almost positive that I wouldn't need anything more than the two rules
you mentioned, but just wanted a vote of confirmation.

Thanks again.

On Sat, Mar 8, 2008 at 11:42 AM, Mark Snow <[EMAIL PROTECTED]> wrote:

> Ahhh - you will miss the headaches and all the aspirin you need to
> take - as they disappear from NOT using DP Pattern :-)
>
> Nope that is it - and here is a great substitution for doing just that:
>
>
> !
>
> 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
> !
>
>
> --
> 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]
> --
> Join our free online support and peer group communities:
> http://www.IPexpert.com/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.
> --
>
> On Mar 7, 2008, at 6:21 PM, jason sung wrote:
>
> > So I have decided not to use dialplan pattern on the CME/CUE box.
> >
> > Can somebody confirm if I will be missing anything else besides the
> > following?
> >
> > full 10 digit caller-id going out.
> > incoming 10 digit to 4-digit extension translation.
> >
> > Thanks in advance.
>
>

[OSL | CCIE_Voice] Dialplan pattern

[jason sung]

So I have decided not to use dialplan pattern on the CME/CUE box.

Can somebody confirm if I will be missing anything else besides the
following?

full 10 digit caller-id going out.
incoming 10 digit to 4-digit extension translation.

Thanks in advance.

Re: [OSL | CCIE_Voice] xlation pattern vs routepattern

[jason sung]

Best advise I can give you is check out Dialed Number analyzer

I do not know if there is a document that explains this but DNA should give
you the answer.

On Fri, Mar 7, 2008 at 9:51 AM, DSCP46EF <[EMAIL PROTECTED]> wrote:

> Guys,
>
> What is the sequence of call hits to/from CCM?
> Note: application dialplan rules is under CCM>RoutePlan>Application -dialplan
> rules
>
> Situation1: Call from CCM to PSTN
> 
> dialed-digit-->RoutePattern>Translation-Pattern>applicationDialPlanRule
>
> Q1: is it okay?
> Q2: Does significent digit (normallly we put 4) play any rule in this
> situation?
>
> Situation2: Call from PSTN to CCM
> --
> Dialed-digit> significent-digit-->Translation-Pattern>
> Q1: Is above call processing sequence okay?
> Q2: does RoutePattern &  applicatoin dialplan-rules play any rules in
> this scenario?
>
> PS: I tried to configure Applicatoin dialplan rules but it doesn't work.
> May be this is only for @ dialplan?
>
>
>   Dial Rules Configuration Status:  Ready
> Note: To create a Dial Rule, set the values for the attributes and click
> Insert
> To modify a rule or to sort the rules, change the settings and click
> Update  Dial Rule Creation *If the phone number begins with  and the 
> number
> of digits is , then remove
> digits from the beginning and prefix it with .*
>   No dial rules found
>
>
>
> --
> Smile, you'll save someone else's day!
> Frog

[OSL | CCIE_Voice] B-ACD issue

[jason sung]

Hello, All

I am trying to send an incoming call from the B-ACD queue to voicemail. here
is my config

application
 service queue flash:app-b-acd-2.1.0.0.tcl
  param aa-hunt3 3004
  param queue-len 10
  param queue-manager-debugs 1
  param aa-hunt2 3210
  param number-of-hunt-grps 2
 !
 service aa flash:app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 1
  param number-of-hunt-grps 2
  param dial-by-extension-option 5
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 3017773000
  param max-extension-length 4
  paramspace english location flash:
  param second-greeting-time 30
  param welcome-prompt _bacd_options_menu.au
  param call-retry-timer 15
  param max-time-call-retry 60
  param voice-mail 53210
  param service-name queue
 !
voice translation-rule 2
 rule 1 /^53210$/ /3210/

voice translation-profile REDIRECT
 translate calling 2
 translate called 2
 translate redirect-called 2

ephone-dn  8
 number 53210 no-reg both
 call-forward all 3600
!

dial-peer voice 36 voip
 translation-profile outgoing REDIRECT
 destination-pattern 3600
 session protocol sipv2
 session target ipv4:10.2.120.2
 incoming called-number 399[89]
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 37 voip
 translation-profile outgoing VM
 destination-pattern 3017773[126]..
 session protocol sipv2
 session target ipv4:10.2.120.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad


So the issue I am having is when my call gets forwarded to voicemail it is
dialing 3017773600 and thus matches dial-peer 37, as soon as I take dialplan
pattern off from the telephony-service, call matches dial-peer 36 inturn
matches the voice translation-rule and life is good.

Really stuck on this one. Any help will be appreciated

Re: [OSL | CCIE_Voice] Music on Hold questions

[jason sung]

Ok, makes sense.

Thanks Jonathan.

On Tue, Mar 4, 2008 at 8:44 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> No, dead air...
>
>
> Jonathan
>
> On Tue, Mar 4, 2008 at 8:44 PM, jason sung <[EMAIL PROTECTED]> wrote:
> > So if there is no music source on the flash. Users hear tone???
> >
> >
> >
> > On Tue, Mar 4, 2008 at 8:40 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > No, the phone is subscribing to the multicast IP address, when it
> > > does, it picks up the router's stream...
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > > On Tue, Mar 4, 2008 at 8:04 PM, jason sung <[EMAIL PROTECTED]> wrote:
> > > > So it does not pick up the music source on flash, but it actually
> > streams
> > > > across CCM?
> > > >
> > > >
> > > >
> > > >
> > > > On Tue, Mar 4, 2008 at 6:43 PM, Mark Snow <[EMAIL PROTECTED]>
> wrote:
> > > >
> > > > > Exactly - buy to further answer your question - the IP phone never
> > > > > "knows" to pick up MoH from the router locally - instaed the UCM
> tells
> > > > > the phone to join a specific multicast group number and the phone
> does
> > > > > so ... it just 'happens' that the router is serving up that stream
> and
> > > > > not routing the multicast traffic back to the UCM MoH server.
> > > > >
> > > > > HTH,
> > > > >
> > > > > Mark Snow
> > > > > Sr Technical Instructor
> > > > > IPexpert, Inc.
> > > > >
> > > > > Sent from my iPhone
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On Mar 4, 2008, at 3:28 PM, "Ovais Iqbal" <[EMAIL PROTECTED]>
> > wrote:
> > > > >
> > > > > > If device pool is also set to none for MRGL then it will look
> for
> > > > > > any MOH resourse available which is not a member of any MRG,
> means
> > > > > > left out side.
> > > > > >
> > > > > >
> > > > > > Ovais Iqbal
> > > > > > 416-294-7869
> > > > > > Sent from my BlackBerry device
> > > > > >
> > > > > > -Original Message-
> > > > > > From: "Jonathan Charles" <[EMAIL PROTECTED]>
> > > > > >
> > > > > > Date: Tue, 4 Mar 2008 13:02:13
> > > > > > To:"jason sung" <[EMAIL PROTECTED]>
> > > > > > Cc:Mark Snow <[EMAIL PROTECTED]>,CCIE Maillist
> > > >  > > > > > >
> > > > > > Subject: Re: [OSL | CCIE_Voice] Music on Hold questions
> > > > > >
> > > > > >
> > > > > > Out of the MRGL on its device pool...
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > > On Tue, Mar 4, 2008 at 10:39 AM, jason sung <[EMAIL PROTECTED]>
> > > > > > wrote:
> > > > > >> Thanks for your response, also Jonathan.
> > > > > >>
> > > > > >> I am little confused about how an MGCP gateway picks up MOH
> source.
> > > > > >> If the
> > > > > >> MRGL on the gateway configuration is NONE, how does it know to
> pick
> > > > > >> up music
> > > > > >> source from the router???
> > > > > >>
> > > > > >> Thanks.
> > > > > >>
> > > > > >>
> > > > > >>
> > > > > >> On Mon, Mar 3, 2008 at 10:00 PM, Mark Snow <[EMAIL PROTECTED]>
> > > > > >> wrote:
> > > > > >>
> > > > > >>> Performance Monitor is your friend here.
> > > > > >>>
> > > > > >>> 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 Mar 3, 2008, at 1:02 PM, jason sung wrote:
> > > > > >>>
> > > > > >>>> I had two questions.
> > > > > >>>>
> > > > > >>>> 1. How do I verify if the Multicast MOH is playing from pub
> or
> > sub?
> > > > > >>>> I have configured publisher server as MOH. Publisher PerfMon
> > shows
> > > > > >>>> nothing, while subscriber PerfMon shows multicast music
> playing.
> > Is
> > > > > >>>> this because phone is registered to subscriber?
> > > > > >>>>
> > > > > >>>> 2. how do i check multicast moh playing from router?
> > > > > >>>>
> > > > > >>>>
> > > > > >>>> Thanks in advance for replies...
> > > > > >>>
> > > > > >>>
> > > > > >>
> > > > > >>
> > > > >
> > > >
> > > >
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] Music on Hold questions

[jason sung]

So if there is no music source on the flash. Users hear tone???

On Tue, Mar 4, 2008 at 8:40 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> No, the phone is subscribing to the multicast IP address, when it
> does, it picks up the router's stream...
>
>
> Jonathan
>
> On Tue, Mar 4, 2008 at 8:04 PM, jason sung <[EMAIL PROTECTED]> wrote:
> > So it does not pick up the music source on flash, but it actually
> streams
> > across CCM?
> >
> >
> >
> >
> > On Tue, Mar 4, 2008 at 6:43 PM, Mark Snow <[EMAIL PROTECTED]> wrote:
> >
> > > Exactly - buy to further answer your question - the IP phone never
> > > "knows" to pick up MoH from the router locally - instaed the UCM tells
> > > the phone to join a specific multicast group number and the phone does
> > > so ... it just 'happens' that the router is serving up that stream and
> > > not routing the multicast traffic back to the UCM MoH server.
> > >
> > > HTH,
> > >
> > > Mark Snow
> > > Sr Technical Instructor
> > > IPexpert, Inc.
> > >
> > > Sent from my iPhone
> > >
> > >
> > >
> > >
> > > On Mar 4, 2008, at 3:28 PM, "Ovais Iqbal" <[EMAIL PROTECTED]>
> wrote:
> > >
> > > > If device pool is also set to none for MRGL then it will look for
> > > > any MOH resourse available which is not a member of any MRG, means
> > > > left out side.
> > > >
> > > >
> > > > Ovais Iqbal
> > > > 416-294-7869
> > > > Sent from my BlackBerry device
> > > >
> > > > -Original Message-
> > > > From: "Jonathan Charles" <[EMAIL PROTECTED]>
> > > >
> > > > Date: Tue, 4 Mar 2008 13:02:13
> > > > To:"jason sung" <[EMAIL PROTECTED]>
> > > > Cc:Mark Snow <[EMAIL PROTECTED]>,CCIE Maillist
> >  > > > >
> > > > Subject: Re: [OSL | CCIE_Voice] Music on Hold questions
> > > >
> > > >
> > > > Out of the MRGL on its device pool...
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On Tue, Mar 4, 2008 at 10:39 AM, jason sung <[EMAIL PROTECTED]>
> > > > wrote:
> > > >> Thanks for your response, also Jonathan.
> > > >>
> > > >> I am little confused about how an MGCP gateway picks up MOH source.
> > > >> If the
> > > >> MRGL on the gateway configuration is NONE, how does it know to pick
> > > >> up music
> > > >> source from the router???
> > > >>
> > > >> Thanks.
> > > >>
> > > >>
> > > >>
> > > >> On Mon, Mar 3, 2008 at 10:00 PM, Mark Snow <[EMAIL PROTECTED]>
> > > >> wrote:
> > > >>
> > > >>> Performance Monitor is your friend here.
> > > >>>
> > > >>> 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 Mar 3, 2008, at 1:02 PM, jason sung wrote:
> > > >>>
> > > >>>> I had two questions.
> > > >>>>
> > > >>>> 1. How do I verify if the Multicast MOH is playing from pub or
> sub?
> > > >>>> I have configured publisher server as MOH. Publisher PerfMon
> shows
> > > >>>> nothing, while subscriber PerfMon shows multicast music playing.
> Is
> > > >>>> this because phone is registered to subscriber?
> > > >>>>
> > > >>>> 2. how do i check multicast moh playing from router?
> > > >>>>
> > > >>>>
> > > >>>> Thanks in advance for replies...
> > > >>>
> > > >>>
> > > >>
> > > >>
> > >
> >
> >
>

Re: [OSL | CCIE_Voice] Music on Hold questions

[jason sung]

So it does not pick up the music source on flash, but it actually streams
across CCM?

On Tue, Mar 4, 2008 at 6:43 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

> Exactly - buy to further answer your question - the IP phone never
> "knows" to pick up MoH from the router locally - instaed the UCM tells
> the phone to join a specific multicast group number and the phone does
> so ... it just 'happens' that the router is serving up that stream and
> not routing the multicast traffic back to the UCM MoH server.
>
> HTH,
>
> Mark Snow
> Sr Technical Instructor
> IPexpert, Inc.
>
> Sent from my iPhone
>
> On Mar 4, 2008, at 3:28 PM, "Ovais Iqbal" <[EMAIL PROTECTED]> wrote:
>
> > If device pool is also set to none for MRGL then it will look for
> > any MOH resourse available which is not a member of any MRG, means
> > left out side.
> >
> >
> > Ovais Iqbal
> > 416-294-7869
> > Sent from my BlackBerry device
> >
> > -Original Message-
> > From: "Jonathan Charles" <[EMAIL PROTECTED]>
> >
> > Date: Tue, 4 Mar 2008 13:02:13
> > To:"jason sung" <[EMAIL PROTECTED]>
> > Cc:Mark Snow <[EMAIL PROTECTED]>,CCIE Maillist <
> ccie_voice@onlinestudylist.com
> > >
> > Subject: Re: [OSL | CCIE_Voice] Music on Hold questions
> >
> >
> > Out of the MRGL on its device pool...
> >
> >
> >
> > Jonathan
> >
> > On Tue, Mar 4, 2008 at 10:39 AM, jason sung <[EMAIL PROTECTED]>
> > wrote:
> >> Thanks for your response, also Jonathan.
> >>
> >> I am little confused about how an MGCP gateway picks up MOH source.
> >> If the
> >> MRGL on the gateway configuration is NONE, how does it know to pick
> >> up music
> >> source from the router???
> >>
> >> Thanks.
> >>
> >>
> >>
> >> On Mon, Mar 3, 2008 at 10:00 PM, Mark Snow <[EMAIL PROTECTED]>
> >> wrote:
> >>
> >>> Performance Monitor is your friend here.
> >>>
> >>> 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 Mar 3, 2008, at 1:02 PM, jason sung wrote:
> >>>
> >>>> I had two questions.
> >>>>
> >>>> 1. How do I verify if the Multicast MOH is playing from pub or sub?
> >>>> I have configured publisher server as MOH. Publisher PerfMon shows
> >>>> nothing, while subscriber PerfMon shows multicast music playing. Is
> >>>> this because phone is registered to subscriber?
> >>>>
> >>>> 2. how do i check multicast moh playing from router?
> >>>>
> >>>>
> >>>> Thanks in advance for replies...
> >>>
> >>>
> >>
> >>
>

Re: [OSL | CCIE_Voice] Music on Hold questions

[jason sung]

Thanks for your response, also Jonathan.

I am little confused about how an MGCP gateway picks up MOH source. If the
MRGL on the gateway configuration is NONE, how does it know to pick up music
source from the router???

Thanks.

On Mon, Mar 3, 2008 at 10:00 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

> Performance Monitor is your friend here.
>
> 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 Mar 3, 2008, at 1:02 PM, jason sung wrote:
>
> > I had two questions.
> >
> > 1. How do I verify if the Multicast MOH is playing from pub or sub?
> > I have configured publisher server as MOH. Publisher PerfMon shows
> > nothing, while subscriber PerfMon shows multicast music playing. Is
> > this because phone is registered to subscriber?
> >
> > 2. how do i check multicast moh playing from router?
> >
> >
> > Thanks in advance for replies...
>
>

[OSL | CCIE_Voice] Music on Hold questions

[jason sung]

I had two questions.

1. How do I verify if the Multicast MOH is playing from pub or sub?
I have configured publisher server as MOH. Publisher PerfMon shows nothing,
while subscriber PerfMon shows multicast music playing. Is this because
phone is registered to subscriber?

2. how do i check multicast moh playing from router?


Thanks in advance for replies...

[OSL | CCIE_Voice] Gatekeeper Bandwidth

[jason sung]

When I have an active call from BR2-CME to CallManager. Why does the current
interzone bandwidth value empty on the CCM-GK side???

When I place a call from CCM to CME, Both the zones CCM-GK and BR2-CME show
interzone bandwidth value.


HQ#sh gatek zone status
 GATEKEEPER ZONES
 
GK name  Domain Name   RAS Address PORT  FLAGS
---  ---   --- - -
CCM-GK   ccie.com  10.1.110.1  1719  LS
  BANDWIDTH INFORMATION (kbps) :
Maximum total bandwidth : unlimited
Current total bandwidth : 0
Maximum interzone bandwidth : 128
*Current interzone bandwidth : 0*  * < This shows zero*
Maximum session bandwidth : unlimited
  SUBNET ATTRIBUTES :
All Other Subnets : (Enabled)
  PROXY USAGE CONFIGURATION :
Inbound Calls from all other zones :
  to terminals in local zone CCM-GK : use proxy
  to gateways in local zone CCM-GK  : do not use proxy
  to MCUs in local zone CCM-GK  : do not use proxy
Outbound Calls to all other zones :
  from terminals in local zone CCM-GK : use proxy
  from gateways in local zone CCM-GK  : do not use proxy
  from MCUs in local zone CCM-GK  : do not use proxy

BR2-CME  ccie.com  10.1.110.1  1719  LS
  BANDWIDTH INFORMATION (kbps) :
Maximum total bandwidth : unlimited
Current total bandwidth : 16
Maximum interzone bandwidth : 128
*Current interzone bandwidth : 16* *  This shows 16K*
Maximum session bandwidth : unlimited
Total number of concurrent calls : 1
  SUBNET ATTRIBUTES :
All Other Subnets : (Enabled)
  PROXY USAGE CONFIGURATION :
Inbound Calls from all other zones :
  to terminals in local zone BR2-CME : use proxy
  to gateways in local zone BR2-CME  : do not use proxy
  to MCUs in local zone BR2-CME  : do not use proxy
Outbound Calls to all other zones :
  from terminals in local zone BR2-CME : use proxy
  from gateways in local zone BR2-CME  : do not use proxy
  from MCUs in local zone BR2-CME  : do not use proxy

HQ#sh run | b gatek
gatekeeper
 zone local CCM-GK ccie.com 10.1.110.1
 zone local BR2-CME ccie.com
 zone prefix CCM-GK 2...
 zone prefix BR2-CME 3...
 zone prefix CCM-GK 4...
 bandwidth interzone default 128
 no shutdown

[OSL | CCIE_Voice] ATA fax passthrough

[jason sung]

What is the correct passthrough setting for ATA?

IPexpert workbook has AudioMode - 0x00140014 and ConnectMode - 0x0400

and

CCO has AudioMode - 00150015 and ConnectMode- 0x9400


http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/sccp3_0/sccpaapd.htm

Thanks.

Re: [OSL | CCIE_Voice] AAR issue

[jason sung]

Here are the details,

AAR Group: HQ and BR1, from HQ's perspective prefix 91 and same from BR1
side.

Location bandwidth was set to 23 on BR1 side

Yes, external mask was specified.

CCM service parameter for AAR was set to TRUE and restarted CCM service.

Created seperate partitions and calling search space for AAR

I pulled a callmanager trace and did see that my call from HQ picks up the
external mask and prefixes 91 and I also see the partiton and CSS match.

Only reason I can think of is may be my IP communicator???

On Tue, Feb 26, 2008 at 9:10 AM, Mark Snow <[EMAIL PROTECTED]> wrote:

> Please include a few more specifics:
>
> - AAR Group and what the Prefix is for both Groups (Groups should/must
> differ from Phone in SiteA and Phone in SiteB)
> - Please note what Location bandwidth you have for both Phones (only
> one location must be set to 23 or less)
> - Please note if you have any TEHO Route Patterns configured and IF
> you do - please temporarily set them to: Block the Call and play an
> distinguishable Annunciator message - this way if TEHO is the problem
> we will get Annunciator instead of reorder tone
> - Please note the External Number Mask specifically of the Phone in
> Site B (assuming Phone in Site A is calling Phone in Site B)
>
> - Also please note if you have restarted the CCM Service once enabling
> the AAR Service Parameter (required).
>
>
> Thanks,
>
> 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 Feb 25, 2008, at 8:37 PM, jason sung wrote:
>
> > Hello,
> >
> > i am having issue getting AAR to work (so simple but still...)
> >
> > when I call from a phone in site A to site B, the Phone is site A
> > displays network congestion but does not actually dial out instead
> > gives  a busy.
> >
> > I have groups configured and assigned it to phones,
> > I have proper locations configured,
> > I have proper router patterns configured,
> > I have external Mask configured,
> > I have AAR css assigned,
> >
> > but still no go...
> >
> > any ideas
>
>

Re: [OSL | CCIE_Voice] AAR issue

[jason sung]

Yes, I do.



On Mon, Feb 25, 2008 at 7:43 PM, boonchin .ng <[EMAIL PROTECTED]> wrote:

> What about the CallManager services parameter *Automated Alternate Routing
> Enable* *set to TRUE?
>
>
> On 2/26/08, jason sung <[EMAIL PROTECTED]> wrote:
> >
> > Hello,
> >
> > i am having issue getting AAR to work (so simple but still...)
> >
> > when I call from a phone in site A to site B, the Phone is site A
> > displays network congestion but does not actually dial out instead gives  a
> > busy.
> >
> > I have groups configured and assigned it to phones,
> > I have proper locations configured,
> > I have proper router patterns configured,
> > I have external Mask configured,
> > I have AAR css assigned,
> >
> > but still no go...
> >
> > any ideas
> >
>
>

[OSL | CCIE_Voice] AAR issue

[jason sung]

Hello,

i am having issue getting AAR to work (so simple but still...)

when I call from a phone in site A to site B, the Phone is site A displays
network congestion but does not actually dial out instead gives  a busy.

I have groups configured and assigned it to phones,
I have proper locations configured,
I have proper router patterns configured,
I have external Mask configured,
I have AAR css assigned,

but still no go...

any ideas

[OSL | CCIE_Voice] Locations & TEHO

[jason sung]

Can somebody tell me how do locations affect TEHO?

My call from HQ to BR1 did not work until I increased the bandwidth between
remote sites.

I am off the opinion to just leave the gateways out of Locations.

Any ideas???

[OSL | CCIE_Voice] Multicast moh

[jason sung]

I had a question, probably an easy one.

How do I verify Multicast moh playing from my flash?

Re: [OSL | CCIE_Voice] CME- B-ACD

[jason sung]

Thanks JD, that was exactly the issue and fix.

On Thu, Feb 21, 2008 at 4:04 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> That's a good question.  Your config seems correct.  Did you change your
> config after the script has been loaded?  You can try to stop and restart
> the script by doing "show call application session" to get the session ID
> number and then do a "call application session stop id xx" to stop the
> script and restart it by issuing "call application voice load queue"
> assuming queue is the name of the service.
>
> JD
>
>  --
> Date: Wed, 20 Feb 2008 22:34:44 -0600
> From: [EMAIL PROTECTED]
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] CME- B-ACD
>
>
> I thought that this command " param aa-hunt3 3210 " meant that it
> specifies option number 3 for hunt group 3210.
>
> Am I wrong?
>
> I have the BACD configured but no matter what I configure the mentioned
> command as, the queue still recognizes 1 for the huntgroup and says 3 is
> invalid.
>
> application
>  service queue flash:app-b-acd-2.1.0.0.tcl
>   param aa-hunt3 3210
>   param queue-len 10
>   param number-of-hunt-grps 1
>  !
>  service aa flash:app-b-acd-aa-2.1.0.0.tcl
>   paramspace english index 1
>   param number-of-hunt-grps 1
>   param dial-by-extension-option 4
>   param handoff-string aa
>   paramspace english language en
>   param max-time-vm-retry 2
>   param max-extension-length 4
>   param aa-pilot 3000
>   paramspace english location flash:
>   param second-greeting-time 60
>   param welcome-prompt _bacd_welcome.au
>   param call-retry-timer 15
>   param max-time-call-retry 600
>   param voice-mail 3600
>   param service-name queue
>  !
>
>
> DEBUG
>
> *Feb 20 14:51:52.955: //101//TCL :/tcl_PutsObjCmd:
> proc init_perCallvars
> *Feb 20 14:51:52.955:
> *Feb 20 14:51:52.963: //101//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing
> Welcome Prompt and options menu ++
> *Feb 20 14:51:56.064: //5//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> B-ACD
> Service Running <<<
> *Feb 20 14:51:58.187: //5//TCL :/tcl_PutsObjCmd: TCL B-ACD: ++ Message
> received from IOS ++
> *Feb 20 14:51:58.195: //101//TCL :/tcl_PutsObjCmd: TCL AA: +++ Invalid
> option selected +++
> *Feb 20 14:52:03.087: //5//TCL :/tcl_PutsObjCmd: TCL B-ACD: ++ Message
> received from IOS ++
>
>
> --
> Connect and share in new ways with Windows Live. Get it 
> now!
>

[OSL | CCIE_Voice] CME- B-ACD

[jason sung]

I thought that this command " param aa-hunt3 3210 " meant that it specifies
option number 3 for hunt group 3210.

Am I wrong?

I have the BACD configured but no matter what I configure the mentioned
command as, the queue still recognizes 1 for the huntgroup and says 3 is
invalid.

application
 service queue flash:app-b-acd-2.1.0.0.tcl
  param aa-hunt3 3210
  param queue-len 10
  param number-of-hunt-grps 1
 !
 service aa flash:app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 1
  param number-of-hunt-grps 1
  param dial-by-extension-option 4
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param max-extension-length 4
  param aa-pilot 3000
  paramspace english location flash:
  param second-greeting-time 60
  param welcome-prompt _bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 600
  param voice-mail 3600
  param service-name queue
 !


DEBUG

*Feb 20 14:51:52.955: //101//TCL :/tcl_PutsObjCmd:
proc init_perCallvars
*Feb 20 14:51:52.955:
*Feb 20 14:51:52.963: //101//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing
Welcome Prompt and options menu ++
*Feb 20 14:51:56.064: //5//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> B-ACD
Service Running <<<
*Feb 20 14:51:58.187: //5//TCL :/tcl_PutsObjCmd: TCL B-ACD: ++ Message
received from IOS ++
*Feb 20 14:51:58.195: //101//TCL :/tcl_PutsObjCmd: TCL AA: +++ Invalid
option selected +++
*Feb 20 14:52:03.087: //5//TCL :/tcl_PutsObjCmd: TCL B-ACD: ++ Message
received from IOS ++

Re: [OSL | CCIE_Voice] CUE FNA FB Problem

[jason sung]

Sounds like your dial-peers are not engaging.

Tried a reboot yet?

On Wed, Feb 20, 2008 at 3:37 PM, Jose Linero Welcker <
[EMAIL PROTECTED]> wrote:

> Hi Jason:
>
> Thanks for your answer, yes I have a Txcodec configured, and I tried
> configuring the cinming dial-peer with G711ulaw and still receive the SIP
> message:
>
> Feb 20 21:37:13.366: //-1//SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 302 Moved Temporarily
> Via: SIP/2.0/UDP  162.1.103.1:5060;branch=z9hG4bKEC3A
> From: "BR1-Phone2" ;tag=AABABC-24C1
> To: ;tag=38DDEC-B6F
> Date: Wed, 20 Feb 2008 21:37:03 GMT
> Call-ID: [EMAIL PROTECTED]
> Timestamp: 1203543423
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 101 INVITE
> Allow-Events: telephone-event
> Contact: 
> Diversion: ;reason=no-answer
> Content-Length: 0
>
> Regards,
>
> Jose
>
>  --
> Date: Wed, 20 Feb 2008 15:33:27 -0600
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Problem
> CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
>
>
> You might have a codec issue here.
>
> Do you have transcoder configured on the cme router.
>
> If you do, try to configure the sip incoming dial-peer with "codec
> g711ulaw"
>
> On Wed, Feb 20, 2008 at 3:29 PM, Jose Linero Welcker <
> [EMAIL PROTECTED]> wrote:
>
> Hi JD:
>
> Actually I have it:
>
> dial-peer voice 4 voip
>  session protocol sipv2
>  incoming called-number 3...
>  dtmf-relay rtp-nte
>
> Any ideas?
>
> Regards,
>
> Jose
>
>
>  --
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Date: Wed, 20 Feb 2008 13:22:47 -0800
> Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Problem
>
>
> You are missing an incoming dialpeer for your g729 voip connection on your
> cme router.  Either create a dedicated incoming dialpeer for your g729 voip
> connections or you can create a general dialpeer like the one below.
>
> dial-peer voice 100 voip
>  incoming called-number .
>
> JD
>
>  --
> From: [EMAIL PROTECTED]
> To: ccie_voice@onlinestudylist.com
> Date: Wed, 20 Feb 2008 21:17:11 +
> Subject: [OSL | CCIE_Voice] CUE FNA FB Problem
>
> Hi all:
>
> I have this configuration:
>
> CM --- H323 Trunk --- IPIPGW --- SIP Trunk --- CME
>
> The H323 is using G711 and the the SIP Trunk is using G729, the calls
> between the phones in CCM and CME are working without problems, however when
> I am going to forward the call to CUE due CFNA or CFB I received a reorder
> tone and I am seeing this SIP Message:
>
> Sent:
> SIP/2.0 302 Moved Temporarily
> Via: SIP/2.0/UDP  162.1.103.1:5060;branch=z9hG4bK92655
> From: "BR1-Phone2" ;tag=8EC914-101D
> To: ;tag=1CEC58-818
> Date: Wed, 20 Feb 2008 21:06:32 GMT
> Call-ID: [EMAIL PROTECTED]
> Timestamp: 1203541592
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 101 INVITE
> Allow-Events: telephone-event
> Contact: 
> Diversion: ;reason=no-answer
> Content-Length: 0
>
> and the IPIPGW answer is an ACK not a reinvite, in the CME I have
> configured this for CUE:
>
> telephony-service
>  voicemail 3600
>  call-forward pattern .T
>  web admin system name Admin password cisco
>
> dial-peer voice 6 voip
>  destination-pattern 3600
>  session protocol sipv2
>  session target ipv4:10.1.202.2
>  dtmf-relay sip-notify rtp-nte
>  codec g711ulaw
>  no vad
>
> voice service voip
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>
> ephone-dn  1  dual-line
>  number 3001
>  call-forward busy 3600
>  call-forward noan 3600 timeout 10
> !
>
>
> Does anybody see this problem, I am trying to reach the voicemail, but I
> can't.
>
> Regards,
>
> Jose
>
>
>
> --
> Express yourself instantly with MSN Messenger! MSN 
> Messenger
>
>
> --
> Need to know the score, the latest news, or you need your Hotmail(R)-get
> your "fix". Check it out. 
>
>
> --
> Express yourself instantly with MSN Messenger! MSN 
> Messenger
>
>
>
> --
> Express yourself instantly with MSN Messenger! MSN 
> Messenger
>

Re: [OSL | CCIE_Voice] CUE FNA FB Problem

[jason sung]

You might have a codec issue here.

Do you have transcoder configured on the cme router.

If you do, try to configure the sip incoming dial-peer with "codec g711ulaw"

On Wed, Feb 20, 2008 at 3:29 PM, Jose Linero Welcker <
[EMAIL PROTECTED]> wrote:

> Hi JD:
>
> Actually I have it:
>
> dial-peer voice 4 voip
>  session protocol sipv2
>  incoming called-number 3...
>  dtmf-relay rtp-nte
>
> Any ideas?
>
> Regards,
>
> Jose
>
>
>  --
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
> Date: Wed, 20 Feb 2008 13:22:47 -0800
> Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Problem
>
>
> You are missing an incoming dialpeer for your g729 voip connection on your
> cme router.  Either create a dedicated incoming dialpeer for your g729 voip
> connections or you can create a general dialpeer like the one below.
>
> dial-peer voice 100 voip
>  incoming called-number .
>
> JD
>
>  --
> From: [EMAIL PROTECTED]
> To: ccie_voice@onlinestudylist.com
> Date: Wed, 20 Feb 2008 21:17:11 +
> Subject: [OSL | CCIE_Voice] CUE FNA FB Problem
>
> Hi all:
>
> I have this configuration:
>
> CM --- H323 Trunk --- IPIPGW --- SIP Trunk --- CME
>
> The H323 is using G711 and the the SIP Trunk is using G729, the calls
> between the phones in CCM and CME are working without problems, however when
> I am going to forward the call to CUE due CFNA or CFB I received a reorder
> tone and I am seeing this SIP Message:
>
> Sent:
> SIP/2.0 302 Moved Temporarily
> Via: SIP/2.0/UDP  162.1.103.1:5060;branch=z9hG4bK92655
> From: "BR1-Phone2" ;tag=8EC914-101D
> To: ;tag=1CEC58-818
> Date: Wed, 20 Feb 2008 21:06:32 GMT
> Call-ID: [EMAIL PROTECTED]
> Timestamp: 1203541592
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 101 INVITE
> Allow-Events: telephone-event
> Contact: 
> Diversion: ;reason=no-answer
> Content-Length: 0
>
> and the IPIPGW answer is an ACK not a reinvite, in the CME I have
> configured this for CUE:
>
> telephony-service
>  voicemail 3600
>  call-forward pattern .T
>  web admin system name Admin password cisco
>
> dial-peer voice 6 voip
>  destination-pattern 3600
>  session protocol sipv2
>  session target ipv4:10.1.202.2
>  dtmf-relay sip-notify rtp-nte
>  codec g711ulaw
>  no vad
>
> voice service voip
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>
> ephone-dn  1  dual-line
>  number 3001
>  call-forward busy 3600
>  call-forward noan 3600 timeout 10
> !
>
>
> Does anybody see this problem, I am trying to reach the voicemail, but I
> can't.
>
> Regards,
>
> Jose
>
>
>
> --
> Express yourself instantly with MSN Messenger! MSN 
> Messenger
>
>
> --
> Need to know the score, the latest news, or you need your Hotmail(R)-get
> your "fix". Check it out. 
>
>
> --
> Express yourself instantly with MSN Messenger! MSN 
> Messenger
>

Re: [OSL | CCIE_Voice] called number in Unity's CallViewer mimicsForwardingnumber

[jason sung]

Brilliant.

On Feb 18, 2008 12:50 PM, Vik Malhi <[EMAIL PROTECTED]> wrote:

> I'll post that again with the correction:-) the translation pattern should
> be 166x in my earlier example
>
>
> (1) Ensure that the HQ site has some spare DID numbers- e.g. in PL we
> route
> 21222x1...to the HQ gateway.
>
> (2) Use the alias command in call-manager-fallback to route each extension
> to a unique DID number. E.g.
>
> Call-manager-fallback
>  voicemail 912122211600
>  alias 1 2001 to 2001 cfw 912122211661 timeout 12
>  alias 2 2002 to 2002 cfw 912122211662 timeout 12
>  alias 3 2003 to 2003 cfw 912122211663 timeout 12
>
> (3) On the CallManager create a Translation Pattern as shown below:
>
> DN = 166X / pt-internal
> CSS = css-internal
> Called # Mask = 200x
>
> (4) When the CCM receives the call it tries to ring 200X which is not
> registered. It will then use the call fwd no answer setting (which should
> be
> send to VM).
>
> (5) Add the Alternate Extension on Unity so that direct calls are routed
> to
> subscriber sign-in.
>
> Vik Malhi
> CCIE Voice Instructor / Developer - IPexpert, Inc.
> CCIE Voice #13890 CCSI #31584
> URL: http://www.IPexpert.com <http://www.ipexpert.com/>
> Toll Free: +1.866.225.8064
> International: +1.810.326.1444
>
> -Original Message-
> From: Vik Malhi [mailto:[EMAIL PROTECTED]
> Sent: Monday, February 18, 2008 10:49 AM
> To: 'Matthew Cody'; '[EMAIL PROTECTED]'; 'Juan Lopez Hernandez -X
> (jlopezhe - IBM - INS at Cisco)'; 'jason sung'
> Cc: 'ccie_voice@onlinestudylist.com'; 'Matthew Saskin'
>  Subject: RE: [OSL | CCIE_Voice] called number in Unity's CallViewer
> mimicsForwardingnumber
>
> I'll present a different strategy that works when a router is in SRST mode
> and you cannot rely on the RDNIS for whatever reason. It allows you to
> avoid
> dealing with vm-integration. Let me stress, in the field the workaround is
> to upgrade the IOS to a release with the RDNIS issue fixed and beg your
> telco to pass the RDNIS.
>
> (1) Ensure that the HQ site has some spare DID numbers- e.g. in PL we
> route
> 21222x1...to the HQ gateway.
>
> (2) Use the alias command in call-manager-fallback to route each extension
> to a unique DID number. E.g.
>
> Call-manager-fallback
>  voicemail 912122211600
>  alias 1 2001 to 2001 cfw 912122211661 timeout 12  alias 2 2002 to 2002
> cfw
> 912122211662 timeout 12  alias 3 2003 to 2003 cfw 912122211663 timeout 12
>
> (3) On the CallManager create a Translation Pattern as shown below:
>
> DN = 1661 / pt-internal
> CSS = css-internal
> Called # Mask = 200x
>
> (4) When the CCM receives the call it tries to ring 200X which is not
> registered. It will then use the call fwd no answer setting (which should
> be
> send to VM).
>
> (5) Add the Alternate Extension on Unity so that direct calls are routed
> to
> subscriber sign-in.
>
>
> Vik Malhi
> CCIE Voice Instructor / Developer - IPexpert, Inc.
> CCIE Voice #13890 CCSI #31584
> URL: http://www.IPexpert.com <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 Matthew Cody
> Sent: Monday, February 18, 2008 10:19 AM
> To: [EMAIL PROTECTED]; 'Juan Lopez Hernandez -X (jlopezhe - IBM - INS
> at
> Cisco)'; 'jason sung'
> Cc: ccie_voice@onlinestudylist.com; 'Matthew Saskin'
> Subject: Re: [OSL | CCIE_Voice] called number in Unity's CallViewer
> mimicsForwardingnumber
>
> There is also a somewhat exhaustive discussion of a similar topic at
> www.voiceie.com; look for a thread with 100 or so replies - that'll be the
> one.
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ovais Iqbal
> Sent: Monday, February 18, 2008 11:16 AM
> To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); jason sung
> Cc: ccie_voice@onlinestudylist.com; Matthew Saskin
> Subject: Re: [OSL | CCIE_Voice] called number in Unity's Call Viewer
> mimicsForwardingnumber
>
> If u really want to see the correct # then under the D-channel on your
> SRST
> gateway add the isdn ie-redirected command, please check the exact command
> I
> can not recall right now.
>
> Remeber this may break your vm-integration configuration.
>
> The above command will help in case of RDNIS configuration.
> Ovais Iqbal
> 416-294-7869
> Sent from my BlackBerry device
>
> -Original Message-
> From: "Juan Lopez Hernandez -X (jlopezhe - 

Re: [OSL | CCIE_Voice] called number in Unity's Call Viewer mimicsForwardingnumber

[jason sung]

I see...

I will go with VM-Integration for the lab and if I have time I might even
give RDNIS a shot. (Time: who am I kidding)...

Thanks Ovais, Matt.

On Feb 18, 2008 8:51 AM, Ovais Iqbal <[EMAIL PROTECTED]> wrote:

> Hey Jason,
>
> U really need vm-integration configuration for your real lab, its fine if
> you want to test both options in your home lab.
> Ovais Iqbal
> 416-294-7869
> Sent from my BlackBerry device
>
> -Original Message-
>  From: Matthew Saskin <[EMAIL PROTECTED]>
>
> Date: Mon, 18 Feb 2008 09:49:23
> To:jason sung <[EMAIL PROTECTED]>
> Cc:[EMAIL PROTECTED], ccie_voice@onlinestudylist.com, "Juan Lopez
> Hernandez -X (jlopezhe - IBM - INS at Cisco)" <[EMAIL PROTECTED]>
> Subject: Re: [OSL | CCIE_Voice] called number in Unity's Call Viewer
>  mimicsForwarding
>  number
>
>
> There's no guarantee that RDNIS will actually work however - I've come
> across plenty of telco's where RDNIS just gets lost in the network...
>
> -matt
>
> jason sung wrote:
> > Juan,
> >
> > You are correct about VM-Integration needed only for analog or CAS
> > circuits, because in this case DTMF tones are used to route call to
> > the appropriate mailbox.  For PRIs all you have do is configure
> > complete 10 digit number as voicemail under fallback and check "RDNIS
> > outgoing" on central site PRI.
> >
> > Mark or Vik: Can you please confirm this step?
> >
> > Neverthless Ovais, thanks for the attahed configuration note.
> > Appreciate it.
> >
> > Thanks
> >
> > On Feb 17, 2008 8:56 AM, Ovais Iqbal <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>> wrote:
> >
> > Vm-integration also works for PRI ports but u need to configure it
> > properly. I am not infront of my computer right now, I can send
> > you a config later if u want.
> >
> >
> > Ovais Iqbal
> > 416-294-7869
> > Sent from my BlackBerry device
> >
> > -Original Message-
> > From: "Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)"
> > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>
> >
> > Date: Sun, 17 Feb 2008 10:59:25
> > To:"ovais Iqbal" <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>>, "jason sung" <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>>
> > Cc: > <mailto:ccie_voice@onlinestudylist.com>>
> > Subject: RE: [OSL | CCIE_Voice] called number in Unity's Call
> > Viewer mimicsForwarding number
> >
> >
> > Hi Ovais and Jason,
> >
> > thanks for the reply on this. On CCM I already checked
> > the'"Redirecting Number IE Delivery - Inbound'' checkbox on the
> > (incoming) HQ gateway. You can also see the Called Party Number
> > and the RedirectingNumber that comes into CCM from the trace below
> > - but Unity's Call Viewer doesn't show the correct ''dialed
> > number'' though...
> >
> > Ovais, the VM-Integration, is that supported for PRI calls? I
> > thought it only worked for analog - or T1 CAS/E1 r2 - line's - but
> > I can be wrong of course?
> >
> > Kind regards,
> > Juan
> >
> >
> > 
> >  From: [EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>
> > [mailto:[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>] On Behalf Of
> > ovais Iqbal
> > Sent: Saturday, February 16, 2008 4:47 AM
> > To: jason sung
> > Cc: ccie_voice@onlinestudylist.com
> > <mailto:ccie_voice@onlinestudylist.com>; Juan Lopez Hernandez -X
> > (jlopezhe - IBM - INS at Cisco)
> > Subject: Re: [OSL | CCIE_Voice] called number in Unity's Call
> > Viewer mimicsForwarding number
> >
> >
> > by the way my understanding is that in the real LAB the RDNIS does
> > not work today - you would need VM-Integration option to be
> > configured to get it work properly in your real lab. I know at
> > home I made it to work with redirected number option.
> >
> >
> > On 2/15/08, jason sung <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]> <mailto:[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>> > wrote:
> > Juan,
> >
> > You may have already done this, Is the RDNIS box checked on the HQ
> > PRI?
> >

Re: [OSL | CCIE_Voice] called number in Unity's Call Viewer mimicsForwarding number

[jason sung]

Juan,

You are correct about VM-Integration needed only for analog or CAS circuits,
because in this case DTMF tones are used to route call to the appropriate
mailbox.  For PRIs all you have do is configure complete 10 digit number as
voicemail under fallback and check "RDNIS outgoing" on central site PRI.

Mark or Vik: Can you please confirm this step?

Neverthless Ovais, thanks for the attahed configuration note. Appreciate it.

Thanks

On Feb 17, 2008 8:56 AM, Ovais Iqbal <[EMAIL PROTECTED]> wrote:

> Vm-integration also works for PRI ports but u need to configure it
> properly. I am not infront of my computer right now, I can send you a config
> later if u want.
>
>
> Ovais Iqbal
> 416-294-7869
> Sent from my BlackBerry device
>
> -Original Message-
> From: "Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)" <
> [EMAIL PROTECTED]>
>
> Date: Sun, 17 Feb 2008 10:59:25
> To:"ovais Iqbal" <[EMAIL PROTECTED]>, "jason sung" <[EMAIL PROTECTED]
> >
> Cc:
> Subject: RE: [OSL | CCIE_Voice] called number in Unity's Call Viewer
> mimicsForwarding number
>
>
> Hi Ovais and Jason,
>
> thanks for the reply on this. On CCM I already checked the'"Redirecting
> Number IE Delivery - Inbound'' checkbox on the (incoming) HQ gateway. You
> can also see the Called Party Number and the RedirectingNumber that comes
> into CCM from the trace below - but Unity's Call Viewer doesn't show the
> correct ''dialed number'' though...
>
> Ovais, the VM-Integration, is that supported for PRI calls? I thought it
> only worked for analog - or T1 CAS/E1 r2 - line's - but I can be wrong of
> course?
>
> Kind regards,
> Juan
>
>
> 
>  From: [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] On Behalf Of ovais Iqbal
> Sent: Saturday, February 16, 2008 4:47 AM
> To: jason sung
> Cc: ccie_voice@onlinestudylist.com; Juan Lopez Hernandez -X (jlopezhe -
> IBM - INS at Cisco)
> Subject: Re: [OSL | CCIE_Voice] called number in Unity's Call Viewer
> mimicsForwarding number
>
>
> by the way my understanding is that in the real LAB the RDNIS does not
> work today - you would need VM-Integration option to be configured to get it
> work properly in your real lab. I know at home I made it to work with
> redirected number option.
>
>
> On 2/15/08, jason sung <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> >
> wrote:
> Juan,
>
> You may have already done this, Is the RDNIS box checked on the HQ PRI?
>
>
>
>  On Feb 15, 2008 2:28 PM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at
> Cisco) <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > wrote:
>
>
> Hello,
>
> I need some help as I can't figure out the following: I try to test Unity
> with BR1 in SRST. All is going fine, the call to unity is set up over PSTN,
> redirecting number works fine. From the CCM trace files I see the call
> coming in as:
>
> SETUP  pd = 8  callref = 0x00AB
> Bearer Capability i = 0x8090A3
> Channel ID i = 0xA98381
> Calling Party Number i = '!', 0x80, '6175252001'
> Called Party Number i = 0x80, '2122251600'<--- 4 last digits are
> the Unity pilot number - this incoming number gets stripped to ''1600''
> Redirecting Number i = 0x82, '6175252002'
>
> When looking into Unity's Call Viewer, I notice that the ''dialed number''
> always follows the ''forwarding number''. So instead of seeing the number
> 1600 as dialed number in the Call Viewer , I see whatever is the forwarding
> number. Is there a setting that can be changed so you see the correct
> ''dialed number''?
>
> Kind regards,
> Juan
>
>
>
>
>
>
>
>
> --
> Ovais Iqbal
> 416-294-7869
>

[OSL | CCIE_Voice] CBWFQ on sub ints

[jason sung]

Is CBWFQ supported on sub interfaces?

Does the IOS even let you configure an LLQ service-policy on a sub
interface?

Re: [OSL | CCIE_Voice] called number in Unity's Call Viewer mimics Forwarding number

[jason sung]

Juan,

You may have already done this, Is the RDNIS box checked on the HQ PRI?

On Feb 15, 2008 2:28 PM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at
Cisco) <[EMAIL PROTECTED]> wrote:

>  Hello,
>
> I need some help as I can't figure out the following: I try to test Unity
> with BR1 in SRST. All is going fine, the call to unity is set up over PSTN,
> redirecting number works fine. From the CCM trace files I see the call
> coming in as:
>
> SETUP  pd = 8  callref = 0x00AB
> Bearer Capability i = 0x8090A3
> Channel ID i = 0xA98381
> Calling Party Number i = '!', 0x80, '6175252001'
> Called Party Number i = 0x80, '2122251600'<--- 4 last digits are
> the Unity pilot number - this incoming number gets stripped to ''1600''
> Redirecting Number i = 0x82, '6175252002'
>
> When looking into Unity's Call Viewer, I notice that the ''dialed number''
> always follows the ''forwarding number''. So instead of seeing the number
> 1600 as dialed number in the Call Viewer , I see whatever is the forwarding
> number. Is there a setting that can be changed so you see the correct
> ''dialed number''?
>
> Kind regards,
> Juan
>
>
>

Re: [OSL | CCIE_Voice] Attendant Console

[jason sung]

Ok, here is the fix incase anybody is curious.

Solution: JTAPI password reset.

On Fri, Feb 15, 2008 at 3:13 PM, jason sung <[EMAIL PROTECTED]> wrote:

> Hello All,
>
> I am having issues making Attendant Console hunt group work.
> I have done the following so far.
> Created a Pilot point named ACPILOT, with a DN 1550, pt-internal and
> CSS-hq-all
> Created a hunt group and added a DN >> this is where I get an error
> message saying " Hunt group memeber 1 does not have a valid DN"
> but it still takes the member and I can successfully update.
> Created an ac user and associated pilot point and the phone number.
> Finally I reset TCD service.
>
> When I call 1550, I get busy. I have verified CSS and partitions.
>
> any ideas???
>
> I have made this work before, but this time it is just not working.
>
> -Jason
>

[OSL | CCIE_Voice] Attendant Console

[jason sung]

Hello All,

I am having issues making Attendant Console hunt group work.
I have done the following so far.
Created a Pilot point named ACPILOT, with a DN 1550, pt-internal and
CSS-hq-all
Created a hunt group and added a DN >> this is where I get an error message
saying " Hunt group memeber 1 does not have a valid DN"
but it still takes the member and I can successfully update.
Created an ac user and associated pilot point and the phone number.
Finally I reset TCD service.

When I call 1550, I get busy. I have verified CSS and partitions.

any ideas???

I have made this work before, but this time it is just not working.

-Jason

Re: [OSL | CCIE_Voice] Re;Re: QoS question

[jason sung]

correct.

On Thu, Feb 14, 2008 at 3:43 PM, DSCP46EF <[EMAIL PROTECTED]> wrote:

> Thanks Jason,
>
> To over come from this 75% limitation, we should use "max-reserved
> bandwidth" - am I right?
>
> Cheers
> Frog
>
>
> On Fri, Feb 15, 2008 at 4:27 AM, jason sung <[EMAIL PROTECTED]> wrote:
>
> > Frog,
> >
> > ValueB> this is not the usable bandwidth, You are sitll allowed to use
> > all 364800.
> > The (75%) is the total amount you can configure under your policy maps.
> > For example: CIR-364800
> > class rtp
> > priority percent 50
> > class sig
> > bandwidth percent 10
> > class any
> > bandwidth percent 15
> > class class-default
> > (THIS CLASS NEEDS SOME BANDWIDTH WHICH IT CAN USE FROM THE LEFT OVER 25%
> > OF 364800)
> >
> > 50+10+15= 75%
> >
> > On Thu, Feb 14, 2008 at 9:57 AM, DSCP46EF <[EMAIL PROTECTED]>
> > wrote:
> >
> > > Hi,
> > >
> > > Question: Lets say today bandwidth is 384 and we can only use 75% of
> > > configured bandwidth by default.
> > >
> > > valueA = Cir = 95% of Bandwidth = 364kbps
> > > ValueB = Usable bandwidth = 75% of total bandwidth = 291 kbps (if it
> > > is less than what is required for qos then max-reserve bandwidth kicks in)
> > >
> > > MY QUETIONS:
> > > --
> > > Q1: Bandwidth percent 5 - is that calculated from CIR=384kbps or from
> > > Total useble bw=291?
> > >
> > >
> > >
> > > As the original question by BOONng says:
> > >
> > > In the policy-map LLQ, I have "bandwidth percent 5". So can I
> > > > assume that is 5% of the CIR value? If I want to use "priority x"
> > > and
> > > > "bandwidth y" without using the *percent command*, how I can
> > > calculate the
> > > > bandwidth value for class media and class control?
> > >
> > >
> > >
> > > Frog
> > >
> > >
> > >
> >
>
>
> --
> Smile, you'll save someone else's day!
> Frog

Re: [OSL | CCIE_Voice] QoS question

[jason sung]

G729 on frame-relay with FR.12 is not 24K it is 28K

so your 5 g729 calls will be

5 x 28 = 140






On Thu, Feb 14, 2008 at 11:36 AM, boonchin .ng <[EMAIL PROTECTED]>
wrote:

> Hi Jason,
> Thanks for the explanation.
>
> So, I would like to understand another scenario.
>
> Similarly, I have a 384K frame-relay WAN link, so I have:
>  !
> map-class frame-relay FRTS-384
>  frame-relay cir 364800
>  frame-relay bc 3648
>  frame-ralay be 0
>  frame-relay mincir 364800
>  frame-relay fragment 480
> !
>
> Now, for LLQ I want to configure to only prioritized 5 g729 calls:
>  !
> policy-map LLQ
>  class media
>   priority x
>  class control
>   bandwidth 16
>   class class-default
>   fair-queue
> !
>
> So, the x value is 5 g729 calls. How can I calculate the x value? Based on
> CCM bandwidth for a g729? x = 5 x 24
>
> Please kindly advice.
>
> Thanks.
>
>
> -- Forwarded message --
> From: jason sung <[EMAIL PROTECTED]>
> Date: Thu, Feb 14, 2008 at 11:26 PM
> Subject: Re: [OSL | CCIE_Voice] QoS question
> To: "boonchin .ng" <[EMAIL PROTECTED]>
> Cc: CCIE Maillist 
>
>
> When you have minCIR configured the LLQ picks 33% of that.
>
> If you do not have minCIR configured than the CIR divided by two is used
> in the calculation.
>
>
>
> In your example, since you have mincir configured.
> Priority bandwidth will be 364800 * .33 = about 120K
> LLQ will allow you to configure 75% of the total bandwidth.
> if you want to alter this, use the command
> max-reserved bandwidth xxx
>
> -Jason
>
>   On Thu, Feb 14, 2008 at 5:45 AM, boonchin .ng <[EMAIL PROTECTED]>
> wrote:
>
> > Hi,
> >
> > I have a 384K frame-relay link. Hence, I have:
> >
> > !
> > map-class frame-relay FRTS-384
> >  frame-relay cir 364800
> >  frame-relay bc 3648
> >  frame-ralay be 0
> >  frame-relay mincir 364800
> >  frame-relay fragment 480
> > !
> >
> > Now, I want to do LLQ on voice, so:
> >
> > !
> > policy-map LLQ
> >  class media
> >   priority percent 33
> >  class control
> >   bandwidth percent 5
> >  class class-default
> >   fair-queue
> > !
> >
> > *Question*: In the policy-map LLQ, I have "bandwidth percent 5". So can
> > I assume that is 5% of the CIR value? If I want to use "priority x" and
> > "bandwidth y" without using the *percent command*, how I can calculate
> > the bandwidth value for class media and class control?
> >
> > Thanks.
> >
>
>
>

Re: [OSL | CCIE_Voice] Re;Re: QoS question

[jason sung]

Frog,

ValueB> this is not the usable bandwidth, You are sitll allowed to use all
364800.
The (75%) is the total amount you can configure under your policy maps.
For example: CIR-364800
class rtp
priority percent 50
class sig
bandwidth percent 10
class any
bandwidth percent 15
class class-default
(THIS CLASS NEEDS SOME BANDWIDTH WHICH IT CAN USE FROM THE LEFT OVER 25% OF
364800)

50+10+15= 75%

On Thu, Feb 14, 2008 at 9:57 AM, DSCP46EF <[EMAIL PROTECTED]> wrote:

> Hi,
>
> Question: Lets say today bandwidth is 384 and we can only use 75% of
> configured bandwidth by default.
>
> valueA = Cir = 95% of Bandwidth = 364kbps
> ValueB = Usable bandwidth = 75% of total bandwidth = 291 kbps (if it is
> less than what is required for qos then max-reserve bandwidth kicks in)
>
> MY QUETIONS:
> --
> Q1: Bandwidth percent 5 - is that calculated from CIR=384kbps or from
> Total useble bw=291?
>
>
>
> As the original question by BOONng says:
>
> In the policy-map LLQ, I have "bandwidth percent 5". So can I
> > assume that is 5% of the CIR value? If I want to use "priority x" and
> > "bandwidth y" without using the *percent command*, how I can calculate
> the
> > bandwidth value for class media and class control?
>
>
>
> Frog
>
>
>

Re: [OSL | CCIE_Voice] QoS question

[jason sung]

When you have minCIR configured the LLQ picks 33% of that.

If you do not have minCIR configured than the CIR divided by two is used in
the calculation.



In your example, since you have mincir configured.
Priority bandwidth will be 364800 * .33 = about 120K
LLQ will allow you to configure 75% of the total bandwidth.
if you want to alter this, use the command
max-reserved bandwidth xxx

-Jason

On Thu, Feb 14, 2008 at 5:45 AM, boonchin .ng <[EMAIL PROTECTED]> wrote:

> Hi,
>
> I have a 384K frame-relay link. Hence, I have:
>
> !
> map-class frame-relay FRTS-384
>  frame-relay cir 364800
>  frame-relay bc 3648
>  frame-ralay be 0
>  frame-relay mincir 364800
>  frame-relay fragment 480
> !
>
> Now, I want to do LLQ on voice, so:
>
> !
> policy-map LLQ
>  class media
>   priority percent 33
>  class control
>   bandwidth percent 5
>  class class-default
>   fair-queue
> !
>
> *Question*: In the policy-map LLQ, I have "bandwidth percent 5". So can I
> assume that is 5% of the CIR value? If I want to use "priority x" and
> "bandwidth y" without using the *percent command*, how I can calculate the
> bandwidth value for class media and class control?
>
> Thanks.
>

Re: [OSL | CCIE_Voice] QoS Design for Specific WAN Link

[jason sung]

I was reading ceritification talk forum.

http://www.certificationtalk.com:81/showflat.php?Cat//Board/voice4twelve/Number/29002/page/0/view/collapsed/sb/5/o//fpart/1

Is it true that we can use MLP PPP over frame-relay?

JD, if this is true then I answered Question#1 incorrectly? but I doubt that
MLP PPP can be used over frame-relay.

On Feb 13, 2008 8:02 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> Hi,
>
> I just went over the WAN Edge Link-Specific QoS Design section of the QoS
> SRND document and i read that Cisco recommends the following:
>
> 1) For leased line < 768Kbps, use MLP LFI and cRTP.  Essentially,
> attaching the  MQC policy for LLQ to a Multilink interface.
> 2) For frame relay < 768Kbps, use FRF.12 and cRTP.  Essentially, attaching
> the MQC policy for LLQ to a frame relay map class.
> 3) For ATM < 768Kbps, use MLP LFI and cRTP.  Essentially, attaching the
> MQC policy for LLQ to a Virtual-Interface interface.
>
> So my questions are:
>
> 1. What's the difference between each method?
> 2. What's the difference between multilink interface and
> virtual-interface?
> 3. Since the WAN for the voice lab is based on frame relay, do we even
> need to know multilink and virtual interfaces?
>
> Thanks for any input?
>
> JD
>
> --
> Connect and share in new ways with Windows Live. Get it 
> now!
>

Re: [OSL | CCIE_Voice] QoS Design for Specific WAN Link

[jason sung]

1. This will be for leased PPP lines not Frame-relay, so if the POD has
leased lines and not Frame-realy this can come into the picture.
2. Multilink interface is for leased lines and Virutal templates are for
frame-relay, they both are used to achieve fragmentation without using
FRF.12
3. I would stress more on Virtual-template but not ignore MLP PPP as who
knows what the POD has.





On Feb 13, 2008 8:02 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> Hi,
>
> I just went over the WAN Edge Link-Specific QoS Design section of the QoS
> SRND document and i read that Cisco recommends the following:
>
> 1) For leased line < 768Kbps, use MLP LFI and cRTP.  Essentially,
> attaching the  MQC policy for LLQ to a Multilink interface.
> 2) For frame relay < 768Kbps, use FRF.12 and cRTP.  Essentially, attaching
> the MQC policy for LLQ to a frame relay map class.
> 3) For ATM < 768Kbps, use MLP LFI and cRTP.  Essentially, attaching the
> MQC policy for LLQ to a Virtual-Interface interface.
>
> So my questions are:
>
> 1. What's the difference between each method?
> 2. What's the difference between multilink interface and
> virtual-interface?
> 3. Since the WAN for the voice lab is based on frame relay, do we even
> need to know multilink and virtual interfaces?
>
> Thanks for any input?
>
> JD
>
> --
> Connect and share in new ways with Windows Live. Get it 
> now!
>

Re: [OSL | CCIE_Voice] Can't get Virtual-Template to work

[jason sung]

I think you have your classes wrong. You should nest your VATS under
frame-relay.

Is VATS configured on the other end as well? You should see a virtual-access
interface come up.

One other thing is before you apply your Virtual template, please remove ip
and DLCI under your sub-interfaces.

-jason

On Feb 13, 2008 3:46 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> Can someone please let me know what I am missing from my configuration
> below for the Virtual-Template configuration?  I can't seem to get the
> interface working.  It's in the down state.  Any help is greatly
> appreciated.  Thanks.
>
> Here is my configuration
> ---
> class-map match-any SIGNAL
>  match ip dscp cs3  af31
> class-map match-all VOICE
>  match ip dscp ef
>
> policy-map LLQ
>  class VOICE
>   priority percent 33
>compress header ip rtp
>  class SIGNAL
>   bandwidth percent 5
>  class class-default
>   fair-queue
>   random-detect
>
> policy-map CB-VATS
>  class class-default
>   shape average 729600 3648 0
>   shape adaptive 364800
>   shape fr-voice-adapt deactivation 30
>   service-policy LLQ
>
> interface Serial0/1/0:0
>  no ip address
>  encapsulation frame-relay IETF
>  frame-relay fragmentation voice-adaptive deactivation 30
>  frame-relay lmi-type ansi
>
> interface Serial0/1/0:0.1 point-to-point
>  ip address 162.1.101.1 255.255.255.0
>  ip ospf mtu-ignore
>  frame-relay interface-dlci 201
>   class WAN-EDGE
>
> interface Serial0/1/0:0.2 point-to-point
>  ip ospf mtu-ignore
>  frame-relay interface-dlci 202 ppp Virtual-Template1
>
> interface Virtual-Template1
>  bandwidth 768
>  ip address 162.1.102.1 255.255.255.0
>  ip ospf mtu-ignore
>  ppp multilink
>  ppp multilink fragment delay 10
>  ppp multilink interleave
>  service-policy output CB-VATS
>
> map-class frame-relay WAN-EDGE
>  frame-relay fragment 480
>  service-policy output CB-VATS
>
>
>
>
>
> --
> Helping your favorite cause is as easy as instant messaging. You IM, we
> give. Learn 
> more.
>

[OSL | CCIE_Voice] Multicast MOH

[jason sung]

I am little confused on Multicast MOH hops. Can somebody correct me if I am
wrong about the following.

1. MOH server is in the same vlan as voice vlan
In this case if I were to stream MMOH to Branch 1: My hop count will be 3.
1st hop is local voice vlan, 2nd hop is the WAN link and 3rd hop is the
branch 1 phones.



2. MOH server is in a seperate server vlan.
In this case if I were to stream MMOH to Branch 1: My hop count will still
be 3.
1st hop is server vlan, 2nd hop is my WAN link and 3rd hop is the branch 1
phones.

Do I need to place "ip pim-dense mode" command on the serial interfaces as
well?

Jason.

Re: [OSL | CCIE_Voice] CME TCL B-ACD

[jason sung]

Try reloading the prompt.
 
audio-prompt load flash:en_bacd_welcome.au


Date: Tue, 12 Feb 2008 18:06:04 -0800From: [EMAIL PROTECTED]: [EMAIL 
PROTECTED]: [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] CME TCL B-ACD
Alright I ahve it connecting, buts its not playing any audio or respdonding to 
key presses.  Anything come to the top of anyone's head?
 
 
Thanks
 
Chad 
On 2/12/08, Patel, Mrugesh <[EMAIL PROTECTED]> wrote: 



I always think far. J
 

From: Chad Stachowicz [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 12, 
2008 7:15 PMTo: Patel, MrugeshCc: Allam Hassan; CCIE Maillist 
Subject: Re: [OSL | CCIE_Voice] CME TCL B-ACD



 

patel,

   just have 2 incoming dial peers ;)

 

Chad 

On 2/12/08, Patel, Mrugesh <[EMAIL PROTECTED]> wrote: 


So, we have to hair-pin the call back in from PSTN or IPIPGW. No?
 


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allam 
HassanSent: Tuesday, February 12, 2008 6:18 PMTo: Chad StachowiczCc: CCIE 
MaillistSubject: Re: [OSL | CCIE_Voice] CME TCL B-ACD

 
Service Applications cannot be triggered on Outbound Dial Peers ... only 
Inbound Dial Peers.

 

 


On Feb 12, 2008, at 6:59 PM, Chad Stachowicz wrote:
 

below is the relevant config, I don't understand why no outgoing dial-peer is 
being matched?  Any help?

 

Thanks

 

!application service queue flash:app-b-acd-2.1.0.0.tcl  param queue-len 20  
param number-of-hunt-grps 1  param aa-hunt2 3210 ! service aa 
flash:app-b-acd-aa-2.1.0.0.tcl  paramspace english index 1  param 
number-of-hunt-grps 1  param dial-by-extension-option 4  param handoff-string 
aa  paramspace english language en  param max-time-vm-retry 2  param 
max-extension-length 4  param aa-pilot 3200  paramspace english location flash: 
 param second-greeting-time 30  param welcome-prompt en_bacd_welcome.au  param 
call-retry-timer 15  param max-time-call-retry 600  param voice-mail 3600  
param service-name queue controller E1 0/0/0!interface Loopback0 ip 
address 172.4.102.1 255.255.255.255 ip ospf network point-to-point h323-gateway 
voip interface h323-gateway voip bind srcaddr 172.4.102.1!interface 
FastEthernet0/0 no ip address duplex auto speed auto!interface 
FastEthernet0/0.140 encapsulation dot1Q 140 native no snmp trap 
link-status!interface FastEthernet0/0.240 encapsulation dot1Q 240 ip address 
10.4.202.1 255.255.255.0 no snmp trap link-status!interface Service-Engine0/0 
no ip address shutdown!interface FastEthernet0/1 no ip address shutdown duplex 
auto speed auto!interface Serial0/1/0 no ip address encapsulation frame-relay 
IETF no fair-queue frame-relay lmi-type ansi!interface Serial0/1/0.1 
point-to-point ip address 162.4.102.2 255.255.255.0 frame-relay interface-dlci 
102!router ospf 1 log-adjacency-changes network 10.4.102.0 0.0.0.255 area 0 
network 10.4.202.0 0.0.0.255 area 0 network 162.4.102.0 0.0.0.255 area 0 
network 172.4.102.0 0.0.0.255 area 0!ip classless!!ip http serverno ip http 
secure-servertftp-server flash:P00307010100.bintftp-server 
flash:P00303020214.bintftp-server flash:P00403020214.bintftp-server 
flash:P00305000600.sbntftp-server flash:P00307020200.bintftp-server 
flash:P00307020200.loadstftp-server flash:P00307020200.sb2tftp-server 
flash:P00307020200.sbn!control-plane!!!dial-peer voice 10 voip service aa 
destination-pattern 3200!dial-peer voice 20 voip session target ipv4:10.4.202.1 
incoming called-number .!dial-peer voice 35 voip service aa destination-pattern 
3313243200gatekeeper shutdown!!telephony-service load 7910 P00403020214 
load 7960-7940 P00307020200 max-ephones 15 max-dn 20 ip source-address 
10.4.202.1 port 2000 create cnf-files version-stamp Jan 01 2002 00:00:00 
dialplan-pattern 1 3313243... extension-length 4 max-conferences 8 gain -6 
call-forward pattern .T moh moh_file.wav transfer-system full-consult 
transfer-pattern .T!!ephone-dn  1  dual-line number 3001 caller-id 
block!!ephone-dn  2  dual-line number 3002!!ephone-dn  9 number 3999 mwi 
on!!ephone-dn  10 number 3998 mwi off!!ephone  1 mac-address 
0030.94C4.22D6!!!ephone  2 mac-address 0011.BBE1.ADDF!!ephone-hunt 1 peer pilot 
3210 list 3001, 3002 timeout 12 statistics collect!!!line con 0line aux 0line 
194 no activation-character no exec transport preferred none transport input 
all transport output allline vty 0 4 privilege level 15 transport input 
telnetline vty 5 15 privilege level 15 transport input 
telnet!warm-rebootscheduler allocate 2 1000!end

P4-BR2-RTR(config-dial-peer)#*Feb 12 23:55:57.339: 
//-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore:   Calling Number=1000, 
Called Number=3200, Voice-Interface=0x0,   Timeout=TRUE, Peer Encap 
Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,   Peer Info 
Type=DIALPEER_INFO_SPEECH*Feb 12 23:55:57.339: 
//-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore:   Result=Success(0) after 
DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=20*Feb 12 23:55:57.339: 
//-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore:   Calling Number=1000, 
Calle