Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMware

[Alex Mendoza]

Hi AllI'm running my lab in W8.1 64bits, vmware workstation 10, core i7, 16GB RAM, 480GB SSD for vmware and 100GB SSD for OS.And works like a champ. It's been some dificult to tune up but is worth. If you go with vmware workstation feel free to ask if you found problems in your way.best regards.On Dec 10, 2013, at 08:25 PM, "Kenneth Staples (kstaples)"  wrote:Thanks for sharing that valuable info Josh. Me and my peers were speaking about that exact thing (having the routers and switches. I'll make sure and use the ESXi over the desktop version. Thanks again.  Kenneth Staples (Shift 7am – 4pm CST) Customer Support Engineer Cisco RMS - Unified Communications kstap...@cisco.comPhone: +1 512 340 3143 CCNOC = 866-777-6269 GVNOC = 866-643-9428 TelePresence = (888) 654-9113 Cisco Systems, Inc. 9500 Amberglen Blvd. United States Cisco.com - http://www.cisco.com  This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message.  For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.htmlFrom: Josh Petro  Date: Tuesday, December 10, 2013 8:21 PM To: Kenneth Staples  Cc: "ccie_voice@onlinestudylist.com"  Subject: Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMwareIf you're serious about studying, I would avoid the desktop version and go with ESXi and a good, reliable machine. You'll be fighting fatigue, the labs and learning new stuff, so adding troubleshooting a VMware Desktop issue will only frustrate you unnecessarily.I have 7961's and they were fine. I would recommend getting two 9900 series phones if you can so you can simulate video. Trust me, having the phones, gateways and switch will help a ton. Also, just find a 3750 switch and forget the router switch-modules. It's much easier and you'll still learn the needed commands. Just make sure you pay attention to the commands for those switch-modules (for VLAN setup, etc.) when studying. Anyone else, please feel free to comment. Josh On Tue, Dec 10, 2013 at 2:13 PM, Kenneth Staples (kstaples)  wrote:Thanks josh. I've been hearing that the VMWare desktop should be all that's needed with the exception of some physical routers/switches for the branch sites and phones of course. The the Lab requirements for the new Collab IE, what more equipment may be needed….or will your current set up be sufficient? Might need additional phones of course for video. Thanks  Kenneth Staples (Shift 7am – 4pm CST) Customer Support Engineer Cisco RMS - Unified Communications kstap...@cisco.comPhone: +1 512 340 3143 CCNOC =  866-777-6269 GVNOC = 866-643-9428 TelePresence =  (888) 654-9113 Cisco Systems, Inc. 9500 Amberglen Blvd. United States Cisco.com - http://www.cisco.com  This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message.  For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.htmlFrom: Josh Petro  Date: Tuesday, December 10, 2013 12:37 PM To: Kenneth Staples  Cc: "ccie_voice@onlinestudylist.com"  Subject: Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMwareKennethI run my entire lab on ESXi 5.1 in demo mode. Saves having to purchase since this is lab anyhow. Just shutdown the VMs and the host when you're don't for the day and you'll get some time out of it. I have a dell workstation with a Xeon processor with 6 cores and 24gb RAM. The more important equipment is the routers. I did two attempts on the voice lab and having the routers makes a huge difference for home studying. For collaboration, you can get by with most of the 2800/3800 features until a 2900/3900 is available on the cheap.Josh  On Tuesday, December 10, 2013, Kenneth Staples (kstaples) wrote:I wanted to ask if there is anyone out there preparing for the Voice CCIE lab using a VM server for the all UC servers involved and maybe even for the routers/switches as well? I could understand if your running all the UC servers on one VM server and the routers/switches that are needed are seperate from the VM. The reason I ask is because I'd like to see if it's possible to study convenient from my home (without having run 4 to 5 servers) rather than coming into the office to study for the lab. Ideally, I'd like to have one VM server running all UC nodes, and maybe a couple of routers/switches (non VM if necessary) and 

Re: [OSL | CCIE_Voice] CME as SRST help.

[Alex Mendoza]

HiFor telephony-service I used:srst mode auto-provision allsrst dn line-mode dualAnd for call-manager-fallback I used:max-ephone 4max-dn 8 dual-lineAm I wrong using these configurations?I avoid octo-lines for the bug mentioned.best regards!AlexOn Oct 17, 2013, at 10:18 AM, Ramcharan Arya  wrote:Hi Bill,I believe this might be related to a bug with using octo lines in CME SRST.Come out of SRST and reload the router this might resolve the issue.Regards,Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher  wrote:It’s not working!!  Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register.  They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com  Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com  Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___
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Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP

[Alex Mendoza]

As Dave says, you can book at Tokyo or other location.

I'm from Mexico and can book at RTP in february just one week ago.

More pressure because will be my 2nd and last attempt.

If you are so close to get your CCIE, look for a seat at other location even if 
you must pay for travel expenses.

All my best for the last candidates.

best regards
Alex
On Sep 27, 2013, at 10:57 AM, Martin Sloan  wrote:

> I'm really disappointed as well.  I just failed my second attempt on Wed and 
> was worried about getting a 4th try in when I logged on to see no seats left 
> for a 3rd!  I figured it would get tight but this is nuts.  I made a big 
> improvement on my score from the first try and feel like the third time could 
> have been the charm.  Oh well.
> 
> 
> On Fri, Sep 27, 2013 at 11:35 AM, Dane Warner  
> wrote:
> There are no open dates in either San Jose or RTP anymore, period.
> 
>  
> 
> Looks like if we want to take the Voice exam, which I’m sure Cisco doesn’t 
> want us to do anymore, then it’s either Tokyo or we’re SOL.
> 
>  
> 
> Very disappointing.
> 
>  
> 
> Dane Warner, CCVP
> 
> Sr. Network Engineer
> 
> Epoch Universal, Inc.
> 
> (909)226-0755
> 
> dwar...@epochuniversal.com 
> 
> 
> 
>  
> 
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of OSL StudyList
> Sent: Friday, September 27, 2013 3:19 AM
> To: Josh Petro
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP
> 
>  
> 
> Do you know what times the lab dates are released for those who have not 
> paid?   I thought it was at midnight SJC time but, I am not sure.  
> 
>  
> 
> On Fri, Sep 27, 2013 at 5:17 AM, Josh Petro  wrote:
> 
> If you mean Voice availability, then you are correct in that RTP is filled. 
> San Jose had a few open spots in Jan Feb last week. 
> I don't believe Collaboration dates are open yet for scheduling.
> Josh
> 
> On Sep 27, 2013 5:58 AM, "OSL StudyList"  wrote:
> 
> Is anyone having any luck scheduling exams at RTP or SJC?   When I try to 
> find an available date, I am seeing NOTHING available.  
> 
> 
> ___
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> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
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> 
>  
> 
> 
> ___
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> visit www.ipexpert.com
> 
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> 
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> 
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Re: [OSL | CCIE_Voice] CCIE Voice to CCIE Collaboration Details

[Alex Mendoza]

If you pass only the written on voice, you must take the lab before Feb 14th 
2014 and pass.

After this date you will stay on ground zero.

best regards
On Sep 19, 2013, at 6:57 AM, Edward Hawkins  wrote:

> So what happens if you only pass the ccie voice written.  Does that mean you 
> are at ground zero?
> 
> On Sep 18, 2013 1:42 PM, "Brian Schear"  wrote:
> They have the details out for CCIE Voice to CCIE Collaboration migration now.
> 
>  
> https://learningnetwork.cisco.com/docs/DOC-21915
> 
>  
> Brian Schear
> 
> CCIE #36045 (Voice)
> 
>  
>  
> 
> 
> This communication (including any attachments) is intended only for the use 
> of the individual or entity to which it is addressed, and may contain 
> information that is privileged, confidential and exempt from disclosure under 
> applicable law. If you are not the intended recipient, any dissemination, 
> distribution or copying of this communication is strictly prohibited. If you 
> have received this communication in error, please notify Vital Support 
> Systems at 515 334 5700 and delete or destroy all copies and the original 
> document.
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> 
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[OSL | CCIE_Voice] Forward to voicemail button

[Alex Mendoza]

Hi all

Please share your thoughts.
What option will you do on this one.

Create speed dial *3001 on device and then...

Note: is unity connection 

Option 1
Create a dn *3001 with fwd all to vm checked, VM profile mask with  to 
strip *

Option 2
Create a new partition and CSS 
Create a dn 3001 in the new partition with fwd all to vm
Create translation-pattern *.3001, strip * with predot and use the new CSS to 
match 3001 in the new dn.

Option 3
Create a dn *3001 with fwd all to vm.
On CUC user options add *3001 as an alternate number.

What option will you use and most important why?

I'll appreciate all your inputs.

Alejandro Mendoza 
Sent from my iPhone 
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[OSL | CCIE_Voice] doubts with digit manipulation

[Alex Mendoza]

Hello guysCould you please share your thoughts on this simple question.When I used mgcp on a gw the ip phone shows the called number without 9.but if I used H323 the called number on ipphone's shows the called number with 9, because all my dial-peer starts with 9 for SRST.My doubt is when im using a rute-list for TEHO or fileover on a specific route pattern.For examplesite a  users dial 9555777, match my route pattern 9.[2-9]XX, at this level I strip off the 9 on the Called Party Transformations.the route-patter have the Route-list sitea-fileover-siteb.first choice is SiteA (MGCP) route-group, set the calling to subscriber, 7 digits ani, and set called to sub 7 digits.at this path the ip phones shows "To 555"second choice is siteb (h323) route-group, set the calling to subscriber, 7 digits ani, and set the called with pre-dot, prefix 91212, to match the dial-peer.at this path the ip phones shows "To 91212555"If on real lab ask me to do a similar route-pattern I'll loose points if both paths doesn't shows "To 555" or it doesn't matter.sorry for my bad english.best regards!AM___
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[OSL | CCIE_Voice] TFTP service on subscriber.

[Alex Mendoza]

 Hi, All 

Is mandatory in Subscriber, TFTP server activated? 

I'll activate TFTP on sub until I finished all tasks (hope so) 

 What do you think?

best regards!
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Re: [OSL | CCIE_Voice] 7970 Phones are black dead how to recover them?

[Alex Mendoza]

Hi, Hesham.

Connect the ip phone (black screen)  to a PoE switch, turn on debug ip DHCP 
server all to see if the ip phone are trying to get an IP address.

If not, I don't know what to do, but If the phone are trying to get an IP 
address you can use a CME to put a correct firmware and bring back to live.


regards!


On Sep 3, 2013, at 2:28 AM, Hesham Abdelkereem  wrote:

> Dear Experts,
>  
> I have couple of 7970's using them for my homelab for practicing. Some of the 
> phones were frozen due to normal boot/upgrade process then went black and 
> unable to recover them.
> I have used this URL
>  
> http://greenwirecommunications.com/phone-systems/cisco-ip-phones/guide-faq-unbrick-reflash-cisco-7970g/
>  
> as well as tried to reboot then press # till lights became amber then release 
> # then put 123456789*0# as well as the other one 1673492850*#
>  
> All that never worked at all.
>  
> I have CUCM , CME , POE Switches , Laptop.
> Whats the best way to recover a phone from a black screen?
> When you connect the phone to a POE switch. I just see the headset , speaker 
> and mute button blinks in the beginning then nothing. No logo or anything is 
> shown.
>  
> Thank you very much in advance,
>  
> Hesham
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[OSL | CCIE_Voice] Multicast MoH.

[Alex Mendoza]

Hi All.Is there a solution on this...GW (h323) is configured with Outbound Fast Start using IOS MTP software and is working good."Media Termination Point Required" box checked"Enable Outbound FastStart" box checked with G711u-law 64KAlso, I configure Multicast MoH for this site and is working good for calls from other IP Phones on the cluster.but PSTN calls trough this h323 GW is not, when I place the call on hold, PSTN caller hear unicast moh.To solve this issue, I need to remove MTP required form H323 CUCM config.I see this is an expected behavior, see the note from cisco doc.http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmoh.htmlNote	The following restriction exists for multicast music on hold (MOH) when a media termination point (MTP) is invoked. When an MTP resource gets invoked in a call leg at a site that is using multicast MOH, the caller receives silence instead o music on hold. To avoid this scenario, configure unicast MOH or Tone on Hold instead of multicast MOH.Is there a trick to get multicast on a PSTN call, when "MTP required" is active on H323 GW?Any thoughts?Alex___
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Re: [OSL | CCIE_Voice] LAN QoS on HWIC-4ESW

[Alex Mendoza]

Hi Sam,My doubts is about how i'll mark packets on a remote site that are using HWIC-4ESW to connect IP phones. I see packets hitting the policy-map on HQ's serial interface, is because I configured this.-- SW-HQ --interface FastEthernet0/14 description hq ipphone mls qos trust device cisco-phone mls qos trust cos!interface FastEthernet0/10 description HQ TRUNK HQ TRUNK HQ mls qos trust dscp!!---Here is the config for WAN QoS.-- HQ --interface Serial0/1/0:0.1 point-to-point bandwidth 384 ip address 10.110.33.1 255.255.255.128 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 CISCO  class AutoQoS-FR-Se0/1/0:0-201  auto qos voip trust frame-relay ip rtp header-compression!!- Remote --interface Serial0/1/0:0.1 point-to-point bandwidth 384 ip address 10.110.33.2 255.255.255.128 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 CISCO  class AutoQoS-FR-Se0/1/0:0-101  auto qos voip trust frame-relay ip rtp header-compression!!---best regardsAlexOn Aug 27, 2013, at 12:33 AM, Sam Wilson  wrote:You will have to either trust the dscp marking done by the switch or you will have to mark them again. Please post your router configs and may be we can get a bit more detailedHthSent from my Windows PhoneFrom: Alex Mendoza Sent: 8/26/2013 10:54 PMTo: CCIE Study Subject: [OSL | CCIE_Voice] LAN QoS on HWIC-4ESWHi All,I'm wondering about QoS on remote site.I just applied auto-qos between HQ and remote site, everything looks fine.When I run this command on both routers, in remote site I can't see packets.HQ    Class-map: AutoQoS-VoIP-RTP-Trust (match-any)      11139 packets, 704260 bytes      5 minute offered rate 0 bps, drop rate 0 bps      Match: ip dscp ef (46)        11139 packets, 704260 bytes        5 minute rate 0 bps      Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0REMOTE    Class-map: AutoQoS-VoIP-RTP-Trust (match-any)      0 packets, 0 bytes      5 minute offered rate 0 bps, drop rate 0 bps      Match: ip dscp ef (46)        0 packets, 0 bytes        5 minute rate 0 bps      Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0For remote site I don't use HWIC-4ESW (expensive) instead am using ports from 3560 (same as HQ),after I put the commands:SW(config)#interface FastEthernet0/17SW(config-if)#shutSW(config-if)# mls qos trust device cisco-phoneSW(config-if)# mls qos trust cosSW(config-if)#no shutI started to see packets on policy-map.    Class-map: AutoQoS-VoIP-RTP-Trust (match-any)      186 packets, 11904 bytes      5 minute offered rate 0 bps, drop rate 0 bps      Match: ip dscp ef (46)        186 packets, 11904 bytes        5 minute rate 0 bps      Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0How I'll mark the packets on remote site to hit the policy-map in the FR link?Thanks in advancedMendoza___
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[OSL | CCIE_Voice] LAN QoS on HWIC-4ESW

[Alex Mendoza]

Hi All,I'm wondering about QoS on remote site.I just applied auto-qos between HQ and remote site, everything looks fine.When I run this command on both routers, in remote site I can't see packets.HQ    Class-map: AutoQoS-VoIP-RTP-Trust (match-any)      11139 packets, 704260 bytes      5 minute offered rate 0 bps, drop rate 0 bps      Match: ip dscp ef (46)        11139 packets, 704260 bytes        5 minute rate 0 bps      Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0REMOTE    Class-map: AutoQoS-VoIP-RTP-Trust (match-any)      0 packets, 0 bytes      5 minute offered rate 0 bps, drop rate 0 bps      Match: ip dscp ef (46)        0 packets, 0 bytes        5 minute rate 0 bps      Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0For remote site I don't use HWIC-4ESW (expensive) instead am using ports from 3560 (same as HQ),after I put the commands:SW(config)#interface FastEthernet0/17SW(config-if)#shutSW(config-if)# mls qos trust device cisco-phoneSW(config-if)# mls qos trust cosSW(config-if)#no shutI started to see packets on policy-map.    Class-map: AutoQoS-VoIP-RTP-Trust (match-any)      186 packets, 11904 bytes      5 minute offered rate 0 bps, drop rate 0 bps      Match: ip dscp ef (46)        186 packets, 11904 bytes        5 minute rate 0 bps      Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0How I'll mark the packets on remote site to hit the policy-map in the FR link?Thanks in advancedMendoza___
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[OSL | CCIE_Voice] AAR Not Attempting to ReRoute

[Alex Pishko]

All,

Hopefully someone can give me an idea on this one.  Working on my own
equipment but following the labs and am running into an issue with AAR.

I'm sending a call from HQ to BR 1 using 4 digit dial 5002 ---> 1002.  When
I attempt to make the call I receive the message Not enough bandwidth,
however I never see AAR actually get invoked.  In my setup as a quick way
to simulate congestion I set the bandwidth to 23 Kbps between Hub non (hq)
and BR1.

When I place the call from HQ as I said I get the banner of not enough
bandwidth but I never see the call actually hit the HQ gateway.  I've run
debug voip dialpeer, q931 as well voice ccapi inout and neither shows any
sort of traffic hitting the gateway.

In testing I can successfully dial into the HQ GW, I can dial emergency
services from the HQ phone, just doesn't seem like it's ever invoking AAR.

I also checked that the external number mask is correctly defined on the
1002 extension.

AAR CSS is assigned to the HQ phone, AAR group is assigned to the HQ line.
AAR group is prefixing 91 and there is a RP assigned to a partition that
falls within the AAR CSS that is for 91617XXX that has a RL pointed to
the HQ GW.

I did see earlier in the lab that they recommend using 7962 phones, however
I don't have any avaialble at the moment, so just wanted to make sure that
this might not be it.

Any help would be much appreciated.

Thank you
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[OSL | CCIE_Voice] mva

[Alex Mendoza]

Hi colleagues I set up mva and I think is working good, but I don't know if this is an issue.Calling from my SNR to MVA is working, MVA asks for my pin number, then press 1, after that I dialed internal 4 digit extension but this internal phone only shows the caller number and not the caller name.I think is normal behavior, but when a calling from my SNR directly to a internal extension, it shows the caller number and the caller id.What do you think ?best regards!AM 
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[OSL | CCIE_Voice] MGCP fail over

[Alex Mendoza]

Hi All

I configured BR1 as a MGCP gateway, and is working good.

Call takes Bchan 1 on incoming and outgoing calls.

My doubts are with mgcp-fallback. On SRST mode, outgoing calls take bchan 3. I 
used the "isdn bchan-number-order ascending" to fix this.

It's valid use this command or there is nothing to complain if I used this 
command or not.

thanks.

AA

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Re: [OSL | CCIE_Voice] Lab 4A BR1 MGCP Issue

[Alex Pishko]

Thanks, removing the l3 bind, doing a shut, no shut on the serial interface 
then adding the l3 bind statement back in did it. 

Thank you, much appreciated. 

On Jul 28, 2013, at 7:42, "ccie2k12"  wrote:

> try removing
>  
> ccm-manager config server 10.10.210.11  
> ccm-manager config
> 
> and remove
>  isdn bind-l3 ccm-manager
> and put it back again.
>  
> also
> BR1
>  pri-group timeslots 1-3,24
> PSTN
>  pri-group timeslots 1-3,23-24
>  
> should be same.
>  
>  
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Pishko
> Sent: Saturday, July 27, 2013 11:39 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Lab 4A BR1 MGCP Issue
>  
> All,
>  
> Struggling with the MGCG PRI circuit in this lab.  I've been through it a few 
> times and never had issues, however for this go around I cannot seem to get 
> L2 up when I issue the show isdn status command.  Also, when I look at the 
> serial inteface is shows up/up, however it's up/up (spoofing).
>  
> The MGCP portion appears to be workign correctly as when I issue a show 
> ccm-manager command I see it registered to the primary and within CUCM I see 
> the PRI as registered.  Below is my relevant configuration from the BR1 
> Router as well as the PSTN router.  Also, on the PSTN I've tried to change 
> the pri-group to just 1-3,24 both had the same result.  Also, when I issue 
> debug isdn q921 command I do not see any response for the SAMBE packet.  Any 
> help on this would be much appreciated.  I'm starting to think I may have a 
> hardware problem, but need a second or multiple sets of eyes on this before I 
> head down that router.
>  
> Thank you all.
>  
> Alex
>  
> BR1 Router
>  
> network-clock-participate wic 1 
> network-clock-select 1 T1 0/1/0
>  
> controller T1 0/1/0
>  framing esf
>  linecode b8zs
>  cablelength long 0db
>  pri-group timeslots 1-3,24 service mgcp
>  description **T1 Voice Connection to PSTN 0/1/1**
>  
> interface Serial0/1/0:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
>  
> voice-port 0/1/0:23
> !
> ccm-manager switchback immediate
> ccm-manager redundant-host 10.10.210.10
> ccm-manager mgcp
> ccm-manager music-on-hold
> ccm-manager config server 10.10.210.11  
> ccm-manager config
> !
> mgcp
> mgcp call-agent 10.10.210.11 2427 service-type mgcp version 0.1
> mgcp rtp unreachable timeout 1000 action notify
> mgcp modem passthrough voip mode nse
> mgcp package-capability rtp-package
> mgcp package-capability sst-package
> mgcp package-capability pre-package
> no mgcp package-capability res-package
> no mgcp timer receive-rtcp
> mgcp sdp simple
> mgcp rtp payload-type g726r16 static
> mgcp bind control source-interface Loopback0
> mgcp bind media source-interface Loopback0
>  
>  
> PSTN Router
>  
> network-clock-participate wic 1
>  
> controller T1 0/1/1
>  framing esf
>  clock source internal
>  linecode b8zs
>  cablelength long 0db
>  pri-group timeslots 1-3,23-24
>  description **T1 VOICE CONNECTION TO BR1-RTR 0/1/0**
>  
> interface Serial0/1/1:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-ni
>  isdn protocol-emulate network
>  isdn incoming-voice voice
>  no cdp enable
>  
> voice-port 0/1/1:23
>  translation-profile incoming block-call-into-BR1
>  translation-profile outgoing display-proper-ani-into-BR1
>  
>  
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[OSL | CCIE_Voice] Lab 4A BR1 MGCP Issue

[Alex Pishko]

All,
 
Struggling with the MGCG PRI circuit in this lab.  I've been through it a few 
times and never had issues, however for this go around I cannot seem to get L2 
up when I issue the show isdn status command.  Also, when I look at the serial 
inteface is shows up/up, however it's up/up (spoofing).
 
The MGCP portion appears to be workign correctly as when I issue a show 
ccm-manager command I see it registered to the primary and within CUCM I see 
the PRI as registered.  Below is my relevant configuration from the BR1 Router 
as well as the PSTN router.  Also, on the PSTN I've tried to change the 
pri-group to just 1-3,24 both had the same result.  Also, when I issue debug 
isdn q921 command I do not see any response for the SAMBE packet.  Any help on 
this would be much appreciated.  I'm starting to think I may have a hardware 
problem, but need a second or multiple sets of eyes on this before I head down 
that router.
 
Thank you all.
 
Alex
 
BR1 Router
 
network-clock-participate wic 1 
network-clock-select 1 T1 0/1/0
 
controller T1 0/1/0
 framing esf
 linecode b8zs
 cablelength long 0db
 pri-group timeslots 1-3,24 service mgcp
 description **T1 Voice Connection to PSTN 0/1/1**
 
interface Serial0/1/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
 
voice-port 0/1/0:23
!
ccm-manager switchback immediate
ccm-manager redundant-host 10.10.210.10
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.10.210.11  
ccm-manager config
!
mgcp
mgcp call-agent 10.10.210.11 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
 
 
PSTN Router 
 
network-clock-participate wic 1 
 
controller T1 0/1/1
 framing esf
 clock source internal
 linecode b8zs
 cablelength long 0db
 pri-group timeslots 1-3,23-24
 description **T1 VOICE CONNECTION TO BR1-RTR 0/1/0**
 
interface Serial0/1/1:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable
 
voice-port 0/1/1:23
 translation-profile incoming block-call-into-BR1
 translation-profile outgoing display-proper-ani-into-BR1___
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Re: [OSL | CCIE_Voice] CUPC

[Alex Mendoza]

My mistake, in the past a have this error.

The problem was that the ACL for the dmvpn on the router don't allow the subnet 
for one block of ip phones.

So I tried to said that my problem was in the router for an ACL misconfigured.

best regards 

Alejandro Mendoza 
Sent from my iPhone 

On 24/07/2013, at 23:59, Devakanth Gangavarapu  wrote:

> If there is a routing issue, the CUPC would not get registered to CUCM :) 
> 
> Dev
> 
> 
> On Thu, Jul 25, 2013 at 3:50 PM, Alex Mendoza  wrote:
>> Could be FW and maybe a wan accelerator as well
>> 
>> But you can not discard the possibility to be a routing issue
>> 
>> Signaling is from CUCM
>> 
>> RTP is from Ip phone and I supposed is in different subnet 
>> 
>> Hope you solve the issue soon
>> 
>> Kind regards 
>> 
>> 
>> Alejandro Mendoza 
>> Sent from my iPhone 
>> 
>> On 24/07/2013, at 22:43, Pavan K  wrote:
>> 
>>> Its most likely a firewall blocking rtp. Cannot be routes as the signaling 
>>> is OK (as you have ring back)
>>> On Jul 24, 2013 9:20 PM, "Alex Mendoza"  wrote:
>>>> Must check your routes
>>>> 
>>>> 
>>>> Try pinging the ip phone's address from  CUPC PC.
>>>> 
>>>> If it is unsuccessful do a tracert, to see which hop do not know how to 
>>>> reach the voice vlan.
>>>> 
>>>> 
>>>> I think is easy to figure out what is going on.
>>>> 
>>>> Best regards
>>>> 
>>>> Alejandro Mendoza
>>>> Sent from my iPhone 
>>>> 
>>>> On 24/07/2013, at 20:12, Dharambir kumar varma  
>>>> wrote:
>>>> 
>>>> > Hi Team.
>>>> >
>>>> > i have one phone CUPC over internet...and one cisco 7941 phone internal..
>>>> > both registered to call manager.
>>>> >
>>>> > when i call from cupc to 7941 or viceversa,,ring out happens and when
>>>> > call is connected, only dead air/ No audio..
>>>> > where can i check...
>>>> > ___
>>>> > For more information regarding industry leading CCIE Lab training, 
>>>> > please visit www.ipexpert.com
>>>> >
>>>> > Are you a CCNP or CCIE and looking for a job? Check out 
>>>> > www.PlatinumPlacement.com
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training, please 
>>>> visit www.ipexpert.com
>>>> 
>>>> Are you a CCNP or CCIE and looking for a job? Check out 
>>>> www.PlatinumPlacement.com
>> 
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com
> 
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Re: [OSL | CCIE_Voice] CUPC

[Alex Mendoza]

Could be FW and maybe a wan accelerator as well

But you can not discard the possibility to be a routing issue

Signaling is from CUCM

RTP is from Ip phone and I supposed is in different subnet 

Hope you solve the issue soon

Kind regards 

Alejandro Mendoza 
Sent from my iPhone 

On 24/07/2013, at 22:43, Pavan K  wrote:

> Its most likely a firewall blocking rtp. Cannot be routes as the signaling is 
> OK (as you have ring back)
> On Jul 24, 2013 9:20 PM, "Alex Mendoza"  wrote:
>> Must check your routes
>> 
>> 
>> Try pinging the ip phone's address from  CUPC PC.
>> 
>> If it is unsuccessful do a tracert, to see which hop do not know how to 
>> reach the voice vlan.
>> 
>> 
>> I think is easy to figure out what is going on.
>> 
>> Best regards
>> 
>> Alejandro Mendoza
>> Sent from my iPhone 
>> 
>> On 24/07/2013, at 20:12, Dharambir kumar varma  wrote:
>> 
>> > Hi Team.
>> >
>> > i have one phone CUPC over internet...and one cisco 7941 phone internal..
>> > both registered to call manager.
>> >
>> > when i call from cupc to 7941 or viceversa,,ring out happens and when
>> > call is connected, only dead air/ No audio..
>> > where can i check...
>> > ___
>> > For more information regarding industry leading CCIE Lab training, please 
>> > visit www.ipexpert.com
>> >
>> > Are you a CCNP or CCIE and looking for a job? Check out 
>> > www.PlatinumPlacement.com
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] CUPC

[Alex Mendoza]

Must check your routes 


Try pinging the ip phone's address from  CUPC PC. 

If it is unsuccessful do a tracert, to see which hop do not know how to reach 
the voice vlan.


I think is easy to figure out what is going on.

Best regards

Alejandro Mendoza 
Sent from my iPhone 

On 24/07/2013, at 20:12, Dharambir kumar varma  wrote:

> Hi Team.
> 
> i have one phone CUPC over internet...and one cisco 7941 phone internal..
> both registered to call manager.
> 
> when i call from cupc to 7941 or viceversa,,ring out happens and when
> call is connected, only dead air/ No audio..
> where can i check...
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] 7960 CUCM SIP Registration

[Alex Pishko]

Thanks all, it was firmware.  Forgot to mention before that I had made sure 
there was an extension and had checked replication as that was my inital 
thought as well.  Appreciate the assistance.
 
Alex
 


 From: Tony Zunt 
To: Alex Pishko  
Cc: "ccie_voice@onlinestudylist.com"  
Sent: Wednesday, July 17, 2013 9:49 PM
Subject: Re: [OSL | CCIE_Voice] 7960 CUCM SIP Registration
  


Hi Alex

Firmware.  I had a lot of old 7960s on hand and ran into the same issue.  
Basically had to try earlier and earlier revisions of SIP firmware to get them 
to work.  There's release notes on CCO for 7960 describing compatibility for 
SIP firmware 8.7.  

http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_notes_list.html
 
 
I had to go back all the way to v3 SIP firmware to get mine particular ones 
working.  Not sure why, but those phones were made so long ago, there were 
likely different hardware revisions. 

This guide has much good info and eventually helped me get them registered w/ 
SIP: 

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml
 

Thanks




On Wed, Jul 17, 2013 at 9:17 PM, Alex Pishko  wrote:

All,
>
>Having an issue with a 7960 registering to CUCM using SIP.  On the actual 
>phone itself keep getting the error registration rejected.  Within CUCM I also 
>see rejected on the phone page.   
>
>I've verifed multiple times that the MAC address is correct and my SIP profile 
>is pretty basic as there is no security applied to the phone itself.  This 
>should be something that's really simple as I have other SIP phones registered 
>to the cluster (not type 7960 though).  If I register the same phone via SCCP 
>it works fine.  Also, have tried to convert from SCCP to SIP using the BAT 
>tool; all with the same results. 
>
>Has anyone else seen this or have some additional input into what may be 
>happening?
>
>Thank you,
>Alex
>___
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>visit http://www.ipexpert.com/
>
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[OSL | CCIE_Voice] 7960 CUCM SIP Registration

[Alex Pishko]

All,
 
Having an issue with a 7960 registering to CUCM using SIP.  On the actual phone 
itself keep getting the error registration rejected.  Within CUCM I also see 
rejected on the phone page.  
 
I've verifed multiple times that the MAC address is correct and my SIP profile 
is pretty basic as there is no security applied to the phone itself.  This 
should be something that's really simple as I have other SIP phones registered 
to the cluster (not type 7960 though).  If I register the same phone via SCCP 
it works fine.  Also, have tried to convert from SCCP to SIP using the BAT 
tool; all with the same results.
 
Has anyone else seen this or have some additional input into what may be 
happening?
 
Thank you,
Alex___
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Re: [OSL | CCIE_Voice] SIP Gateway with Unity Connection issues

[Alex Mendoza]

Hi 


I'm not an expert on this field but I think the problem is the mod-call 
signaling 


Did you know if the IP address of your CUC is trusted? Maybe this behavior is 
for a toll-fraud prevention.

Regards 
AM

Sent from my iPhone 

On 16/07/2013, at 05:11, Ashok Boinpally  wrote:

> Hi Hesham,
> 
> Have you observed any change at VG for the call getting forwarded through 
> Unity AA to a cell phone? In general, it should be successful modification 
> irrespective of source of call (Phone B or Unity AA) unless the SIP headers 
> are different.
> 
> I guess forwarding through Unity AA is not working because SIP headers might 
> be different and not hitting the SIP profile meant for conversion 
> appropriately.
> 
> 
> On Tue, Jul 16, 2013 at 10:42 AM, Hesham Abdelkereem 
>  wrote:
>> yes I did that to make a normal call forwarding by doing so
>> 
>> oice class sip-profiles 1
>>  request INVITE sip-header Diversion modify "" 
>> "" where XXX is a real DID range that make 
>> it work with me when I call phone A to Phone B while Phone B is forwarded to 
>> cell phone but doesn't work when I call Unity AA to call Phone B while Phone 
>> B is forwarded to a cell phone
>> 
>> 
>> Thanks,
>> 
>> 
>> On 15 July 2013 19:36, Ashok Boinpally  wrote:
>>> Hello,
>>> 
>>> Have you tried to modify SIP header with SIP profiles on Cisco VG while 
>>> going finally out?
>>> 
>>> 
>>> On Tuesday, 16 July 2013, Hesham Abdelkereem wrote:
 Dear All,
 
 I have SIP Verizon and Unity Connection.
 I setup the Unity Connection Automated Attendant to make dial-by-extension 
 feature.
 
 Now suppose I have extension  is forwarded to a cell 408202
 
 If I called from PSTN to AA number then called extension  which is 
 forwarded to cell is not working.
 
 I did debug ccsip messasges and the reason why is because the remote-party 
 or ANI becomes the voicemail pilot
 
 this exactly related to that problem 
 http://www.gossamer-threads.com/lists/cisco/voip/148095
 
 
 
 How can I fix that in the SIP header? knowing that I did a change to let 
 Phone 1 calls Phone 2 and Phone 2 is forwarded to PSTN number
 
 that worked with me but didn't work when I do it via unity connection.
 
 
 Please give me some advice.
>>> 
>>> 
>>> -- 
>>> Ashok Kumar Boinpally.
> 
> 
> 
> -- 
> Ashok Kumar Boinpally.
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[OSL | CCIE_Voice] All Labs Table and Proctor Labs Vlahs

[Alex Pishko]

In setting up my sessions I've noticed the it doesn't appear that when you 
pre-load the configuration the proctor labs vlans match up with the vlan naming 
convention as to what's in the all labs table on the ipexpert site. Such as in 
all labs table hq server vlan is 30, but on preloaded config vlan is 103. Ip 
subnet matches up but vlan ID does not. Is this by design?  If so, should I 
chang the vlan tag in the actual lab?

Thanks,
Alex
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[OSL | CCIE_Voice] HELP with UCCXDv3

[Alex Mendoza]

Hi I want to do the lab guide for this workbook (UCCXDv3) but I don't know much about DB, and I'm stuck in the lab 3-4.I see in a reference that I need to follow this procedure to create the db that is needed in this workbook ... but  the sever does not have installed the MSSQL7, from where do I get the installer? thanks for read.1.  Start the Microsoft SQL Server Enterprise Manager

(i.e., Start | Programs | Microsoft SQL Server 7.0 | Enterprise Manager)

2.  Expand the Microsoft SQL Servers menu and highlight the SQL Server host:

+ Microsoft SQL Server Group
  + SQL Server Group
+ (hostname of your server)

The Getting Started Taskpad is now displayed in the right-hand window.

3.  Select the following options:

+ setup your database solution
  + create a database

The Welcome to the Create Database Wizard is now displayed.  Follow the
instructions on the screen.  Adjust the options as needed.
	+ Database name:  Cisco IP IVR
	+ Database Location:  C:\MSSQL7\data\  (if installing on CallManager DB)
	+ Transaction log file location:  C:\MSSQL7\data\
	+ Next, Use defaults to Finish, no Maintenance plan required.

4.  Start the SQL Server Query Analyzer

(i.e., Tools |  SQL Server Query Analyzer)

5.  Change the default database in the New query window (i.e., the DB: menu
option) to the one you created above in Step 3.

6.  Click the Load SQL Script icon in the New query window and select the
crs_training_db.sql script.  Click Open

7.  Click the Execute Query icon

8.  Verify the results:

+ Microsoft SQL Server Group
  + SQL Server Group
+ (hostname of your server)
  + Databases
+ (the name of your database; i.e., Cisco IP IVR)
  + Tables
- CUSTOMER
- CUSTOMER_ORDER
- HOLIDAYS
- ITEM
- PHYSICIAN_LOCATOR

Right-click on the Table name and select (Open Table | Return all rows).
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[OSL | CCIE_Voice] troubleshooting calls

[Alex Mendoza]

Hello,

I finished the kevin's videos, but I have some issues that are not mentioned on 
the video (or I missed).

1.- MoH between HQ and BR2 is not working, is this an expected behaviour?

2.- If I call to Br2 phone (4001) from HQ phone (2001) via gatekeeper, and 
brph2 divert to voicemail I got busy tone. 
I think problem is the codec mismatch because CUBE set call with G729, and my 
dial peer to reach CUE is G711

3.- When I call from PSTN to UCCX CSQ and none agent are ready, I heard the 
prompt audio, but If Agent's change to ready state, the call drops and the 
agent state changes to reserved.

I did some troubleshoot on MoH and voicemail busy tone but I got nothing.

Could someone point me out where I need to see to solve this issues?

best regards

AA
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[OSL | CCIE_Voice] UCCX issue

[Alex Mendoza]

Hello Experts

Silly question, 

I'm not looking in deep regarding this but let me ask first maybe is something 
easy to resolve.

When I call from PSTN phone to IPCC CSQ (task 9.2 from kevin's videos).

I should hear "thank you for calling all representative …" instead I hear "you 
for calling all representative..."

Seems like CSQ released the audio very quick and PSTN's Phone gets the audio 
with delay.


best regards
AA



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Re: [OSL | CCIE_Voice] Buying home lab or Waiting for Cisco Live in June

[Alex Mendoza]

Did you know the official date for new version?

I assume that I'll be ready for Sep/oct 2013

Best Regards

AA Mendoza 
Sent from my iPhone 

On 28/04/2013, at 10:33, Robert Thomas  wrote:

> Hi, 
> 
> I'm thinking on buying a home lab to start my studies. 
> It would run around 3K investment according to my amazon shopping list. 
> 
> However I'm thinking to wait for June and Cisco Live for announcement about 
> the new version. 
> 
> I don't expect major changes on the setup, perhaps some new phone models like 
> 99XX, or 89XX on the phones. And upgrade to the routers 29XX. 
> 
> However I don't expect major new features from the 29XX roll out on the exam. 
> 
> I would appreciate your opinions on this. 
> 
> -- 
> Robert Thomas Zamora
> tho...@gmail.com +50689389544
> http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
> CCNP, CCNP Voice
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Re: [OSL | CCIE_Voice] Help with building a homelab

[Alex Mendoza]

Which version of VMware did you use to create this images?

You can use VMware workstation 9 to change the compatibility to open with 
esxi5.1

Regards
AA

Sent from my iPhone 

On 21/04/2013, at 06:53, Hesham Abdelkereem  wrote:

> Dear Experts,
>  
> I am building my homelab now and I have bought an external hard drive that 
> contains all VMDK and VMX files for all clusters.
> I have Dell Power Edge 1950 III so I have installed VMWare ESXi v5.1 on the 
> server and for the cluster.
> I have connected the external hard drive into my laptop and then copied to 
> the local store of ESXi server.
> The storage is just local hard drives not SAN.
> After I have finished copying all the cluster to local datastore and then I 
> have added them to the inventory.
> When I try to power up the machine I get this error
>  
> Error message on CUPS7: Cannot open the 
> configuration file 
> /vmfs/volumes/51718924-525819d8-8c27-001d-
> 09644fc2/Datastore1/CUPS7/CUPS7.vmx. 
> error
> 4/20/2013 10:20:52 PM
> CUPS7
> root
>  
> Error message on UCXN7: Cannot open the 
> configuration file 
> /vmfs/volumes/51718924-525819d8-8c27-001d-
> 09644fc2/Datastore1/UCXN7/UCXN7.vmx. 
> error
> 4/20/2013 10:21:13 PM
> UCXN7
> root
>  
>  
> I am unable to open it and always getting error message cannot open the 
> config file.
>  
> Any ideas guys please
>  
>  
> Thanks,
> Hesham
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[OSL | CCIE_Voice] IP phone 7960

[Alex Mendoza]

Hello partners 

Can I do the training lab with this model   "7960" ?

I have doubts about the presence tasks

Thanks in advanced

AA
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Re: [OSL | CCIE_Voice] No List of Expected "Available Devices" in RG

[Alex Mendoza]

I think the GW was associated to a route pattern and as a consequence you can't 
see it in the route group configuration page.

When you deleted the GW, the route patter miss the gateway association and then 
you can see it in the route group configuration page.


AA Mendoza 
Sent from my iPhone 

On 10/12/2012, at 00:43, Suresh Bhandari  wrote:

> Solved with deleting and adding the same gateway and port but still 
> wondering why it is not listed earlier.
> 
> 
> On Mon, Dec 10, 2012 at 12:12 PM, Suresh Bhandari  wrote:
>> Hello Folks,
>> 
>> When adding rg-br1 on Vol 1 Lab 5A, I got no available devices except the 
>> one H.323 gateway (10.10.200.3) that I created.
>> 
>> There is another BR1-RTR.proctorlabs.com gateway registered with DS1-0 port. 
>> Show ccm-manager also displays it as registered.
>> 
>> Why not this port/Gw is available in the "Available Devices" list for Route 
>> Group? Have I missed something, somewhere?
>> 
>> TIA
>> 
>> -- 
>> Suresh Bhandari
> 
> 
> 
> -- 
> Suresh Bhandari
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com
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[OSL | CCIE_Voice] Network infrastructure - Frame Relay

[Alex Mendoza]

Hi Vir,
I complete this task using a ISR 2851(is the same that I'm using as PSTN), 
VWIC2-2MFT-T1/E1 cards and RJ45 cross cables.
Read this blogs/post.
http://pandaeatsbamboo.blogspot.mx/2012/02/frame-relay-switch-configuration.htmlhttp://blog.f85.net/2010/11/frame-relay-lab-switch-setup-with-t1.html
best regards
AA



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Re: [OSL | CCIE_Voice] rsvp bandwidth problem

[Alex Eldridge]

Thanks for the help DQ. MTP device pool was the issue. I thought I had checked 
everything twice. . . .details are key!


On Jun 3, 2012, at 10:25 AM, Dan Quinlan (daquinla) wrote:

> Make sure both MTPs are in the correct device pools / regions on UCM.  If 
> your site A MTP and site C MTP are accidentally in the same region and you 
> are configured to use G711 within the region, you will see this behavior: you 
> negotiate bandwidth to use G711 for RSVP but throttle back when the endpoints 
> answer and actually use G729. 
> 
> DQ
> d...@cisco.com
> 
> Sent from my iPhone
> 
> On Jun 2, 2012, at 4:28 PM, "Alex Eldridge"  wrote:
> 
>> I did see that bug, and already tried g729 for both "intra" and "inter" 
>> region audio in Service Parameters. Still, I see 96k initially requested 
>> (ringing). When the call connects I see 24k requested. GK isn't involved in 
>> this particular scenario. I simply have Site A (region / device pool) and 
>> Site C (region / device pool). G729 is hard set between the 2 regions. The 
>> CUCM version is 7.0.1.11000-2 and the MTP config on both site routers is 
>> below:
>> 
>> !
>> dspfarm profile 1 mtp
>>  codec g729r8
>>  codec pass-through
>>  rsvp
>>  maximum sessions software 10
>>  associate application SCCP
>> !
>> 
>> 
>> 
>> On Jun 2, 2012, at 3:04 PM, Jason Aarons (AM) wrote:
>> 
>>> Exact version of callmanager?  Google intraregion codec g729 bug
>>>  
>>> Intraregion Audio Codec Default to G729 to avoid CSCsl74701
>>>  
>>> From: ccie_voice-boun...@onlinestudylist.com 
>>> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Eldridge
>>> Sent: Saturday, June 02, 2012 12:22 PM
>>> To: ccie_voice@onlinestudylist.com
>>> Subject: [OSL | CCIE_Voice] rsvp bandwidth problem
>>>  
>>> 
>>> 
>>> I'm having an issue where I see the RSVP bandwidth request showing 96kbps, 
>>> instead of 40kbps, for a g729 call (on the initial b/w request while 
>>> ringing). I've double checked my config against the solutions guide and 
>>> verified region settings and MTP configuration are set properly. Is this a 
>>> CUCM 7.0 bug or am I missing something?
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please 
>>> visit www.ipexpert.com 
>>> 
>>> Are you a CCNP or CCIE and looking for a job? Check out 
>>> www.PlatinumPlacement.com 
>>> 
>>> 
>>> itevomcid
>> 
>> ___
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>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com

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Re: [OSL | CCIE_Voice] rsvp bandwidth problem

[Alex Eldridge]

I did see that bug, and already tried g729 for both "intra" and "inter" region 
audio in Service Parameters. Still, I see 96k initially requested (ringing). 
When the call connects I see 24k requested. GK isn't involved in this 
particular scenario. I simply have Site A (region / device pool) and Site C 
(region / device pool). G729 is hard set between the 2 regions. The CUCM 
version is 7.0.1.11000-2 and the MTP config on both site routers is below:

!
dspfarm profile 1 mtp
 codec g729r8
 codec pass-through
 rsvp
 maximum sessions software 10
 associate application SCCP
!



On Jun 2, 2012, at 3:04 PM, Jason Aarons (AM) wrote:

> Exact version of callmanager?  Google intraregion codec g729 bug
>  
> Intraregion Audio Codec Default to G729 to avoid CSCsl74701
>  
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Eldridge
> Sent: Saturday, June 02, 2012 12:22 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] rsvp bandwidth problem
>  
> 
> 
> I'm having an issue where I see the RSVP bandwidth request showing 96kbps, 
> instead of 40kbps, for a g729 call (on the initial b/w request while 
> ringing). I've double checked my config against the solutions guide and 
> verified region settings and MTP configuration are set properly. Is this a 
> CUCM 7.0 bug or am I missing something?
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com 
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com 
> 
> 
> itevomcid

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[OSL | CCIE_Voice] rsvp bandwidth problem

[Alex Eldridge]

I'm having an issue where I see the RSVP bandwidth request showing 96kbps, 
instead of 40kbps, for a g729 call (on the initial b/w request while ringing). 
I've double checked my config against the solutions guide and verified region 
settings and MTP configuration are set properly. Is this a CUCM 7.0 bug or am I 
missing something?
___
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[OSL | CCIE_Voice] Study Partner

[Alex Guardado]

I am looking for a study partner in the USA for CCIE Voice Lab.  E-mail me 
directly to touch base.___
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Re: [OSL | CCIE_Voice] UCCX Scripts End Step

[Alex K]

Guys,
 
Anybody can recommend UCCX script scenarious to practice for the Lab?
 
Thanks
Alex
 



Date: Wed, 25 Apr 2012 13:36:28 +0530
From: kew...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] UCCX Scripts End Step


Sometimes , our scripts send calls to an agent at the middle of the script.
In that case should we include an "End" step right below call-contact  step ?
 
Is it recommended / not recommended to have multiple End steps in a single 
script?
 
Thanks
 
 
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Re: [OSL | CCIE_Voice] (no subject)

[Alex Goh]

check also did the mva was enable for the user under end user page.

On Sat, Aug 27, 2011 at 1:00 AM, Mini Me  wrote:

> Did you enable it in Service Parameters?
>
> HTH
>
> From: Ray 
> Reply-To: Ray 
> Date: Fri, 26 Aug 2011 08:32:19 -0700 (PDT)
> To: "ccie_voice@onlinestudylist.com" 
> Subject: [OSL | CCIE_Voice] (no subject)
>
> when i dial 3033300 from 525 pstn 2nd and put the pin 12345, MVA IVR
> ask me to press 2 to turn on the remote dest or press 3 to turn it off.. it
> doesnt let me dial any extension.. help
>
> SB#sho run
> Building configuration...
>
>
> Current configuration : 5001 bytes
> !
> ! Last configuration change at 05:10:01 CST Fri Aug 26 2011
> ! NVRAM config last updated at 23:36:37 CST Mon Aug 22 2011
> !
> version 15.0
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname BR1
> !
> boot-start-marker
> boot-end-marker
> !
> !
> no aaa new-model
> !
> !
> !
> clock timezone CST -6
> clock summer-time CST recurring
> network-clock-participate wic 2
> !
> dot11 syslog
> ip source-route
> !
> !
> ip cef
> !
> ip dhcp pool phone
>network 177.2.11.0 255.255.255.0
>default-router 177.2.11.1
>option 150 ip 10.11.11.19
> !
> !
> no ipv6 cef
> !
> multilink bundle-name authenticated
> !
> !
> !
> !
> isdn switch-type primary-ni
> !
> !
> !
> voice class codec 1
>  codec preference 1 g711ulaw
>  codec preference 2 g729r8
> !
> voice class h323 1
>   h225 timeout tcp establish 3
>   call start slow
> !
> !
> !
> voice translation-rule 1
>  rule 1 // // type any unknown plan any isdn
> !
> voice translation-rule 2
>  rule 1 // // type any subscriber plan any isdn
> !
> voice translation-rule 3
>  rule 1 // // type any national plan any isdn
> !
> voice translation-rule 4
>  rule 1 // // type any international plan any isdn
> !
> voice translation-rule 5
>  rule 1 /.*\(3...\)/ /\1/
> !
> voice translation-rule 11
>  rule 1 /^3...$/ /972303&/ type any unknown plan any isdn
> !
> voice translation-rule 12
>  rule 1 /^3...$/ /303&/ type any subscriber plan any isdn
> !
> voice translation-rule 13
>  rule 1 /^3...$/ /972303&/ type any national plan any isdn
> !
> voice translation-rule 14
>  rule 1 /^3...$/ /+1972303&/ type any international plan any isdn
>  rule 2 /^4...$/ /+8522404&/ type any international plan any isdn
> !
> !
> voice translation-profile 911
>  translate calling 11
>  translate called 1
> !
> voice translation-profile INC
>  translate called 5
> !
> voice translation-profile intl
>  translate calling 14
>  translate called 4
> !
> voice translation-profile local
>  translate calling 12
>  translate called 2
> !
> voice translation-profile nat
>  translate calling 13
>  translate called 3
> !
> !
> voice-card 0
> !
> !
> application
>  service mva http://10.11.11.19:8080/ccmivr/pages/IVRMainpage.vxml
>  !
> !
> !
> !
> !
> !
> license udi pid CISCO2811 sn FTX0902D1ZG
> !
> redundancy
> !
> !
> controller T1 0/2/0
>  pri-group timeslots 1-4,24
> !
> !
> !
> !
> !
> !
> !
> !
> !
> interface Loopback0
>  ip address 18.1.1.1 255.255.255.0
>  ip ospf network point-to-point
>  h323-gateway voip interface
>  h323-gateway voip bind srcaddr 18.1.1.1
>  !
> !
> interface FastEthernet0/0
>  no ip address
>  shutdown
>  duplex auto
>  speed auto
>  !
> !
> interface FastEthernet0/1
>  no ip address
>  shutdown
>  duplex auto
>  speed auto
>  !
> !
> interface FastEthernet0/1/0
>  switchport trunk native vlan 12
>  switchport mode trunk
>  switchport voice vlan 11
>  spanning-tree portfast
>  !
> !
> interface FastEthernet0/1/1
>  switchport trunk native vlan 12
>  switchport mode trunk
>  switchport voice vlan 11
>  spanning-tree portfast
>  !
> !
> interface FastEthernet0/1/2
>  switchport trunk native vlan 12
>  switchport mode trunk
>  switchport voice vlan 11
>  spanning-tree portfast
>  !
> !
> interface FastEthernet0/1/3
>  switchport trunk native vlan 12
>  switchport mode trunk
>  switchport voice vlan 11
>  spanning-tree portfast
>  !
> !
> interface Serial0/0/0
>  no ip address
>  encapsulation frame-relay
>  no frame-relay inverse-arp
>  !
> !
> interface Serial0/0/0.101 point-to-point
>  ip address 177.0.101.2 255.255.255.0
>  frame-relay interface-dlci 602
> !
> interface Serial0/0/1
>  no ip address
>  shutdown
>  clock rate 200
>  !
> !
> interface Serial0/2/0:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn outgoing ie facility
>  isdn outgoing display-ie
>  no cdp enable
>  !
> !
> interface Vlan1
>  no ip address
>  !
> !
> interface Vlan11
>  ip address 177.2.11.1 255.255.255.0
>  ip helper-address 10.11.11.19
>  !
> !
> interface Vlan12
>  ip address 177.2.12.1 255.255.255.0
>  !
> !
> !
> router ospf 1
>  log-adjacency-changes
>  network 18.1.1.1 0.0.0.0 area 0
>  network 177.0.101.0 0.0.0.255 area 0
>  network 177.2.11.0 0.0.0.255 area 0
>  network 177.2.12.0 0.0.0.255 area 0
> !
> ip forward-protocol nd
> no ip http 

[OSL | CCIE_Voice] AIM-CUE CF Card Issue

[Alex Goh]

Hi Guys,

Hope someone can assist me on the following issue.

I've an AIM-CUE module which one day after the power cycle, I got the
following error during the module boot up.

"Not a cisco supported CF. Please use cisco supported CF and reinstall the
software."

Understand my CF card could be toasted as it can't even detected by the card
reader.

I've hence doing some google and saw the original AIM-CUE-1GBCF= cost even
more than the module itself. then someone mentioned an CF card with 2001888
sectors could be a cheaper replacement.

I've success to found two 1GB CF card with the same sector (one of it is
cisco router used cf 1gb), and since have a chance to borrow the other
working AIM-CUE CF card on hand, I've used fedora core 15 and "DDed" the
working CF to an Image with the following command

"dd if=/dev/sdb of=/image bs=32768"

and write the image to the two card that i've bought, but guess what, i've
not gotten any luck on this.

Still after using the bootloader to initialize the module, i still getting
the "not a cisco supported CF" error.

By any chance anyone successfully to run AIM-CUE on other CF card and can
shed some light?

P/S I'm using this module for my CCIE lab preparation, and it is a used
unit, hence there isn't anyway I could do RMA : (

Regards,
Alex
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Re: [OSL | CCIE_Voice] voice traffic EF outbound / inbount packet count

[Alex Goh]

Any kind soul able to advise?

On Thu, Aug 4, 2011 at 12:20 PM, Alex Goh  wrote:

> Hi Guys,
>
>
>
> Hope I can seek a little help here.
>
>
>
> I've MLP QOS configured on my lab and I notice that the ef packet count for
> outbound and inbound on the router WAN interface are almost identical.
>
>
>
> For example I've two office, HQ and Branch, both office have phones
> registered to CUCM located at HQ and there is a 384 WAN link between.
>
> both voice gateway are running MGCP. When the call is established, I notice
> the following output:
>
>
>
> HQ Router
>
>
>
> Service-policy input: EF-Inbound
>
>
>
> Class-map: AutoQoS-VoIP-RTP-Trust (match-any)
>
>   *332* packets, 21248 bytes
>
>   5 minute offered rate 6000 bps
>
>   Match: ip dscp ef (46)
>
> 332 packets, 21248 bytes
>
> 5 minute rate 6000 bps
>
>
>
> Class-map: class-default (match-any)
>
>   11 packets, 674 bytes
>
>   5 minute offered rate 0 bps, drop rate 0 bps
>
>   Match: any
>
>
>
>   Service-policy output: AutoQoS-Policy-Trust
>
>
>
> queue stats for all priority classes:
>
>   Queueing
>
>   queue limit 64 packets
>
>   (queue depth/total drops/no-buffer drops) 0/0/0
>
>   (pkts output/bytes output) 335/20770
>
>
>
> Class-map: AutoQoS-VoIP-RTP-Trust (match-any)
>
>   *335* packets, 20770 bytes
>
>   5 minute offered rate 8000 bps, drop rate 0 bps
>
>   Match: ip dscp ef (46)
>
> 335 packets, 20770 bytes
>
> 5 minute rate 8000 bps
>
>   Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0
>
>
>
> Branch Router
>
>
>
> Virtual-Access3
>
>
>
>   Service-policy input: EF-Inbound
>
>
>
> Class-map: AutoQoS-VoIP-RTP-UnTrust (match-any)
>
>   *190* packets, 12160 bytes
>
>   5 minute offered rate 1 bps
>
>   Match: protocol rtp audio
>
> 190 packets, 12160 bytes
>
> 5 minute rate 1 bps
>
>   Match: access-group name AutoQoS-VoIP-RTCP
>
> 0 packets, 0 bytes
>
> 5 minute rate 0 bps
>
>
>
> Class-map: class-default (match-any)
>
>   12 packets, 1712 bytes
>
>   5 minute offered rate 0 bps, drop rate 0 bps
>
>   Match: any
>
>
>
>   Service-policy output: AutoQoS-Policy-UnTrust
>
>
>
> queue stats for all priority classes:
>
>   Queueing
>
>   queue limit 64 packets
>
>   (queue depth/total drops/no-buffer drops) 0/0/0
>
>   (pkts output/bytes output) 188/11656
>
>
>
> Class-map: AutoQoS-VoIP-RTP-UnTrust (match-any)
>
>   *187* packets, 11594 bytes
>
>   5 minute offered rate 1 bps, drop rate 0 bps
>
>   Match: protocol rtp audio
>
> 187 packets, 11594 bytes
>
> 5 minute rate 1 bps
>
>   Match: access-group name AutoQoS-VoIP-RTCP
>
> 0 packets, 0 bytes
>
> 5 minute rate 0 bps
>
>   QoS Set
>
> dscp ef
>
>   Packets marked 188
>
>   Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0
>
>
>
>
>
>
>
>
>
> As my understanding, aren't the inbound ef packet count should be same or
> at least close to the number of the sender's outbound?
>
> for this example HQ is sending 335 packet and isn't Branch should be
> receiving 335 ef packet? (assuming no packet lost)
>
>
>
> Also i notice on the same router, the inbound / outbound ef packet count is
> almost identical, is that correct pattern for voice traffic?
>
> When checking the statistics on the phone and is showing the same.
>
>
>
> if I've have huge number mismatch on this will that means the call suffer
> quality issue?
>
>
>
>
>
> Appreciate if someone can enlighten me on this.
>
>
>
> Thanks
>
>
>
> Regards,
>
> Alex
>
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[OSL | CCIE_Voice] B-ACD call queue display on 7941

[Alex Goh]

Hi Guys,

I was trying the b-acd drop-through mode, and notice that "number of call in
queue" display doesn't show on the my 7941 if the hunt member is line 2 of
my 7941.
but if i change the line 1 as the hunt member, i can see the queue display.
However my other phone 7961 has no such issue when i configure line 1 or
line 2 as member. I've check the both phones's firmware and it is the same.

i have the following config

ephone-dn  2  octo-line
 number 5003
 name BR2 PHONE 3
 ephone-dn-template 1
!
ephone-dn  3  octo-line
 number 5001
 name BR2 PHONE 1
 ephone-dn-template 1

ephone-dn  9  dual-line
 number 5101 no-reg both
 name Agent
!
ephone-dn  10  dual-line
 number 5102 no-reg both
 name Agent

ephone-hunt 1 longest-idle
 pilot 5100
 list 5101, 5102

ephone  2
 device-security-mode none
 mac-address
 ephone-template 1
 type 7941
 button  1:2 2:10

!
ephone  3
 device-security-mode none
 mac-address
 ephone-template 1
 type 7961
 button  1:3 2:9

anyone encountered this before or is it 7941 phones behavior?

Regards,
Alex
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Re: [OSL | CCIE_Voice] Presence, CUPS and CUPC

[Alex Goh]

Hi Randall,

Thanks for the info. but I notice that this way, the contact is not
"dialable". the contact's presence status can be seen and IM is working, but
not calling the contact using deskphone/softphone mode. when edit the
contact's details the phone number is not saved.

is that the normal behaviour when LDAP is not used?

Regards,
Alex

On Mon, Jul 25, 2011 at 10:23 PM, Randall Saborio  wrote:

> That is correct. It is because of not using LDAP.
> The alternative is to log into http:///ccmuser  and log in
> with each user, then on the directory or contacts, create the contact from
> there where it works regardless of ldap integration.
>
>
> On Sat, Jul 23, 2011 at 11:10 PM, Vega Wong  wrote:
>
>> Hi
>>
>> Thanks for the advices,
>>
>> I sort out the offline issue, it was because I didnt setup DNS for CUPC to
>> connect to CUPS. Once thats setup, IM works as well as Presence status.
>>
>> Just one more thing, by manually adding the contact, I couldnt save the
>> phone number of the contact. That means I cant "place a call" to the contact
>> in CUPC. But it works when I use the phone control (in CUPC) and actually
>> dial the extension. Is this also due to not using ldap?
>>
>> Cheers
>>
>>
>>  --
>> * From: * Adil Shaikh ;
>> * To: * Vega Wong ;
>> * Cc: * ;
>> * Subject: * Re: [OSL | CCIE_Voice] Presence, CUPS and CUPC
>> * Sent: * Sun, Jul 24, 2011 4:51:08 AM
>>
>>
>> Hi Vega,
>>
>> You wrote:
>> "However, at the bottom of the CUPC, it always shown as
>> "Connected(limited)". Also, I cant search the contact within the CUPS.With
>> the contact added, it always shown as offline. "
>>
>> Connected (Limited) seems to be normal behaviour in absense of LDAP.
>> If you have added contact from the User page in CUPS then it should be
>> working provided in CUCM you have done Line Association with the User. (that
>> is on DN page, select the device, press line Edit Line Appearance then at
>> the bottom of the page Associate End User).
>>
>> HTH
>> -adil
>>
>>
>>
>>
>> On Fri, Jul 22, 2011 at 11:25 PM, Vega Wong wrote:
>>
>>> Hi Experts
>>>
>>> I am working on CUPS and CUPC at the moment, I just have some questions
>>> with the setup. So far I have done:
>>>
>>>- I have successfully set up the integration between CUCM and CUPS.
>>>As I can see all the users and the SIP trunks automatically appears in 
>>> CUPS.
>>>
>>>- IPPM works on IP phones, I can send messages between IP Phones
>>>- I can add contacts using IPPM, or the User page of CUPS. The
>>>contacts will shows up in CUPC
>>>- When I run the system troubleshooter in CUPS, no Red crosses shown.
>>>(Except those items I havent configured - LDAP, voicemail)
>>>
>>>
>>> My issue is with the CUPC, I can log in the system using the User ID. I
>>> can see the contact added through IPPM or CUPS User page. I can use the CUPC
>>> to control the IP phone. However, at the bottom of the CUPC, it always shown
>>> as "Connected(limited)". Also, I cant search the contact within the
>>> CUPS.With the contact added, it always shown as offline.
>>>
>>> I have read that I will need LDAP in order to make the presence status to
>>> work in CUPC, is that true? Can i make this work without the LDAP?
>>>
>>> Please help
>>>
>>> Cheers
>>>
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>
>>
>>
>> --
>>   .. . .
>> _7___|___|_|_|adil.sha...@gmail.com
>>
>> . .
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
>
> --
> Randall "da ill" Saborio
> CCIE Voice Wannabe #10054675811
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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[OSL | CCIE_Voice] Cisco Live 2011 Voice Techtorial

[Alex Goh]

Hi Guys,

By any chance any have the latest CCIE Voice Techtorial from Cisco Live 2011
and willing to share?

Thanks,

Regards,
Alex
___
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Re: [OSL | CCIE_Voice] MLP on FR issue

[Alex Goh]

Hi Guys,

Managed to resolve the issue. is my OSPF config causing it, as I notice
pinging the VT IP on both router is success.

Regard,
Alex


On Sun, Jul 17, 2011 at 9:24 PM, Alex Goh  wrote:

> Hi Guys,
>
> Hope I can seek a little help here, I've encountered the MLP broke the WAN
> link issue, which can't be resolved by any work around discussed before.
>
> What happened is I'm trying to configured an MLP for HQ and BR1 with auto
> qos and also manually, but both landed me on broken WAN where router reboot
> on HQ and BR1, even PSTN doesn't help. I've also try to remove the
> frame-relay interface-dlci command and add it back but no luck.
>
> The virtual template never having a status UP and no connectivity between
> HQ and BR1
>
> Appreciate if any can help on this.
>
> Regards,
> Alex
>
> My config is as below:
>
> *HQ:*
> class-map match-any AutoQoS-VoIP-RTP-Trust
>  match ip dscp ef
> class-map match-any AutoQoS-VoIP-Control-Trust
>  match ip dscp cs3
>  match ip dscp af31
> !
> !
> policy-map AutoQoS-Policy-Trust
>  class AutoQoS-VoIP-RTP-Trust
> priority percent 33
>  class AutoQoS-VoIP-Control-Trust
> bandwidth percent 5
>  class class-default
> fair-queue
>
> interface Serial0/0/1:0
>  no ip address
>  encapsulation frame-relay
>  frame-relay traffic-shaping
>
> interface Serial0/0/1:0.101 point-to-point
>  description WAN Link to BR1
>  bandwidth 384
>  snmp trap link-status
>  frame-relay interface-dlci 101 ppp Virtual-Template1
>   class AutoQoS-FR-Se0/0/1:0-101
>
> interface Virtual-Template1
>  bandwidth 384
>  ip address 142.1.67.1 255.255.255.252
>  ppp multilink
>  ppp multilink interleave
>  ppp multilink fragment delay 10
>  service-policy output AutoQoS-Policy-Trust
>
> map-class frame-relay AutoQoS-FR-Se0/0/1:0-101
>  frame-relay cir 364800
>  frame-relay bc 3648
>  frame-relay be 0
>  frame-relay mincir 364800
>
> Interface  IP-Address  OK? Method Status
> Protocol
> Serial0/0/1:0.101  unassigned  YES NVRAM  up
> up
> Serial0/0/1:0.201  142.1.67.5  YES NVRAM  up
> up
> Virtual-Access1unassigned  YES unset  down
> down
> Virtual-Template1  142.1.67.1  YES manual down
> down
> Virtual-Access2142.1.67.1  YES TFTP   up
> up
> Virtual-Access3142.1.67.1  YES TFTP   up
> up
>
> (config-subif)#do sh ppp multilink
>
> Virtual-Access3
>   Bundle name: R2
>   Remote Endpoint Discriminator: [1] BR1
>   Local Endpoint Discriminator: [1] HQ
>   Bundle up for 00:14:20, total bandwidth 384, load 1/255
>   Receive buffer limit 12192 bytes, frag timeout 1000 ms
>   Interleaving enabled
> 0/0 fragments/bytes in reassembly list
> 0 lost fragments, 0 reordered
> 0/0 discarded fragments/bytes, 0 lost received
> 0x5D received sequence, 0x2 sent sequence
>   Member links: 1 (max not set, min not set)
> Vi2, since 00:14:20, 480 weight, 470 frag size
> No inactive multilink interfaces
>
>
> *BR1:*
>
> class-map match-any AutoQoS-VoIP-Remark
>  match ip dscp ef
>  match ip dscp cs3
>  match ip dscp af31
> class-map match-any AutoQoS-VoIP-Control-UnTrust
>  match access-group name AutoQoS-VoIP-Control
> class-map match-any AutoQoS-VoIP-RTP-UnTrust
>  match protocol rtp audio
>  match access-group name AutoQoS-VoIP-RTCP
> !
> !
> policy-map AutoQoS-Policy-UnTrust
>  class AutoQoS-VoIP-RTP-UnTrust
>   set dscp ef
> priority percent 33
>  class AutoQoS-VoIP-Control-UnTrust
> bandwidth percent 5
>   set dscp af31
>  class AutoQoS-VoIP-Remark
>   set dscp default
>  class class-default
> fair-queue
>
> interface Serial0/0/1:0
>  no ip address
>  encapsulation frame-relay
>  frame-relay traffic-shaping
> !
> interface Serial0/0/1:0.101 point-to-point
>  bandwidth 384
>  frame-relay interface-dlci 101 ppp Virtual-Template1
>   class AutoQoS-FR-Se0/0/1:0-101
> !
> interface Virtual-Template1
>  bandwidth 384
>  ip address 142.1.67.2 255.255.255.252
>  ppp multilink
>  ppp multilink interleave
>  ppp multilink fragment delay 10
>  service-policy output AutoQoS-Policy-UnTrust
>
> ip access-list extended AutoQoS-VoIP-Control
>  permit tcp any any eq 1720
>  permit tcp any any range 11000 11999
>  permit udp any any eq 2427
>  permit tcp any any eq 2428
>  permit tcp any any range 2000 2002
>  permit udp any any eq 1719
>  permit udp any any eq 5060
> ip access-list extended AutoQoS-VoIP-RTCP
>  permit udp any any range 16384 32767
> !
> !
> map-class frame-relay AutoQoS-FR-Se0/0/1:0-101
>  frame-rel

[OSL | CCIE_Voice] MLP on FR issue

[Alex Goh]

Hi Guys,

Hope I can seek a little help here, I've encountered the MLP broke the WAN
link issue, which can't be resolved by any work around discussed before.

What happened is I'm trying to configured an MLP for HQ and BR1 with auto
qos and also manually, but both landed me on broken WAN where router reboot
on HQ and BR1, even PSTN doesn't help. I've also try to remove the
frame-relay interface-dlci command and add it back but no luck.

The virtual template never having a status UP and no connectivity between HQ
and BR1

Appreciate if any can help on this.

Regards,
Alex

My config is as below:

*HQ:*
class-map match-any AutoQoS-VoIP-RTP-Trust
 match ip dscp ef
class-map match-any AutoQoS-VoIP-Control-Trust
 match ip dscp cs3
 match ip dscp af31
!
!
policy-map AutoQoS-Policy-Trust
 class AutoQoS-VoIP-RTP-Trust
priority percent 33
 class AutoQoS-VoIP-Control-Trust
bandwidth percent 5
 class class-default
fair-queue

interface Serial0/0/1:0
 no ip address
 encapsulation frame-relay
 frame-relay traffic-shaping

interface Serial0/0/1:0.101 point-to-point
 description WAN Link to BR1
 bandwidth 384
 snmp trap link-status
 frame-relay interface-dlci 101 ppp Virtual-Template1
  class AutoQoS-FR-Se0/0/1:0-101

interface Virtual-Template1
 bandwidth 384
 ip address 142.1.67.1 255.255.255.252
 ppp multilink
 ppp multilink interleave
 ppp multilink fragment delay 10
 service-policy output AutoQoS-Policy-Trust

map-class frame-relay AutoQoS-FR-Se0/0/1:0-101
 frame-relay cir 364800
 frame-relay bc 3648
 frame-relay be 0
 frame-relay mincir 364800

Interface  IP-Address  OK? Method Status
Protocol
Serial0/0/1:0.101  unassigned  YES NVRAM  up
up
Serial0/0/1:0.201  142.1.67.5  YES NVRAM  up
up
Virtual-Access1unassigned  YES unset  down
down
Virtual-Template1  142.1.67.1  YES manual down
down
Virtual-Access2142.1.67.1  YES TFTP   up
up
Virtual-Access3142.1.67.1  YES TFTP   up
up

(config-subif)#do sh ppp multilink

Virtual-Access3
  Bundle name: R2
  Remote Endpoint Discriminator: [1] BR1
  Local Endpoint Discriminator: [1] HQ
  Bundle up for 00:14:20, total bandwidth 384, load 1/255
  Receive buffer limit 12192 bytes, frag timeout 1000 ms
  Interleaving enabled
0/0 fragments/bytes in reassembly list
0 lost fragments, 0 reordered
0/0 discarded fragments/bytes, 0 lost received
0x5D received sequence, 0x2 sent sequence
  Member links: 1 (max not set, min not set)
Vi2, since 00:14:20, 480 weight, 470 frag size
No inactive multilink interfaces


*BR1:*

class-map match-any AutoQoS-VoIP-Remark
 match ip dscp ef
 match ip dscp cs3
 match ip dscp af31
class-map match-any AutoQoS-VoIP-Control-UnTrust
 match access-group name AutoQoS-VoIP-Control
class-map match-any AutoQoS-VoIP-RTP-UnTrust
 match protocol rtp audio
 match access-group name AutoQoS-VoIP-RTCP
!
!
policy-map AutoQoS-Policy-UnTrust
 class AutoQoS-VoIP-RTP-UnTrust
  set dscp ef
priority percent 33
 class AutoQoS-VoIP-Control-UnTrust
bandwidth percent 5
  set dscp af31
 class AutoQoS-VoIP-Remark
  set dscp default
 class class-default
fair-queue

interface Serial0/0/1:0
 no ip address
 encapsulation frame-relay
 frame-relay traffic-shaping
!
interface Serial0/0/1:0.101 point-to-point
 bandwidth 384
 frame-relay interface-dlci 101 ppp Virtual-Template1
  class AutoQoS-FR-Se0/0/1:0-101
!
interface Virtual-Template1
 bandwidth 384
 ip address 142.1.67.2 255.255.255.252
 ppp multilink
 ppp multilink interleave
 ppp multilink fragment delay 10
 service-policy output AutoQoS-Policy-UnTrust

ip access-list extended AutoQoS-VoIP-Control
 permit tcp any any eq 1720
 permit tcp any any range 11000 11999
 permit udp any any eq 2427
 permit tcp any any eq 2428
 permit tcp any any range 2000 2002
 permit udp any any eq 1719
 permit udp any any eq 5060
ip access-list extended AutoQoS-VoIP-RTCP
 permit udp any any range 16384 32767
!
!
map-class frame-relay AutoQoS-FR-Se0/0/1:0-101
 frame-relay cir 364800
 frame-relay bc 3648
 frame-relay be 0
 frame-relay mincir 364800


Interface  IP-Address  OK? Method Status
Protocol
Serial0/0/1:0.101  unassigned  YES NVRAM  up
up
Virtual-Access1unassigned  YES unset  down
down
Virtual-Template1  142.1.67.2  YES manual down
down
Virtual-Access2142.1.67.2  YES TFTP   up
up
Virtual-Access3142.1.67.2  YES TFTP   up
up

(config-subif)#do sh ppp multi

Virtual-Access3
  Bundle name: R1
  Remote Endpoint Discriminator: [1] HQ
  Local Endpoint Discriminator: [1] BR1
  Bundle up for 00:14:40, total bandwidth 384, load 1/255
  Receive buffer limit 12192 bytes, frag timeout 1000 ms
  Interleaving enabled
0/0 fragments/bytes in reassembly list
0 lost fragments, 0 reordered
0/0 discarded fragments/bytes, 0 lost received
0x2 received sequence, 0x5F sent sequence
  Member links: 1 

Re: [OSL | CCIE_Voice] SIP Phones on Lab

[Alex Goh]

it would be best it we have access to a copy of the techtorial slides like
2010...

On Wed, Jul 13, 2011 at 1:18 AM, Brian Mulgrew wrote:

>
> "As was already announced, there will not be any Core Knowledge/Open Ended
> Questions after August 15, 2011"
>
> Er.. I thought these were no longer tested as of May 2010?
>
> Sent from my iPad
>
> On 12 Jul 2011, at 17:38, "Chris Martin"  wrote:
>
>
> 
> http://blog.ipexpert.com/2011/07/12/cisco-live-news-and-updates-ccie-voice/#more-7578
>
> Chris
>
> On Tue, Jul 12, 2011 at 10:45 AM, Bill Lake < 
> whl...@gmail.com> wrote:
>
>> do you have a link to Jason's blog
>>
>> On Tue, Jul 12, 2011 at 9:57 AM, Chris Martin < 
>> clm.c...@gmail.com> wrote:
>> > That was really what was said.  No SIP endpoints, TODAY, but they could
>> be
>> > there within the next two weeks, take that for what you will.  Jason's
>> blog
>> > entry did cover a lot of the info, I will say there was a lot
>> of emphasis
>> > from Ben on troubleshooting, and being able to get info from debugs and
>> > traces.  IE: they may ask you to post the relevant debug lines for h245
>> > negotiation or SIP negotiations.
>> > Chris
>> >
>> > On Tue, Jul 12, 2011 at 8:59 AM, Bryan Byrne < 
>> ccie.25...@gmail.com> wrote:
>> >>
>> >> Jason just posted a blog entry with information from the CCIE Voice
>> >> session at Cisco Live.  In his post he stated "There are no SIP phones
>> on
>> >> the currently available labs.  They are looking into including them,
>> but
>> >> that will require the development of a new lab."  Are we to assume that
>> the
>> >> lab blue print is incorrect and that SIP endpoints for CUCM and CUCME
>> are
>> >> not something we should be concerned with?  Was anyone else in the
>> session
>> >> that could provide some additional input?
>> >>
>> >> -Bryan
>> >> ___
>> >> For more information regarding industry leading CCIE Lab training,
>> please
>> >> visit www.ipexpert.com
>> >>
>> >> Are you a CCNP or CCIE and looking for a job? Check out
>> >> www.PlatinumPlacement.com
>> >
>> >
>> > ___
>> > For more information regarding industry leading CCIE Lab training,
>> please
>> > visit www.ipexpert.com
>> >
>> > Are you a CCNP or CCIE and looking for a job? Check out
>> > www.PlatinumPlacement.com
>> >
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Couldn't browse script/prompt repository

[Alex Goh]

Hi Ron,

Yes I already have an application using the default ICD script, but still I
can't browse or save on the editor's repository

Hi Santiago,

I'm log in with uccxadmin account. I know anonymous account will not be able
to run reactive script and browse/save script repository.

Regards,
Alex

On Mon, Jul 4, 2011 at 11:53 PM, Santiago Figueroa wrote:

> **
>
> When you logg in CCX editor to do that uccxadmin?
>
> ** **
>  --
>
> *De:* **ccie_voice-boun...@onlinestudylist.com** [mailto:**
> ccie_voice-boun...@onlinestudylist.com**] *En nombre de *Alex Goh
> *Enviado el:* Lunes, 04 de Julio de 2011 09:32 a.m.
> *Para:* OSL
> *Asunto:* [OSL | CCIE_Voice] Couldn't browse script/prompt repository
>
> ** **
>
> Hi Guys,
>
> I'm having a weird issue with my UCCX 7.0 (1) Build 109. When login to the
> CCX editor, I'm not able to save or browse the script/prompt by clicking the
> repository button
> (see attached). I've tried reinstalled my UCCX with the repair options,
> rebooted my UCCX server, created a new uccx admin, and tried running CCX
> editor from another box all no lucks.
>
> Also, I notice the UCCX only works on default script, whenever I "save as"
> a default script like icd.aef without modifying it to the
> script/system/default folder. the UCCX will turn into partial service
> state,. where "Application Manager" is the one that OOS.
>
> By the way, the uccxadmin end user account was assigned with CCM Super User
> and allow CTI control all group, if this is related.
>
> Can someone shed some light on this?
>
> Regards,
> Alex
>
> --
> La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para
> USO EXCLUSIVO de sus destinatarios. No está permitida su divulgación y/o
> reproducción sin autorización. Si ha recibido este mensaje y no le incumbe,
> le rogamos nos los comunique y proceda a su borrado. Gracias.
>
> Information included in this e-mail and attached files is CONFIDENTIAL and
> only for the EXCLUSIVE USE of the receivers. Circulation and/or copy without
> permission is not allowed. If you have received this e-mail and you are not
> the intended recipient, please let us know and erase the message and
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>
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[OSL | CCIE_Voice] Couldn't browse script/prompt repository

[Alex Goh]

Hi Guys,

I'm having a weird issue with my UCCX 7.0 (1) Build 109. When login to the
CCX editor, I'm not able to save or browse the script/prompt by clicking the
repository button
(see attached). I've tried reinstalled my UCCX with the repair options,
rebooted my UCCX server, created a new uccx admin, and tried running CCX
editor from another box all no lucks.

Also, I notice the UCCX only works on default script, whenever I "save as" a
default script like icd.aef without modifying it to the
script/system/default folder. the UCCX will turn into partial service
state,. where "Application Manager" is the one that OOS.

By the way, the uccxadmin end user account was assigned with CCM Super User
and allow CTI control all group, if this is related.

Can someone shed some light on this?

Regards,
Alex
<>___
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Re: [OSL | CCIE_Voice] Passed

[Alex Goh]

Hi George,

Big congratulation to you and you deserve it!

We appreciate your contribution to the list and hope to see you again.

Now go and get back your social life : )

Cheers,
Alex

On Wed, Jun 29, 2011 at 9:44 PM, George Goglidze  wrote:

> Hi all,
>
> As the subject suggests, it's official, I'm dual CCIE #19926 R&S and Voice
> starting yesterday.
>
> Finally, it's a relief, no more studying late, spending weekends on a
> computer, making calls, my neighbours think I escaped a psychiatric, after
> hearing voices at night "TEST VOICEMAIL FOR HQ PHONE 1" and my personal
> favourite "YOUR POSITION IN QUEUE IS" :)
>
> I would like to transmit a very special thank you to IPExpert, and
> especially Vik Malhi. He's definitely made difference for me as a trainer.
> Just when you think you know it all about something, he would come up with
> something to prove me wrong, to show me the gaps in my knowledge I didn't
> know existed.
>
> Thank you Vik!
>
> Thanks to everyone on this forum too, there are many good people on the
> forum, with big knowledge, and more importantly willing to share it. It was
> big fun, I enjoyed the process a lot.
>
> By the way, I made it technically on my first attempt. Well, I payed once
> only, although I went to exam 3 times.
> 1st time, technical problems, Cisco gave me free retake voucher, 2nd time,
> again technical problems, again free voucher.
> To be honest after so many free retakes, I'm not even sure if I really
> passed, or I was costing them too much so they decided to give it to me :-),
> good samaritans these cisco guys.
>
> Wish you all good luck, and don't get frustrated, and most importantly have
> fun.
>
> Regards,
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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Re: [OSL | CCIE_Voice] GK failover call display

[Alex Goh]

Hi Adil,

It is normal or expected behaviour where + sign will not be able to pass to
h323 gateway, u will to prefix using translation-rule on h323 gateway or
configured incoming prefix on cucm

from the CUCM Help page you will see this

SIP and MGCP gateways can support sending the international escape
character, +, for calls. H.323 gateways do not support the +


HTH

Regards,
Alex

On Wed, Jun 29, 2011 at 8:20 AM, Adil Shaikh  wrote:

> Hi all,
>
>
> I have configured route list with 1st choice as gatekeeper and 2nd choice
> as local PSTN.
> When I shut down the Gatekeeper, the call goes out from PSTN and back into
> branch gateway via PSTN as expected.
>
> debug isdn q931 shows the 'Calling Party Number' in +E164 format but the
> phone display calling party number without plus. The phone is 7965.
>
> Is this what you are getting on your phone? Is this normal behaviour?
>
> One branch site is H323 and other is CME.
>
>
> Thanks
> -adil
>
>
> --
>   .. . .
> _7___|___|_|_|adil.sha...@gmail.com
>
> . .
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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www.ipexpert.com

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Re: [OSL | CCIE_Voice] Setting up home equipment

[Alex Goh]

Hi Randall

If I'm not wrong, 7961 and 7941 series support the + dialing from call list,
which probably you need it to practice the globalization/localization call
routing.

Regards,
Alex

On Wed, Jun 29, 2011 at 8:08 AM, Rrcrumm  wrote:

> Hi
> I'm setting up a switch, router and phones and the proctorlabs racks. I
> plan on using 7960 phones because they are cheaper
>
> Is there any reason to get  7961's? I'm just keeping the cost down
>
> Thanks
> Randall
> Sent from my iPhone
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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Re: [OSL | CCIE_Voice] FRF.12 question

[Alex Goh]

Hi Duy,

Thanks I shall test it out and confirm. my understanding could be wrong.

Regards,
Alex

On Mon, Jun 27, 2011 at 10:04 PM, ccieid1ot  wrote:

> Alex, definitely needed.  Test it out by enabling it on HQ router than
> don't enable on BR2 router.  Now open up the cue gui and see if it operated
> correctly.
>
> duy
> ccie #27737 voice
>
> tmobile g2
> On Jun 27, 2011 6:17 AM, "Alex Goh"  wrote:
> > Hi Shrini,
> >
> > I guess what cristobal trying to mean is when he is using class based
> > shaping, instead of FRTS which required the command on physical
> interface,
> > do he need to care about the qos setting between HQ and BR1.
> >
> > By the way, I have one question though, for the case when FRTS was enable
> on
> > HQ Physical serial interface, do we need to enable FRTS also on the
> opposite
> > site? I remember I tried before and WAN link isn't broken...
> >
> > Alex
> >
> > On Mon, Jun 27, 2011 at 3:44 PM, Shrini  wrote:
> >
> >> **
> >> It looks like not effected but it is.
> >> The bandwidth drops to 56k.
> >> Good idea is to apply the Br2 service policy to Br1 connected srl
> interface
> >> even you not shaping the traffic.
> >>
> >> sh frame-relay pvc  will provide you the details.
> >>
> >> Thanks
> >> Shrini
> >>
> >> On 6/26/2011 2:34 PM, Cristobal Priego wrote:
> >>
> >> hello all
> >>
> >> when you configure FRF.12 manually on your seial interfaces on HQ and
> BR2
> >> on the HQ router where the same physical interface is used to connect
> BR1
> >> and BR2, BR1 link isn't affected at all because traffic shaping isn't
> >> enabled on the physical interface, correct ?
> >> so i can pretty much ignore that link in regards to a basic QoS config
> if
> >> not needed
> >>
> >> thanks
> >>
> >>
> >> ___
> >> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >>
> >>
> >> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
> >>
> >>
> >> ___
> >> For more information regarding industry leading CCIE Lab training,
> please
> >> visit www.ipexpert.com
> >>
> >> Are you a CCNP or CCIE and looking for a job? Check out
> >> www.PlatinumPlacement.com
> >>
>
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Re: [OSL | CCIE_Voice] how to retrieve list of Active IP Phones ONLY...

[Alex Goh]

Hi Rashid,

Looks like there is some issue with the RIS Data Collertor service as per
the screen capture.

Can you confirm that service is running on all the servers?

Regards,
Alex

On Mon, Jun 27, 2011 at 3:23 PM, Rashid Khan  wrote:

> Thanks Julien, that resolved my problem but partially not fully.
>
> I have 2 clusters here.
>
> When I am using this tool for Cluster A, I am getting required results, But
> when I use it for Cluster B, This tool only showing me 24 Registered Devices
> even thought there are almost 250 IP Phones register with this cluster. I am
> also attaching screen shot of RTMT tool output..
>
> Regards
> Rashid.
>
>
> --
> *From:* Julien Krieger 
> *To:* Rashid Khan 
> *Cc:* ccie voice 
> *Sent:* Thu, June 23, 2011 6:37:52 PM
> *Subject:* Re: [OSL | CCIE_Voice] how to retrieve list of Active IP Phones
> ONLY...
>
> Hi Rashid,
>
> RTMT is your tool !!!
> Download it into the plugin's section
>
> Julien
>
> 2011/6/23 Rashid Khan 
>
>> Dear Team,
>>
>> I want to know is there any way to findout a list of Currently active or
>> registed IP Phones with Call Manager.
>> Oneway to do this is, write nothing in Text Box and press Find
>> button, when I do this I also see non active devices Or the devices whose
>> Status is Not found also appearing.
>>
>> I only want list of phones which are Active or working currently,
>>
>> Regards
>> Rashid
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] remote dest profile

[Alex Goh]

it should show 5001

On Mon, Jun 27, 2011 at 1:38 PM, donny f  wrote:

> hi,
>
>
> when we config  Remote Dest Profile  for SNR.
>
> When call come from our PSTN  (6171234) phone to  UCM  ext phone , should
> it show   our ext   5001   or  showing the  SNR  (PSTN  #) ?
>
> for ie :we have SNR6171234and  match to 5001.
>
> ___
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Re: [OSL | CCIE_Voice] FRF.12 question

[Alex Goh]

Hi Shrini,

I guess what cristobal trying to mean is when he is using class based
shaping, instead of FRTS which required the command on physical interface,
do he need to care about the qos setting between HQ and BR1.

By the way, I have one question though, for the case when FRTS was enable on
HQ Physical serial interface, do we need to enable FRTS also on the opposite
site? I remember I tried before and WAN link isn't broken...

Alex

On Mon, Jun 27, 2011 at 3:44 PM, Shrini  wrote:

> **
> It looks like not effected but it is.
> The bandwidth drops to 56k.
> Good idea is to apply the Br2 service policy to Br1 connected srl interface
> even you not shaping the traffic.
>
> sh frame-relay pvc   will provide you the details.
>
> Thanks
> Shrini
>
> On 6/26/2011 2:34 PM, Cristobal Priego wrote:
>
> hello all
>
> when you configure FRF.12 manually on your seial interfaces on HQ and BR2
> on the HQ router where the same physical interface is used to connect BR1
> and BR2,  BR1 link isn't affected at all because traffic shaping isn't
> enabled on the physical interface, correct ?
> so i can pretty much ignore that link in regards to a basic QoS config if
> not needed
>
> thanks
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] IPPM getting invalid device

[Alex Goh]

oso check if you have already assign the user capability under licensing
capability

On Mon, Jun 27, 2011 at 4:47 AM, Cristobal Priego  wrote:

> does yourapp  user matches the user in CUPS ? is ip phone messenger on  in
> CUPS?
> do you have the CTI enabled and the CTI controll all devices group
> associated to your app user ?
>
> 2011/6/26 Alan Gardner 
>
>> "Invalid Device
>>
>> You were trying to access IP Phone Messenger service from a device not
>> provisioned on Cisco CallManager server. Please work with your system
>> administrator to get this device configured."
>>
>>  
>>
>> I am running CUCM 7 and CUPS 7 and I have completed the following steps:*
>> ***
>>
>>  
>>
>> 1. Configured IP PhoneMSG service with CUPS in URL
>>
>> 2. Created "PhoneMessenger" application user and added IPC and 7965 phones
>> in "Controlled Devices"
>>
>> 3. Associated end users with primary DNs on IPC and 7965 and configured
>> end users with "Standard CTI Enabled" user group permissions.
>>
>> 4. Subscribed both phones with "IP PhoneMSG" service
>>
>> ** **
>>
>> Any ideas
>>
>> ** **
>>
>> ** **
>>
>> Best Regards,
>>
>> ** **
>>
>> Alan Gardner
>>
>> ** **
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] MOH between HQ and BR1 Phones

[Alex Goh]

Guys,

Please disregards this message, I found out what's wrong already. Wrong
Region configured on MOH DP :P

Regards,
Alex

On Wed, Jun 22, 2011 at 10:21 PM, Alex Goh  wrote:

> Hi Guys,
>
>
> I was trying some MOH question just now and notice something which I don't
> understand. Basically my HQ and BR1 phone registered to CUCM and resides in
> device pool DP-HQ and DP-BR1, I've configured region between HQ and BR1 for
> G729 only. My MOH server reside in DP-MOH, which have region codec G711 to
> both HQ and SB.
>
> If I don't enable MOH with G729 codec, HQ Phone been put on hold, MOH OK,
> not before BR1 Phone. I hear Tone On Hold instead.
>
> But if I enable MOH with G729, both HQ & BR1 phones been put on hold and
> MOH working fine.
>
> In my case, since I already have MOH on different DP and Region codec G711,
> why would I need to enable G729 for MOH to work on BR1 phones?
>
> By the way, I was testing unicast MOH.
>
> Regards,
> Alex
>
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[OSL | CCIE_Voice] MOH between HQ and BR1 Phones

[Alex Goh]

Hi Guys,


I was trying some MOH question just now and notice something which I don't
understand. Basically my HQ and BR1 phone registered to CUCM and resides in
device pool DP-HQ and DP-BR1, I've configured region between HQ and BR1 for
G729 only. My MOH server reside in DP-MOH, which have region codec G711 to
both HQ and SB.

If I don't enable MOH with G729 codec, HQ Phone been put on hold, MOH OK,
not before BR1 Phone. I hear Tone On Hold instead.

But if I enable MOH with G729, both HQ & BR1 phones been put on hold and MOH
working fine.

In my case, since I already have MOH on different DP and Region codec G711,
why would I need to enable G729 for MOH to work on BR1 phones?

By the way, I was testing unicast MOH.

Regards,
Alex
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Re: [OSL | CCIE_Voice] CUE Voiceview Issue

[Alex Goh]

Hi Vinay & George,

Thanks for pointing that out, I will give it a try. forgot to search thru
the archives before posting : )

Alex

On Fri, Jun 17, 2011 at 6:51 PM, George Goglidze  wrote:

> Hi Alex,
>
> This has been discussed in numerous ocasions, here's one link to archives:
> http://onlinestudylist.com/archives/ccie_voice/2010-April/015608.html
>
> Regards,
>
> On Fri, Jun 17, 2011 at 10:07 AM, Alex Goh  wrote:
>
>> Hi Guys,
>>
>> Anyone encounter this issue before, after voiceview was configured on the
>> CUE and service subscribed to the IP Phone. I've able to login and see
>> number of message.
>> but when I tried to play the message or send a message, I always get
>> "Authentication error. Report this error to your system administrator". My
>> CUE is integrate with CUCM and verified correct license file was installed.
>>
>> It seems like something to do with the Authentication URL in the service,
>> anyone can shed some light on this?
>>
>> trying to do some trace on CUE and this is what I've got
>>
>>
>> 4519 06/17 17:00:15.815 vovw cont 0 Enter Controller Requested URI:
>> /voiceview/common/login.do
>> 4519 06/17 17:00:15.815 vovw sydb 0 /sw/apps/vui/vvconfig/enabled
>> 4519 06/17 17:00:15.816 vovw sydb 0 1
>> 4519 06/17 17:00:15.816 vovw cont 0 Host : 142.1.66.253
>> 4519 06/17 17:00:15.817 vovw cont 0 Connection : close
>> 4519 06/17 17:00:15.817 vovw cont 0 User-Agent :
>> Allegro-Software-WebClient/4.34
>> 4519 06/17 17:00:15.817 vovw cont 0 Accept : x-CiscoIPPhone/*,
>> text/*,image/png,*/*
>> 4519 06/17 17:00:15.817 vovw cont 0 Accept-Language : en_US
>> 4519 06/17 17:00:15.817 vovw cont 0 Accept-Charset :
>> utf-8,iso-8859-1;q=0.8
>> 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneModelName : CP-7961G
>> 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneSDKVersion : 7.0.1
>> 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneDisplay : 298,144,3,G
>> 4519 06/17 17:00:15.818 vovw sydb 0
>> /sw/apps/platformCapabilities/system/preferred_language
>> 4519 06/17 17:00:15.819 vovw sydb 0 en_US
>> 4519 06/17 17:00:15.819 vovw cont 0 Setting session locale en_US
>> 4519 06/17 17:00:15.819 vovw sydb 0 /sw/apps/monitor/ctrl/offline
>> 4519 06/17 17:00:15.820 vovw sydb 0 0
>> 4519 06/17 17:00:15.820 vovw cont 0 Center Controller Requested URI:
>> /voiceview/common/login.do
>> 4519 06/17 17:00:15.821 vovw sess 0  request
>> 4519 06/17 17:00:15.821 vovw sess 0 Querying the phone for its device
>> information.
>> 4519 06/17 17:00:15.953 vovw sess 0 Phone Model  : CP-7961G
>> 4519 06/17 17:00:15.953 vovw sess 0 Phone MAC Address: 001E138C3CFC
>> 4519 06/17 17:00:15.953 vovw sess 0 Phone Primary DN : 4001
>> 4519 06/17 17:00:15.953 vovw sess 0 Checking if PIN less login is
>> configured for 4001
>> 4519 06/17 17:00:15.985 VMSS vmdb 0 Request connection: inUse: 0, active:
>> 2
>> 4519 06/17 17:00:15.985 VMSS vmdb 0 Got connection: 0, inUse: 1, active: 2
>> 4519 06/17 17:00:15.985 VMSS vmdb 7 select mailboxid from vm_mbxusers
>> where owner=true and userdn='/sw/local/users/scph1';
>> 4519 06/17 17:00:15.989 VMSS vmdb 3 PERSONAL_000
>> 4519 06/17 17:00:15.989 VMSS vmdb 0 Freed connection: 0, inUse: 0, active:
>> 2
>> 4519 06/17 17:00:15.990 vovw sess 0 Found mailbox
>> 4519 06/17 17:00:15.990 vovw sess 0 PIN-less login: 0
>> 4519 06/17 17:00:15.990 vovw sess 0 checkPinLess false
>> 4519 06/17 17:00:15.994 vovw cont 0 Exit Controller Requested URI:
>> /voiceview/WEB-INF/screens/phoneobjects/CiscoIPPhoneInput.jsp
>> 4513 06/17 17:00:24.520 vovw cont 0 Enter Controller Requested URI:
>> /voiceview/common/login.do
>> 4513 06/17 17:00:24.520 vovw sydb 0 /sw/apps/vui/vvconfig/enabled
>> 4513 06/17 17:00:24.521 vovw sydb 0 1
>> 4513 06/17 17:00:24.522 vovw cont 0 Submit Type 'LOGIN'
>> 4513 06/17 17:00:24.522 vovw sydb 0 /sw/apps/monitor/ctrl/offline
>> 4513 06/17 17:00:24.523 vovw sydb 0 0
>> 4513 06/17 17:00:24.523 vovw cont 0 Center Controller Requested URI:
>> /voiceview/common/login.do
>> 4513 06/17 17:00:24.524 vovw sess 0 LOGIN request
>> 4513 06/17 17:00:24.539 VMSS vmdb 0 Request connection: inUse: 0, active:
>> 2
>> 4513 06/17 17:00:24.539 VMSS vmdb 0 Got connection: 1, inUse: 1, active: 2
>> 4513 06/17 17:00:24.539 VMSS vmdb 7 select mailboxid from vm_mbxusers
>> where owner=true and userdn='/sw/local/users/scph1';
>> 4513 06/17 17:00:24.544 VMSS vmdb 3 PERSONAL_000
>> 4513 06/17 17:00:24.544 VMSS vmdb 0 Freed connection

[OSL | CCIE_Voice] CUE Voiceview Issue

[Alex Goh]

alog.DialogChannel,type=Cisco
Media Channel,id=1,state=IN_USE,pendingState=null,groupId=0,locked=false]
6216 06/17 17:00:53.149 ACCN CMTS 0 MediaDialogChannel id=1,state=IN_USE
MDC::abort: return without waiting
6216 06/17 17:00:53.149 ACCN CMTS 0 MediaDialogChannel id=1,state=IN_USE
MediaManager resetted.
6216 06/17 17:00:53.149 ACCN LMED 0 PromptPlayer::closeStream
6216 06/17 17:00:53.149 ACCN LMED 0 PromptPlayer::resetSources
6216 06/17 17:00:53.149 ACCN LMED 0 PromptPlayer::rtpData close
6216 06/17 17:00:53.149 ACCN LMED 0 PromptPlayer::closeStream done
6216 06/17 17:00:53.149 ACCN LMED 0 Enter RTPRecorder:stopRecord
6216 06/17 17:00:53.149 ACCN LMED 0 closeStream(), port=0
6216 06/17 17:00:53.149 ACCN LMED 0 Stopping doublebuffer input stream
6216 06/17 17:00:53.149 ACCN LMED 0 Enter:
DoubleBufferDatagramInputStream:stop
6216 06/17 17:00:53.150 ACCN CHMG 0 CHANNEL_STATE_CHANGE:Channel has changed
state: Channel=null,Channel Class=com.cisco.dialog.DialogChannel,Channel
Type=Cisco Media Channel,Channel id=1,Channel implementation id=1,Old
state=IN_USE,New state=IDLE
6216 06/17 17:00:53.151 ACCN CHMG 0 GrpStubImpl:notifyChannelStateChange -
state change recvd from:
ChannelStub[channelClass=com.cisco.dialog.DialogChannel,type=Cisco Media
Channel,id=1,state=IDLE,pendingState=null,groupId=0,locked=true]
6216 06/17 17:00:53.151 ACCN CHMG 0 GrpStubImpl:notifyChannelStateChange -
channel changed from: IN_USE ---> IDLE
for-GroupStub[channelClass=com.cisco.wf.subsystems.cmt.CMTDialogChannelImpl,type=Cisco
Media Channel,id=0,name=0,state=IN_SERVICE,channel=6]
6216 06/17 17:00:53.151 ACCN CHMG 0 GrpStubImpl:notifyChannelIdle - No Group
state change - channel state changed from InUse --> Idle
6216 06/17 17:00:53.151 ACCN CHMG 0 GrpStubImpl:notifyChannelIdle - calling
async.release() to make channel available
6216 06/17 17:00:53.151 ACCN CHMG 0 ChStubImpl:releaseChannelLock() -
channel lock
releasedChannelStub[channelClass=com.cisco.dialog.DialogChannel,type=Cisco
Media Channel,id=1,state=IDLE,pendingState=null,groupId=0,locked=false]
6216 06/17 17:00:53.151 ACCN COMG 0 getApplicationPrivilege(): application
is null for contact: 6
6216 06/17 17:00:53.151 ACCN COMG 0 ContactManager: Releasing a port of
type: AAWorkflowPrivilege
6216 06/17 17:00:53.151 LLMA LAPI 0 Llama: vmPortDeallocate(): Deallocating
vm port for requestor "CRS"
6216 06/17 17:00:53.153 LLMA LSDB 0 Llama: vmPortDeallocate():
LlamaSysdbUser(): setString(): Setting
/sw/apps/limitsManager/vmPort/deallocate/requestor to CRS
2725 06/17 17:00:53.154 LLMA LSDB 0 LlamaVmPortDeallocate: check():
requestor, Value: CRS: returns true
2771 06/17 17:00:53.158 LLMA LVMP 0 LlamaVmPortDeallocate: commit():
Attribute: requestor, Value: CRS
2771 06/17 17:00:53.158 LLMA LLIC 0 LlamaLicense decrementCount VM Port
Feature - countsUsed = 0
6216 06/17 17:00:53.159 LLMA LAPI 0 Llama: vmPortDeallocate(): Deallocate vm
port for requestor "CRS" returns true
6216 06/17 17:00:53.159 ACCN COMG 0 ContactManager: Port of type
AAWorkflowPrivilege released successfully
6216 06/17 17:00:53.160 ACCN COMG 0 IVR port for privilege:
AAWorkflowPrivilege released by contact #6
4519 06/17 17:00:53.162 vovw vcmt 0 CiscoIPPhoneError: 4 null
4519 06/17 17:00:53.165 vovw cont 0 Exit Controller Requested URI:
/voiceview/WEB-INF/screens/phoneobjects/CiscoIPPhoneText.jsp

Thanks

Alex
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[OSL | CCIE_Voice] IPPM Add by Extension

[Alex Goh]

Hi Guys,

Anyone encounter this issue before? when I try to adding contact in IPPM
using the "AddByExt" options,
and it says no "UserID matches the extension...".

I've the extension number configured under the End User page, Telephone
Number field, also selected
the primary extension for the user. I did have the DN's associate with the
end user too.

I can added the contact using UserID, but not AddByExt, other than this, the
IPPM is working fine.

Thanks

Regards,
Alex
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Re: [OSL | CCIE_Voice] DEBUG MGCP PACKET

[Alex Goh]

Ops forgot about the url
http://ccie-musketeers.blogspot.com/2011/04/mgcp-call-preservation-porcess-with.html

On Sun, Jun 5, 2011 at 12:53 PM, Alex Goh  wrote:

> Hi Randall,
>
> See if this help. is about debug of MGCP packet during call preservation
> for CUCM failover.
>
>
> Alex
>
> On Sun, Jun 5, 2011 at 11:01 AM, Randall Crumm  wrote:
>
>> Hi,
>> Does someone have a good example of a debug mgcp packets and brief
>> explanation?
>>
>> RSIP/AUEP/AUCX
>>
>>
>> Thanks,
>> randall
>>
>>
>>
>> ___
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>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] DEBUG MGCP PACKET

[Alex Goh]

Hi Randall,

See if this help. is about debug of MGCP packet during call preservation for
CUCM failover.


Alex

On Sun, Jun 5, 2011 at 11:01 AM, Randall Crumm  wrote:

> Hi,
> Does someone have a good example of a debug mgcp packets and brief
> explanation?
>
> RSIP/AUEP/AUCX
>
>
> Thanks,
> randall
>
>
>
> ___
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[OSL | CCIE_Voice] Ignore Presentation Indicators?

[Alex Goh]

Hi Guys,

Anyone know what is the options Ignore Presentation Indicators (internal
calls only) does under RDP?
reading on the help it sound something to do with call display restriction,
but whether it check or unchecked I can't see any different.

Regards,
Alex
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Re: [OSL | CCIE_Voice] AIM-CUE CF card problem

[Alex Goh]

Hi Rab,

Thanks for the info, I wish it is good omen too : D

Hi Sam,

Thanks, but I guess it might not work, and the card reader can't even
detected the card, i guess the CF was toasted :(

Anyway look like I've no choice but get it off from ebay, as I don't have
extra time to search the Sandisk CF which possible compatible.


Regards,
Alex


On Wed, Jun 1, 2011 at 8:43 AM, Sam Park wrote:

> Alex;
>
> You need to re-image your CF with another known good CF.
> I just did this several weeks ago for a UC500 system.
> I got another CF from a good system, then I used my linux server to do a
> bit by bit copy of the CF using 2 USB multi-card readers.
> If you are not familiar with linux you can use the Ultimate Boot CD
> (partitioning as well as other utilities).
>
> So the hard thing might be getting a known good CF.
>
> Sam.
>
>
> On Tue, May 31, 2011 at 5:43 PM, ccieid1ot  wrote:
>
>> The mem is your ram, CF is your hd.   Try the sandisk CF, I'm sure cisco
>> just oem it from either sandisk or another manufacturer.
>>
>> duy
>> ccie #27737 voice
>>
>> tmobile g2
>> On May 31, 2011 1:22 PM, "Alex Goh"  wrote:
>> > Hi Guys,
>> >
>> > Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me
>> just
>> > 1 week before my exam!
>> > I've getting the error of "Not a cisco supported CF. Please use cisco
>> > supported CF and reinstall the software. System Halted." Anyone know how
>> to
>> > solved this issue?
>> >
>> > I've try to reinstall CUE using the boothelper, but no luck. Possibly
>> the CF
>> > card is gone case.
>> > A search on google mentioned Cisco AIM-CUE check on the CF Card sector
>> size,
>> > else refuse to work. But the used 1GB CF card was asking half the price
>> of
>> > the AIM-CUE module /w 1GB CF itself on ebay :(
>> >
>> > It is anyway I can used on 3rd party CF card? saw it also certain
>> SANDISK CF
>> > might work, but I'm not sure it is still able to find in the market now.
>> >
>> > Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than
>> > AIM-CUE-1GBCF, I wonder will it able to use?
>> >
>> > Any help will be appreciated.
>> >
>> > Regards,
>> > Alex
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
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[OSL | CCIE_Voice] AIM-CUE CF card problem

[Alex Goh]

Hi Guys,

Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me just
1 week before my exam!
I've getting the error of "Not a cisco supported CF. Please use cisco
supported CF and reinstall the software. System Halted." Anyone know how to
solved this issue?

I've try to reinstall CUE using the boothelper, but no luck. Possibly the CF
card is gone case.
A search on google mentioned Cisco AIM-CUE check on the CF Card sector size,
else refuse to work. But the used 1GB CF card was asking half the price of
the AIM-CUE module /w 1GB CF itself on ebay :(

It is anyway I can used on 3rd party CF card? saw it also certain SANDISK CF
might work, but I'm not sure it is still able to find in the market now.

Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than
AIM-CUE-1GBCF, I wonder will it able to use?

Any help will be appreciated.

Regards,
Alex
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Re: [OSL | CCIE_Voice] IPPM on cisco7961 didn't alert

[Alex Goh]

Hi Ki Wi,

I've encounter the same issue also, and I solved it by changing the
Enterprise Parameters Services URL to IP instead of hostname (Apparently, I
miss that part when I reverted my VMware snapshot), remember I saw this
solution from OSL discussion before.

HTH

Cheers,
Alex

On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong wrote:

> Frens,
>
> If i can recall correctly, that was due to that i missed associate the
> phone with
> application user "Phone Messenger". You need the phone messenger
> application user
> to control the IPPM user.
> Without the association, the messaging will still work but funny stuff come
> out, if not wrong
>
> Shingei.
>
>
>
>
>
> On Tue, May 31, 2011 at 4:52 AM, Ki Wi  wrote:
>
>> Hey,
>> Do you still remember how did you resolve this alert issue? I'm still
>> trying to train myself up in CUPS. Last night, my alert was working, my IPPM
>> login wasn't. Today my IPPM is working but no alert. =( All other components
>> are working.
>>
>>
>> On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong wrote:
>>
>>> Guys,
>>> Pls ignore this mail, has managed to figured out the caused.
>>>
>>> thanks
>>> Shingei.
>>>
>>>
>>> On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong wrote:
>>>
>>>> Hi,
>>>> I've configure the IPPM on cisco 7961 phone,
>>>> everything works smooth other that the message receive alert.
>>>> It doesn't "ring" when there is a mgs come in from CIPC or
>>>> other IPPM.i've set the "audible alert" to ON but still got
>>>> no luck.
>>>>
>>>> Another IPPM phone encounter the same issue, so don't think
>>>> is the phone problem. Any idea?
>>>>
>>>>
>>>> Thanks
>>>> Shingei.
>>>>
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Excellent document on LAN QoS

[Alex Goh]

Thanks Shrini for the link, coincidentally I was reading this also, and also
I found another blog post by Joe Astorino that is good reading on the LAN
QOS, although it used 3560 and more for CCIE R&S exam.

Cheers,
Alex

On Wed, May 18, 2011 at 4:54 PM, Shrini  wrote:

>  Just gone through Vik's documentation on LAN QoS , I liked the flow
> chart. Hopefully it is helpful to you too .. so thought of sharing.
>
>
> http://blog.ipexpert.com/2011/05/16/campus-qos-part-1-classification-and-marking-on-the-catalyst-3750/
>
> Thanks
> Shrini
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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Re: [OSL | CCIE_Voice] Call Manager and CUE Integration

[Alex Goh]

do you mind to post your config on your HQ and CUE?

Alex

On Tue, May 17, 2011 at 3:19 AM, Stephen Manuel wrote:

> In my home lab, I have the following
>
>
>
> 2811 router w/NM-CUE module w/7.0.1 software and CCM license.
>
> VM Ware Call Manager 7.0.1 software
>
>
>
> Router has VWIC-2MFT-T1 cards that are connected to my BR1 and BR2 routers
> both w/VWIC-1MFT-T1 cards, all are showing multi frame established.
>
> HQ router is MGCP controlled and contains the CUE Module.
>
> I have CFB and Xcoder resources registered in UCM for the HQ router
>
> I’ve rechecked the region, mrgl, location settings in UCM and they appear
> to be correct.
>
>
>
> Originally I had the CUE module working when the router was a CME router,
> however I tried to wipe out the router config and start over.
>
>
>
> I have all the phone and gateway registered with UCM.
>
>
>
> I then reinstalled the CUE license and software to make it work with Call
> Manager vs. CME.
>
>
>
> I have the CUE ports registered in Call Manager
>
> The CUE is showing it’s registered with Call Manager.
>
>
>
> The issue is when I press the messages button on  a phone that has a VM
> box, I get an immediate fast busy.
>
>
>
> When I call from another phone and the call rolls to VM same result,
> immediate fast busy.
>
>
>
> I’m sore of stumped, I’ve suspected that the issue is Codec related, but
> I’m unsure how to go about determining that.
>
>
>
> Any basic guidance would be greatly appreciated.
>
>
>
> Stephen Manuel
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] busy trigger per button

[Alex Goh]

Hi Voiceboy,

This is my understanding.

huntstop usually is used on shared ephone-dn to limit the incoming call.

if you just wanted to have the next incoming call during an active call,
busy trigger will do the trick.

whereas for max-call-per button, if it is an octo-line, the default is
already 8

source:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_m1ht.html#wp1093876

Cheers,
Alex

2011/5/8 voice boy 

>
> Hi,
>
> I need to ask about to use huntstop channel 1 in srst
> or to use busy trigger per button 1
>
> So that the calls  will be forwarded to voicemail if ot have active call
>
> Also do i need to use maximum calls per button 4 to be as cucm while it is
> in srst ??
>
>
>
> Thanks
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Vol2 Lab 2 Supplementary services with GK Trunk

[Alex Goh]

Hi Amit,

have to you tried to disable the wait for FE H245 TCS?

what is the codec that u configured for the MTP? mind to post ur MTP config
on the HQ-RTR here?

Regards,
Alex

On Sat, May 7, 2011 at 9:33 PM, amit batra  wrote:

> Hello Everyone.
>
>  I have finished IPexperts Volume 2 lab 2 .. I managed to finish
> everything apart from VPIM and Supplementary services ...VPIM  is a license
> issue so not worried ..
>
> I have configured a GK trunk between CUCM and HQ router (Gatekeeper) and
> BR2 router.
>
> all endpoints are registered to the gatekeepers. No probs till here ..
> Call from CUCM to CME and vice versa are working.
>
> The problem i am facing is , when i make a call from CUCM phone 5001 (SCCP)
> to a CME phone 3002 (SCCP) audio works fine. i can press hold button on the
> CUCM phone. When i do that on CME phone i hear beep. but when i press resume
> on CUCM phone, CME phone keep's giving that beep sound. when i press hold
> button on my CME and resume , audio start to flow again..
>
> I have configured software MTP on HQ router. Device pool assigned to the
> GK-Trunk and this software MTP is the same .
>
> On GK-Trunk MRGL is assigned ..
> Media Termination Point Required (ON)
> Retry Video Call as Audio (ON)
> Wait for Far End H.245 Terminal Capability Set (ON)
> Inbound faststart enabled (ON)
>
> when i make a call from any device , i can see that my IOS MTP is invoked
> and participating in the call .." show sccp connections"
>
> Am i missing anything here ? or do i need to enable anything else..?
>
> I hope i am making some sense.. If the question is not clear please let me
> know. and 1.30 am i cannot write anything more than this..
>
> Thanks in advance ..
>
> Regards
> Amit
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Meet Me + Ad-hoc Participant (WB2 Lab1 6.2)

[Alex Goh]

Hi Claude,

Thanks for the reply, I do have the hardware conference bridge configured on
the router, with support of
G729 codec of course. That verify by BR1 Phones can join into meet me
conference by dialing the number.

Issue come in when one of the meetme participant trying bring in another
participant on different codec
region into the bridge.

Thanks

Regards,
Alex

On Sat, Apr 23, 2011 at 6:59 PM, Friderich Claude wrote:

>  Hello,
>
>
>
> I think you are wrong for this question.
>
> You must invoke a conference bridge on the router and thanks to the
> hardware conference bridge it will support g729
>
>
>
> Just add the codec g729r8 in the dspfarm profile conference ….
>
>
>
> Should work.
>
>
>
> Of course do not forget to put your conference bridge in the MRG and MRGL
> of your CCM
>
>
>
> Regards
>
>
>
> Claude.
>
>
>
> *Claude Friderich*
>
> *PreSales Support*
>
> *[image: ccvp_voice_sm]***
>
> *NETCORE PSF S.A.***
>
> 49 rue du Baerendall
>
> B.P.65 L-8201 Mamer
>
> Téléphone: 31 33 80-407
>
> Fax: 31 33 80 8-407
>
> GSM: 621 303 616
>
> E-mail: cfrider...@netcore.lu
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Alex Goh
> *Sent:* samedi 23 avril 2011 12:10
> *To:* OSL
> *Subject:* [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)
>
>
>
> Hi All,
>
> I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have
> the MeetMe conference setup successfuly,
> HQ Phn, BR1 & PSTN all can dial into MeetMe number too. The H/W Conference
> Bridge was configured in HQ-RTR,
> and I can verified it was utilized.
>
> The problem comes in when I try to add an ad-hoc participant into the
> MeetMe Conference Bridge on different region.
> e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc
> participant BR1 PH1 which is on G729,
> the BR1 PH1 will get dropped.
>
> I know this is due to codec mismatch issue (verified by changing region
> from HQ to BR1 as G711, it works fine), but I've
> transcoder added in both HQ & BR1 DP MRGL. It looks like the transcoder
> doesn't get invoked in this case or do transcoder
> needed to get this working? since I already have H/W Conference configured.
>
> Appreciate if anyone can shed more light on this.
>
> Thanks
>
> Regards,
> Alex
>
>
>
> --
>
> This email was Anti Virus checked.
>
>
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[OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)

[Alex Goh]

Hi All,

I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have
the MeetMe conference setup successfuly,
HQ Phn, BR1 & PSTN all can dial into MeetMe number too. The H/W Conference
Bridge was configured in HQ-RTR,
and I can verified it was utilized.

The problem comes in when I try to add an ad-hoc participant into the MeetMe
Conference Bridge on different region.
e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc
participant BR1 PH1 which is on G729,
the BR1 PH1 will get dropped.

I know this is due to codec mismatch issue (verified by changing region from
HQ to BR1 as G711, it works fine), but I've
transcoder added in both HQ & BR1 DP MRGL. It looks like the transcoder
doesn't get invoked in this case or do transcoder
needed to get this working? since I already have H/W Conference configured.

Appreciate if anyone can shed more light on this.

Thanks

Regards,
Alex
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Re: [OSL | CCIE_Voice] Device mobility issue.

[Alex Goh]

Hi Vik,

The device mobility is on for the phone, and the subnet is correctly
attached for BR1 and HQ site. I'm will try to redo the DMI see how it goes.


Thanks

Regards,
Alex

On Sun, Apr 17, 2011 at 11:38 AM, Vik Malhi  wrote:

> Either roaming is not enabled for the phone. Or you have not attached the
> subnet of the HQ site to the HQ device pool (using device mobility info
> under the system menu).
>
>
>
> --
> Vik Malhi – CCIE #13890
> Managing Partner / Instructor - IPexpert, Inc.
> Mailto: vma...@ipexpert.com
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Live Assistance, Please visit: <http://www.ipexpert.com/chat>
> www.ipexpert.com/chat
> <http://www.ipexpert.com/chat>
>
> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio
> Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
> (R&S, Voice, Wireless, Security & Service Provider) certification(s) with 
> training
> locations throughout the United States, Europe, South Asia and Australia.
> Be sure to visit our online communities at www.ipexpert.com/communities <
> http://www.ipexpert.com/communities>  and our public website at
> www.ipexpert.com <http://www.ipexpert.com/>
>
>
> On Apr 16, 2011, at 20:22, Alex Goh  wrote:
>
> Any kind soul able to help?
>
> On Fri, Apr 15, 2011 at 5:17 PM, Alex Goh < 
> ncsalex@gmail.com> wrote:
>
>> Any kind soul able to help?
>>
>>
>> On Thu, Apr 14, 2011 at 5:40 PM, Alex Goh < 
>> ncsalex@gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> Understand that when an IP Phone was roaming and physical location of
>>> home DP and roaming DP is different,
>>> the roaming sensitive setting of the roaming DP will apply to the phone.
>>> However, when i moved my BR1 phone
>>> to HQ, the "View Current Device Mobility Settings" of the BR1 phone
>>> showing the roaming DP is "Not Selected",
>>> I believe it shouldn't be that way?
>>>
>>> Also I notice the Date Time Group on the roaming phone doesn't follow the
>>> roaming DP, i thought DTG suppose
>>> to be roaming sensitive setting and will apply to the roaming phone?
>>>
>>> Thanks
>>>
>>> Regards,
>>> Alex
>>>
>>
>>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit <http://www.ipexpert.com>www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] DHCP Issue with FRTS?

[Alex Goh]

Hi George,

Thanks for pointed out could be fragmentation issue. I notice my map-class
on HQ-RTR was missing the fragment statement. I believe
this could be the issue, going to try it out tomorrow.

For some reason, that day after I tried to use auto qos for FRF.12 and the
map class doesn't created automatically, hence I've manual
create map-class which mistakenly left out the fragmentation statement.

Thanks again George.

Regards,
Alex

On Sun, Apr 17, 2011 at 1:24 AM, George Goglidze  wrote:

> Hi Alex,
>
> Did you enable it on the other side?
>
> It's not because of traffic-shaping itself. It's has probably happened
> because you have frame-relay fragmentation enabled.
> DHCP should not be big packets, so it should not get fragmented, but
> probably .cnf file download was failing from the tftp server.
>
> If you did enable it on the other side too, can you post the config of the
> other router too then?
>
> Regards,
>
>
> On Fri, Apr 15, 2011 at 2:26 AM, Alex Goh  wrote:
>
>> Hi All,
>>
>> Was practicing workbook 2 lab 1, on the question regarding QOS between HQ
>> and BR1/BR2, I've enable
>> auto qos voip trust in BR1 router on the PVC interface. Once the router
>> was reboot, I notice that BR1 IP phones
>> wasn't able to get IP address from the DHCP server, which is CUCM Pub in
>> this case.
>>
>> I've tried removed the frame-relay traffic-shaping command on the BR1 FR
>> physical interface, immediately
>> the phones able to grab IP from DHCP server.
>>
>> Can anyone advise on this? My IOS version is 12.4 (24) T
>>
>>
>> Building configuration...
>>
>>
>> Current configuration : 3601 bytes
>> version 12.4
>> service timestamps debug datetime msec
>> service timestamps log datetime msec
>> no service password-encryption
>> !
>> hostname R2
>> !
>> boot-start-marker
>> boot-end-marker
>> !
>> logging message-counter syslog
>> enable secret 5 $1$BF1c$fGTsUdKaCoeiv8BjWdrw2/
>> !
>> no aaa new-model
>> network-clock-participate wic 0
>> !
>> !
>> !
>> dot11 syslog
>> ip source-route
>> !
>> !
>> ip cef
>> !
>> !
>> no ip domain lookup
>> no ipv6 cef
>> !
>> multilink bundle-name authenticated
>> !
>> !
>> !
>> !
>> isdn switch-type primary-ni
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> !
>> voice-card 0
>>  dsp services dspfarm
>> !
>> !
>> !
>> !
>> !
>> !
>> archive
>>  log config
>>   hidekeys
>> !
>> !
>> controller E1 0/0/0
>>  pri-group timeslots 1-3,16 service mgcp
>> !
>> controller E1 0/0/1
>>  channel-group 0 timeslots 1-31
>> !
>> !
>> class-map match-any AutoQoS-VoIP-RTP-Trust
>>  match ip dscp ef
>> class-map match-any AutoQoS-VoIP-Control-Trust
>>  match ip dscp cs3
>>  match ip dscp af31
>> !
>> !
>> policy-map AutoQoS-Policy-Trust
>>  class AutoQoS-VoIP-RTP-Trust
>> priority 47
>>compress header ip rtp
>>  class AutoQoS-VoIP-Control-Trust
>> bandwidth percent 5
>>  class class-default
>> fair-queue
>> !
>> !
>> !
>> !
>> !
>> interface Loopback0
>>  ip address 172.2.254.1 255.255.255.255
>> !
>> interface FastEthernet0/0
>>  no ip address
>>  duplex auto
>>  speed auto
>> !
>> interface FastEthernet0/1
>>  no ip address
>>  shutdown
>>  duplex auto
>>  speed auto
>> !
>> interface FastEthernet0/1/0
>>  switchport trunk native vlan 10
>>  switchport mode trunk
>>  switchport voice vlan 20
>> !
>> interface FastEthernet0/1/1
>>  switchport trunk native vlan 10
>>  switchport mode trunk
>>  switchport voice vlan 20
>> !
>> interface FastEthernet0/1/2
>> !
>> interface FastEthernet0/1/3
>> !
>> interface Serial0/0/0:15
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-ni
>>  isdn incoming-voice voice
>>  isdn bind-l3 ccm-manager
>>  isdn outgoing display-ie
>>  no cdp enable
>> !
>> interface Serial0/0/1:0
>>  no ip address
>>  encapsulation frame-relay
>> * frame-relay traffic-shaping*
>> !
>> interface Serial0/0/1:0.1 point-to-point
>>  description == FR To HQ
>>  bandwidth 384
>>  ip a

Re: [OSL | CCIE_Voice] Device mobility issue.

[Alex Goh]

Any kind soul able to help?

On Fri, Apr 15, 2011 at 5:17 PM, Alex Goh  wrote:

> Any kind soul able to help?
>
>
> On Thu, Apr 14, 2011 at 5:40 PM, Alex Goh  wrote:
>
>> Hi All,
>>
>> Understand that when an IP Phone was roaming and physical location of home
>> DP and roaming DP is different,
>> the roaming sensitive setting of the roaming DP will apply to the phone.
>> However, when i moved my BR1 phone
>> to HQ, the "View Current Device Mobility Settings" of the BR1 phone
>> showing the roaming DP is "Not Selected",
>> I believe it shouldn't be that way?
>>
>> Also I notice the Date Time Group on the roaming phone doesn't follow the
>> roaming DP, i thought DTG suppose
>> to be roaming sensitive setting and will apply to the roaming phone?
>>
>> Thanks
>>
>> Regards,
>> Alex
>>
>
>
___
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Re: [OSL | CCIE_Voice] Device mobility issue.

[Alex Goh]

Any kind soul able to help?

On Thu, Apr 14, 2011 at 5:40 PM, Alex Goh  wrote:

> Hi All,
>
> Understand that when an IP Phone was roaming and physical location of home
> DP and roaming DP is different,
> the roaming sensitive setting of the roaming DP will apply to the phone.
> However, when i moved my BR1 phone
> to HQ, the "View Current Device Mobility Settings" of the BR1 phone showing
> the roaming DP is "Not Selected",
> I believe it shouldn't be that way?
>
> Also I notice the Date Time Group on the roaming phone doesn't follow the
> roaming DP, i thought DTG suppose
> to be roaming sensitive setting and will apply to the roaming phone?
>
> Thanks
>
> Regards,
> Alex
>
___
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[OSL | CCIE_Voice] DHCP Issue with FRTS?

[Alex Goh]

Hi All,

Was practicing workbook 2 lab 1, on the question regarding QOS between HQ
and BR1/BR2, I've enable
auto qos voip trust in BR1 router on the PVC interface. Once the router was
reboot, I notice that BR1 IP phones
wasn't able to get IP address from the DHCP server, which is CUCM Pub in
this case.

I've tried removed the frame-relay traffic-shaping command on the BR1 FR
physical interface, immediately
the phones able to grab IP from DHCP server.

Can anyone advise on this? My IOS version is 12.4 (24) T


Building configuration...


Current configuration : 3601 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname R2
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
enable secret 5 $1$BF1c$fGTsUdKaCoeiv8BjWdrw2/
!
no aaa new-model
network-clock-participate wic 0
!
!
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
no ip domain lookup
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice-card 0
 dsp services dspfarm
!
!
!
!
!
!
archive
 log config
  hidekeys
!
!
controller E1 0/0/0
 pri-group timeslots 1-3,16 service mgcp
!
controller E1 0/0/1
 channel-group 0 timeslots 1-31
!
!
class-map match-any AutoQoS-VoIP-RTP-Trust
 match ip dscp ef
class-map match-any AutoQoS-VoIP-Control-Trust
 match ip dscp cs3
 match ip dscp af31
!
!
policy-map AutoQoS-Policy-Trust
 class AutoQoS-VoIP-RTP-Trust
priority 47
   compress header ip rtp
 class AutoQoS-VoIP-Control-Trust
bandwidth percent 5
 class class-default
fair-queue
!
!
!
!
!
interface Loopback0
 ip address 172.2.254.1 255.255.255.255
!
interface FastEthernet0/0
 no ip address
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface FastEthernet0/1/0
 switchport trunk native vlan 10
 switchport mode trunk
 switchport voice vlan 20
!
interface FastEthernet0/1/1
 switchport trunk native vlan 10
 switchport mode trunk
 switchport voice vlan 20
!
interface FastEthernet0/1/2
!
interface FastEthernet0/1/3
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 isdn outgoing display-ie
 no cdp enable
!
interface Serial0/0/1:0
 no ip address
 encapsulation frame-relay
* frame-relay traffic-shaping*
!
interface Serial0/0/1:0.1 point-to-point
 description == FR To HQ
 bandwidth 384
 ip address 10.10.11.2 255.255.255.252
 frame-relay interface-dlci 101
  class AutoQoS-FR-Se0/0/1:0-101
  auto qos voip trust
!
interface Vlan1
 no ip address
!
interface Vlan10
 ip address 172.2.12.1 255.255.255.0
!
interface Vlan20
 ip address 172.2.11.1 255.255.255.0
 ip helper-address 172.1.10.10
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.10.11.1
no ip http server
no ip http secure-server
!
!
!
!
map-class frame-relay AutoQoS-FR-Se0/0/1:0-101
 frame-relay cir 384000
 frame-relay bc 3840
 frame-relay be 0
 frame-relay mincir 384000
 frame-relay fragment 480
 service-policy output AutoQoS-Policy-Trust
!
!
!
!
!
!
control-plane
!
rmon event 3 log trap AutoQoS description "AutoQoS SNMP traps for Voice
Drops" owner AutoQoS
rmon alarm 3 cbQosCMDropBitRate.418.3168001 30 absolute rising-threshold
1 3 falling-threshold 0 owner AutoQoS
!
!
voice-port 0/0/0:15
!
voice-port 0/2/0
!
voice-port 0/2/1
!
ccm-manager switchback immediate
ccm-manager redundant-host 172.1.10.10
ccm-manager mgcp
!
mgcp
mgcp call-agent 172.1.10.20 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp fax t38 ecm
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
!
mgcp profile default
!
sccp local Loopback0
sccp ccm 172.1.10.20 identifier 1 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register BR1-XCODER
!
dspfarm profile 1 transcode
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 3
 associate application SCCP
!
!
!
!
!
line con 0
line aux 0
line vty 0 4
 password cisco
 login
 length 0
!
scheduler allocate 2 1000
end
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[OSL | CCIE_Voice] Device mobility issue.

[Alex Goh]

Hi All,

Understand that when an IP Phone was roaming and physical location of home
DP and roaming DP is different,
the roaming sensitive setting of the roaming DP will apply to the phone.
However, when i moved my BR1 phone
to HQ, the "View Current Device Mobility Settings" of the BR1 phone showing
the roaming DP is "Not Selected",
I believe it shouldn't be that way?

Also I notice the Date Time Group on the roaming phone doesn't follow the
roaming DP, i thought DTG suppose
to be roaming sensitive setting and will apply to the roaming phone?

Thanks

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

Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)

[Alex Goh]

Hi All,

Thanks very much for the reply. The issue is due to my mistake that
registering BR2 to wrong zone.

Now the CUCM Call to BR2 is working fine except the supplementary
service e.g hold, Moh doesn't
work, do I need MTP for this?

also, calling from BR2 Sip phone to CUCM is failling, phone ring, but
when answered, it dropped.
my Sip phone is using G729 codec, do I still need MTP on BR2 in this case?

Thanks

Regards,
Alex

On Sun, Apr 10, 2011 at 2:19 AM, Naoufal Kerboute
 wrote:
> Hi,
>
> You have to register the br2 with the UCME zone not the VIA zone.
>
> Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719
>
>  and replace it with
>
> h323-gateway voip id UCME ipaddr 172.1.254.1 1719
>
> Thanks
> Naoufal
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh
> Sent: Saturday, April 09, 2011 9:43 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
>
> Hi Guys,
>
> I'm trying to get the solutions for question 4.2 to work, but apparently I'm 
> missing something and hope someone can help.
> I've search thru the list but doesn't really found a solution work for my 
> case.
>
> The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 
> ring, but when i tried to answered, the call drop.
> I know it might be related to codec issue, but I've my HQ-RTR configured with 
> Xcoder which it is up and active but the call still failing. I also did have 
> the trunk in cucm "Wait for Far End
> H.245 Terminal Capability Set" unchecked.
>
> once things I notice is that, my call doesn't seems get re-originated on the 
> cube router to BR2 router, what I see during ringing state my "show 
> gatekeeper endpoint" show the call is directly from the CUCM to BR2 It is 
> only 2 call legs instead of 4 (see below).
>
> hm, what have I missed?
>
> Some Info:
> HQ Router (R1)
>
> interface Loopback0
>  ip address 172.1.254.1 255.255.255.255
>  h323-gateway voip interface
>  h323-gateway voip id VIA ipaddr 172.1.254.1 1719  h323-gateway voip h323-id 
> R1  h323-gateway voip bind srcaddr 172.1.254.1
>
> gatekeeper
>  zone local UCM 172.1.254.1
>  zone local UCME outvia VIA
>  zone local VIA
>  zone prefix UCME 3...
>  gw-type-prefix 1#* default-technology
>  no shutdown
>
> dial-peer voice 30 voip
>  destination-pattern 3...
>  session target ras
>  codec g711ulaw
> !
> dial-peer voice 31 voip
>  incoming called-number 3...
>
> Total number of active calls = 1.
>                         GATEKEEPER CALL INFO
>                         
> LocalCallID                        Age(secs)   BW
> 511-32797                          6           16(Kbps)
>  Endpt(s): Alias                 E.164Addr
>   src EP: gk_trunk_2            5001
>           CallSignalAddr  Port  RASSignalAddr   Port
>           172.1.10.20     38233 172.1.10.20     32795
>  Endpt(s): Alias                 E.164Addr
>   dst EP: R3                    3003
>           CallSignalAddr  Port  RASSignalAddr   Port
>           172.3.254.1     1720  172.3.254.1     49395
>
>                    GATEKEEPER ENDPOINT REGISTRATION
>                    
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type    Flags
> --- - --- - -             -
> 172.1.10.10     47142 172.1.10.10     32838 UCM               VOIP-GW
>    H323-ID: gk_trunk_1
>    Voice Capacity Max.=  Avail.=  Current.= 0
> 172.1.10.20     38233 172.1.10.20     32795 UCM               VOIP-GW
>    H323-ID: gk_trunk_2
>    Voice Capacity Max.=  Avail.=  Current.= 0
> 172.1.254.1     1720  172.1.254.2     56974 VIA               H323-GW
>    H323-ID: R1
>    Voice Capacity Max.=  Avail.=  Current.= 0
> 172.3.254.1     1720  172.3.254.1     49395 VIA               H323-GW
>    H323-ID: R3
>    Voice Capacity Max.=  Avail.=  Current.= 0 Total number of active 
> registrations = 4
>
> R1(config-if)#do sh gatek gw
> GATEWAY TYPE PREFIX TABLE
> =
> Prefix: 1#*    (Default gateway-technology)
>  Zone UCM master gateway list:
>    172.1.10.20:38233 gk_trunk_2
>    172.1.10.10:47142 gk_trunk_1
>  Zone VIA master gateway list:
>    172.3.254.1:1720 R3
>    172.1.254.2:1720 R1
>
> BR2 Router (R2)
>
> interface Loopback0
>  ip address 172.3.254.1 255.255.255.255
>  h323-gateway voip interface
>  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
>  h323-gateway voip h323-id R3
> h323-gateway voip tech-prefix 1#
> 

Re: [OSL | CCIE_Voice] System prompts in UCCX

[Alex]

Hi Miron
I use the prompts from the system folder directly without specifying the full 
path, but just the prompt name. The only thing need to be modified is to put S 
before P in the path ( SP[promptname]). If the prompt u r after is in sys 
folder subfolder, then u have to add this too (SP[ACD\promptname]). Also, make 
sure u using the correct sys folder corresponding to the language u configured, 
such as en_US. 
I always save my recorded prompts to the sys folders too.
Hope this helped.
Alex
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: Miron Kobelski 
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Sat, 9 Apr 2011 18:25:22 
To: 
Subject: [OSL | CCIE_Voice] System prompts in UCCX

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Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)

[Alex Goh]

Hi ShinGei,

Thanks for much for the reply, I guess I must be overlooked, the BR2
was registered to the wrong zone!
I'm going to try it on my lab tomorrow.

Again thanks!

Regards,
Alex

On Sun, Apr 10, 2011 at 2:18 AM, ShinGei Yong  wrote:
> Hi Alex@ncs,
>
> While observing your config,i noticed that you've 3 zone defined under GK,
> which are UCM,UCME& VIA.
>
> If i remember correctly,ur R3 which is ur CME site should registered to UCME
> instead of zone VIA right?
> Also, what is your region configuration on that pointed to GK?
>
> Thanks
> Shingei.
>
> On Sun, Apr 10, 2011 at 1:42 AM, Alex Goh  wrote:
>>
>> Hi Guys,
>>
>> I'm trying to get the solutions for question 4.2 to work, but
>> apparently I'm missing something and hope someone can help.
>> I've search thru the list but doesn't really found a solution work for my
>> case.
>>
>> The issue I've encounter are when HQ phone 5001 calling BR2 phone
>> 3003, 3003 ring, but when i tried to answered, the call drop.
>> I know it might be related to codec issue, but I've my HQ-RTR
>> configured with Xcoder which it is up and active but the call
>> still failing. I also did have the trunk in cucm "Wait for Far End
>> H.245 Terminal Capability Set" unchecked.
>>
>> once things I notice is that, my call doesn't seems get re-originated
>> on the cube router to BR2 router, what I see during ringing state
>> my "show gatekeeper endpoint" show the call is directly from the CUCM
>> to BR2 It is only 2 call legs instead of 4 (see below).
>>
>> hm, what have I missed?
>>
>> Some Info:
>> HQ Router (R1)
>>
>> interface Loopback0
>>  ip address 172.1.254.1 255.255.255.255
>>  h323-gateway voip interface
>>  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
>>  h323-gateway voip h323-id R1
>>  h323-gateway voip bind srcaddr 172.1.254.1
>>
>> gatekeeper
>>  zone local UCM 172.1.254.1
>>  zone local UCME outvia VIA
>>  zone local VIA
>>  zone prefix UCME 3...
>>  gw-type-prefix 1#* default-technology
>>  no shutdown
>>
>> dial-peer voice 30 voip
>>  destination-pattern 3...
>>  session target ras
>>  codec g711ulaw
>> !
>> dial-peer voice 31 voip
>>  incoming called-number 3...
>>
>> Total number of active calls = 1.
>>                         GATEKEEPER CALL INFO
>>                         
>> LocalCallID                        Age(secs)   BW
>> 511-32797                          6           16(Kbps)
>>  Endpt(s): Alias                 E.164Addr
>>   src EP: gk_trunk_2            5001
>>           CallSignalAddr  Port  RASSignalAddr   Port
>>           172.1.10.20     38233 172.1.10.20     32795
>>  Endpt(s): Alias                 E.164Addr
>>   dst EP: R3                    3003
>>           CallSignalAddr  Port  RASSignalAddr   Port
>>           172.3.254.1     1720  172.3.254.1     49395
>>
>>                    GATEKEEPER ENDPOINT REGISTRATION
>>                    
>> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type
>>  Flags
>> --- - --- - -         
>>  -
>> 172.1.10.10     47142 172.1.10.10     32838 UCM               VOIP-GW
>>    H323-ID: gk_trunk_1
>>    Voice Capacity Max.=  Avail.=  Current.= 0
>> 172.1.10.20     38233 172.1.10.20     32795 UCM               VOIP-GW
>>    H323-ID: gk_trunk_2
>>    Voice Capacity Max.=  Avail.=  Current.= 0
>> 172.1.254.1     1720  172.1.254.2     56974 VIA               H323-GW
>>    H323-ID: R1
>>    Voice Capacity Max.=  Avail.=  Current.= 0
>> 172.3.254.1     1720  172.3.254.1     49395 VIA               H323-GW
>>    H323-ID: R3
>>    Voice Capacity Max.=  Avail.=  Current.= 0
>> Total number of active registrations = 4
>>
>> R1(config-if)#do sh gatek gw
>> GATEWAY TYPE PREFIX TABLE
>> =
>> Prefix: 1#*    (Default gateway-technology)
>>  Zone UCM master gateway list:
>>    172.1.10.20:38233 gk_trunk_2
>>    172.1.10.10:47142 gk_trunk_1
>>  Zone VIA master gateway list:
>>    172.3.254.1:1720 R3
>>    172.1.254.2:1720 R1
>>
>> BR2 Router (R2)
>>
>> interface Loopback0
>>  ip address 172.3.254.1 255.255.255.255
>>  h323-gateway voip interface
>>  h323-gateway voip id VIA ipaddr 172.1.254.1 1719
>>  h323-gateway voip h323-id R3
>>  h323-gateway voip tech-prefix 1#
>>  h323-gateway voip bind srcaddr 172.3.254.1
>>
>> dial-peer voice 10 voip
>>  incoming called-number 3...
>>  dtmf-relay rtp-nte
>>  codec g711ulaw
>> !
>>
>> CUCM Trunk
>> the trunk was assign a separate DP with a region that using G729 when
>> calling HQ and BR2.
>>
>>
>>
>> Regards,
>> Alex
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>
>
___
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[OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)

[Alex Goh]

Hi Guys,

I'm trying to get the solutions for question 4.2 to work, but
apparently I'm missing something and hope someone can help.
I've search thru the list but doesn't really found a solution work for my case.

The issue I've encounter are when HQ phone 5001 calling BR2 phone
3003, 3003 ring, but when i tried to answered, the call drop.
I know it might be related to codec issue, but I've my HQ-RTR
configured with Xcoder which it is up and active but the call
still failing. I also did have the trunk in cucm "Wait for Far End
H.245 Terminal Capability Set" unchecked.

once things I notice is that, my call doesn't seems get re-originated
on the cube router to BR2 router, what I see during ringing state
my "show gatekeeper endpoint" show the call is directly from the CUCM
to BR2 It is only 2 call legs instead of 4 (see below).

hm, what have I missed?

Some Info:
HQ Router (R1)

interface Loopback0
 ip address 172.1.254.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id VIA ipaddr 172.1.254.1 1719
 h323-gateway voip h323-id R1
 h323-gateway voip bind srcaddr 172.1.254.1

gatekeeper
 zone local UCM 172.1.254.1
 zone local UCME outvia VIA
 zone local VIA
 zone prefix UCME 3...
 gw-type-prefix 1#* default-technology
 no shutdown

dial-peer voice 30 voip
 destination-pattern 3...
 session target ras
 codec g711ulaw
!
dial-peer voice 31 voip
 incoming called-number 3...

Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
511-32797  6   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: gk_trunk_25001
   CallSignalAddr  Port  RASSignalAddr   Port
   172.1.10.20 38233 172.1.10.20 32795
 Endpt(s): Alias E.164Addr
   dst EP: R33003
   CallSignalAddr  Port  RASSignalAddr   Port
   172.3.254.1 1720  172.3.254.1 49395

GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
172.1.10.10 47142 172.1.10.10 32838 UCM   VOIP-GW
H323-ID: gk_trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
172.1.10.20 38233 172.1.10.20 32795 UCM   VOIP-GW
H323-ID: gk_trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
172.1.254.1 1720  172.1.254.2 56974 VIA   H323-GW
H323-ID: R1
Voice Capacity Max.=  Avail.=  Current.= 0
172.3.254.1 1720  172.3.254.1 49395 VIA   H323-GW
H323-ID: R3
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 4

R1(config-if)#do sh gatek gw
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*(Default gateway-technology)
  Zone UCM master gateway list:
172.1.10.20:38233 gk_trunk_2
172.1.10.10:47142 gk_trunk_1
  Zone VIA master gateway list:
172.3.254.1:1720 R3
172.1.254.2:1720 R1

BR2 Router (R2)

interface Loopback0
 ip address 172.3.254.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id VIA ipaddr 172.1.254.1 1719
 h323-gateway voip h323-id R3
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 172.3.254.1

dial-peer voice 10 voip
 incoming called-number 3...
 dtmf-relay rtp-nte
 codec g711ulaw
!

CUCM Trunk
the trunk was assign a separate DP with a region that using G729 when
calling HQ and BR2.



Regards,
Alex
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Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization

[Alex]

Yep, the service looked good. Also, if u mess up the service u will not see it 
in the list on ur phone.
Another thing is that Service and Enterprise param were reset to default, so it 
is not coming from there too.
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: George Goglidze 
Date: Wed, 6 Apr 2011 06:34:10 
To: 
Cc: Shrini; ; 

Subject: Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization

hi Alex,

Did you check that the phone service service for received/missed calls is
enabled in phone services?
7965 models rely on that service to show that correctly.

Regards,

On Wed, Apr 6, 2011 at 12:30 AM, Alex  wrote:

> I've seen it the other way around' when 7965 behaves as 7960 and displays
> localized number in missed calls. Anybody have any ideas?
> Sent from my BlackBerry Wireless Handheld
>
> -Original Message-
> From: Shrini 
> Sender: ccie_voice-boun...@onlinestudylist.com
> Date: Tue, 05 Apr 2011 22:37:42
> To: George Goglidze
> Cc: 
> Subject: Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>

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Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization

[Alex]

I've seen it the other way around' when 7965 behaves as 7960 and displays 
localized number in missed calls. Anybody have any ideas?
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: Shrini 
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Tue, 05 Apr 2011 22:37:42 
To: George Goglidze
Cc: 
Subject: Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization

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Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK

[Alex]

Guys,
Most of the people on this list are trying to make way to their numbers without 
cheating.
I also failed a few times but never resorted to real labs.
However sometimes I
I feel that cisco is cheating on me.
Do u think R and S troubleshooting should be a part of Voice exam? Do u think 
that forcing u to use workaronds instead of normal tools available in real live 
adds any value to the certification? Don't u think that if cisco wants to test 
our troubleshoting skills they have to allocate some points and time for it but 
not brake something which u can possibly find only at the end of the exam when 
testing everything? 
I didn't have any problems configuring anything they asked me to. I also 
managed to resolve all the "tricks" they prepared for me (except of one for 
which I spent hours to research and recreate but still dono how did they brake 
it). However these hidden things kill ur time and as the result u don't have 
enough to verify everything and lose points because of typo etc.
So, don't blame the guy much, he is just trying to cheat on cheater.
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: Justin Brady 
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Mon, 4 Apr 2011 00:42:41 
To: George Goglidze; Ccie Voice
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK

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Re: [OSL | CCIE_Voice] Multicast MoH to CUBE

[Alex Goh]

Anyone can enlighten me?

On Sun, Mar 27, 2011 at 8:23 PM, Alex Goh  wrote:
> Hi All,
>
> When trying to practice some MoH lab today, I notice that whenever
> HQ/BR1 phone make a call
> to BR2 Phone via H323 gateway, which BR2 setup as CME
> (telephony-service), when HQ/BR1 phone press hold,
> the MoH will not work (BR2 phone hear silence). But when BR2 phone
> press Hold, HQ/BR1 phone can hear MoH.
>
> Same for when HQ/BR1 tried to send call out to PSTN via the H323
> gateway, the PSTN hear no MoH.
> BR2 Phone call PSTN the MoH is ok though.
>
> My HQ is set to send Unicast MoH, while BR1 is set to send Multicast
> MoH. MoH servers have different DP
> and Region to all site is g711.
>
> I did tried to turn the BR2 into a H323 gateway only (no CUBE) and
> configured the following commands:
> ccm-manager music-on-hold,
> moh music-on-hold.au
> multicast moh 239.1.1.5 port 16384
>
> Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via
> the H323 gateway, also when
> HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working
> fine for both direction.
>
> Is this normal that Multicast MoH or Unicast MoH is not supporting
> CUBE in this case?
> also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine?
>
> Regards,
> Alex
>
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Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incomingtraffic

[Alex]

Brian,
I was thinking about this similar way.
I guessed it could be achieved by putting the traffic in question into PQ, 
removing everything else from there and then policing it as required with 
service policy. 
However I suspect it could be not the way they will want as the solution does 
not look very clear.
Can anybody advise any other way to accomplish this?
Thanks
Alex 
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: Brian Mulgrew 
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Sun, 27 Mar 2011 21:55:17 
To: Randall Saborío Cubero
Cc: 
Subject: Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incoming
traffic

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Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incomingtraffic

[Alex]

Tnx Randall
I think this way u can only police the traffic but not guarantee the bandwidth.
Alex
Sent from my BlackBerry Wireless Handheld

-Original Message-
From: Randall Saborío Cubero 
Date: Sun, 27 Mar 2011 14:19:46 
To: Rogers Ochieng
Cc: ; 
Subject: Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incoming
 traffic

To me "for incoming traffic" means applying an inbound service policy on
a switch interface.

El dom, 27-03-2011 a las 19:08 +0300, Rogers Ochieng escribió:
> With CoS mapping to a queue and doing share/shape on the trunk port to
> router
> 
> On 27 March 2011 18:43,  wrote:
> Hey Experts
> 
> Anybody can clarify on this topic?
> How to GUARANTEE bandwidth for incoming traffic on 3750?
> 
> Thanks
> 
> - Forwarded message from a...@ipcomconsult.com -
>Date: Sat, 12 Mar 2011 02:12:11 -0700
>From: a...@ipcomconsult.com
> Reply-To: a...@ipcomconsult.com
>  Subject: [OSL | CCIE_Voice] 3750 bandwidth guarantee for
> incoming traffic
>  To: ccie_voice@onlinestudylist.com
> 
> 
> Hi guys
> Anybody can advise on how to GUARANTEE bandwidth for incoming
> traffic
> (let's say MGCP) on 3750?
> Policy-map as I understand can only police it but can not
> guarantee
> the bandwidth. Do you have to put it in Q2 removing any other
>     traffic
> from it?
> Any alternative solutions?
>  Tnx
> Alex
> 
> 
> ___
> For more information regarding industry leading CCIE Lab
> training, please visit www.ipexpert.com
> 
> 
> - End forwarded message -
> 
> 
> ___
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> training, please visit www.ipexpert.com
> 
> ___
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> visit www.ipexpert.com


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[OSL | CCIE_Voice] Multicast MoH to CUBE

[Alex Goh]

Hi All,

When trying to practice some MoH lab today, I notice that whenever
HQ/BR1 phone make a call
to BR2 Phone via H323 gateway, which BR2 setup as CME
(telephony-service), when HQ/BR1 phone press hold,
the MoH will not work (BR2 phone hear silence). But when BR2 phone
press Hold, HQ/BR1 phone can hear MoH.

Same for when HQ/BR1 tried to send call out to PSTN via the H323
gateway, the PSTN hear no MoH.
BR2 Phone call PSTN the MoH is ok though.

My HQ is set to send Unicast MoH, while BR1 is set to send Multicast
MoH. MoH servers have different DP
and Region to all site is g711.

I did tried to turn the BR2 into a H323 gateway only (no CUBE) and
configured the following commands:
ccm-manager music-on-hold,
moh music-on-hold.au
multicast moh 239.1.1.5 port 16384

Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via
the H323 gateway, also when
HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working
fine for both direction.

Is this normal that Multicast MoH or Unicast MoH is not supporting
CUBE in this case?
also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine?

Regards,
Alex
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Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus onoutgoing PRI calls

[Alex Goh]

hi i believe it will be more appropriate to add it on voice port if u
sending out to pstn,
except you are sending it to sip dial-peer, otherwise if h323 it wil
get strip off before
sending to session target. correct me if i'm wrong.

cheers,
Alex

On Thu, Mar 24, 2011 at 11:28 PM,   wrote:
> Does this have to go on the voice port or can it go on the dial peer too?
> -Original Message-
> From: adam compton 
> Sender: ccie_voice-boun...@onlinestudylist.com
> Date: Thu, 24 Mar 2011 08:24:41
> To: Bill Lake
> Cc: ; Adam Thompson
> Subject: Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus on
>  outgoing PRI calls
>
> ___
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> visit www.ipexpert.com
>
> ___
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] MVA Hairpin no Audio

[Alex Goh]

Hi,

Not sure is this help, but you might want to turn on MTP on ur h323 gateway?
i remember i read it from cisco notes saying it need to be turn on.

Regards,
Alex

On Fri, Mar 25, 2011 at 1:38 AM, study2b ccie  wrote:
> Hi experts,
> I had configured MVA using hairpin method.
> Everything worked and calls went out, but when I picked it up, there were no
> audio!
> Has anyone seen this problem before?
> Where should I start to troubleshoot?
> FYI, both of dial-peers voip are using no vad and G711ulaw.  I can see calls
> went out on mgcp trunk.
> Thank you,
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 113

[Alex Hannah]

Iwan,

You can use RTMT tool to view media resource stats and allocations.   
There is a counter that show multicast MOH sessions active per server.

HTH

Alex

Sent from my iPhone

On Mar 17, 2010, at 6:09 PM, ccie_voice-requ...@onlinestudylist.com  
wrote:

> 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
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: Lab 7 Question 3- DISA Dialling (Peter Slow)
>   2. Re: Lab 7 Question 3- DISA Dialling (Omotayo)
>   3. Re: Lab 7 Question 3- DISA Dialling (Radhesh Naik)
>   4. Re: MOH Multicast from Router flash (Amy Ryan)
>
>
> --
>
> Message: 1
> Date: Wed, 17 Mar 2010 16:20:10 -0400
> From: Peter Slow 
> Subject: Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
> To: Omotayo 
> Cc: OSL Group 
> Message-ID:
><53fc16d41003171320q95bc401m23e96c92c9f50...@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> do you mean device pool or MRGL? No CFB means CUCM couldnt select a
> CFB from the MRGL it was using for selection.
>
> -Peter
>
> On Wed, Mar 17, 2010 at 10:46 AM, Omotayo   
> wrote:
>> Hello,
>> i meant to say i put all the software conference brige in a device  
>> pool that
>> is not assigned to any user
>> thanks
>>
>> On Wed, Mar 17, 2010 at 9:09 AM, Omotayo   
>> wrote:
>>>
>>> Hello All,
>>> Worked on Lab 7 question 3 - DISA daling. i had two issues with this
>>> section while working on it;
>>>
>>> Q3.3. i configured conference resources on the br2 gateway and  
>>> applied to
>>> the BR2 Device pool. While the phone is "In Remote Use" . Also  
>>> applied
>>> CBarge on the BR2 phones.
>>>
>>> On pressing the red button, i get a reorder tone on both Br2 phone  
>>> and the
>>> Hq phone
>>>
>>> After this i put all the Hardware conference brige in a device  
>>> pool that
>>> is not assigned to any user, this time, when i press the red  
>>> button, i get
>>> "No Conference Bridge"
>>>
>>> Anyone with an idea what the issue is because i have the software
>>> conference bridge registered on the UCM
>>>
>>> Also i want to know why i can not apply 07976852817 as the remote
>>> destination profile with partial match set at the servie  
>>> parameter. i did
>>> and the br2 phone did not blink when calling HQ or BR1 phone.
>>> it was when i added the +447976852817 that it worked
>>>
>>> Thanks for the anticipated response
>>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training,  
>> please
>> visit www.ipexpert.com
>>
>>
>
>
> --
>
> Message: 2
> Date: Wed, 17 Mar 2010 21:21:10 +0100
> From: Omotayo 
> Subject: Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
> To: Peter Slow 
> Cc: OSL Group 
> Message-ID:
><3082f9d41003171321pe532fe8o1b313e9853bbe...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> yes i applied the mgl to the br2 device pool
>
> On Wed, Mar 17, 2010 at 9:20 PM, Peter Slow   
> wrote:
>
>> do you mean device pool or MRGL? No CFB means CUCM couldnt select a
>> CFB from the MRGL it was using for selection.
>>
>> -Peter
>>
>> On Wed, Mar 17, 2010 at 10:46 AM, Omotayo   
>> wrote:
>>> Hello,
>>> i meant to say i put all the software conference brige in a device  
>>> pool
>> that
>>> is not assigned to any user
>>> thanks
>>>
>>> On Wed, Mar 17, 2010 at 9:09 AM, Omotayo   
>>> wrote:
>>>>
>>>> Hello All,
>>>> Worked on Lab 7 question 3 - DISA daling. i had two issues with  
>>>> this
>>>> section while working on it;
>>>>
>>>> Q3.3. i configured conference resources on the br2 gateway and  
>>>> applied
>> to
>>>> the BR2 Device pool.

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 47, Issue 117

[Alex Hannah]

Stephen,

If you want to see best practices for the global dialplan, I would  
pull up the CUCM SRND.  Under the "Call Routing" section for the 7.x  
and 7.1 srnd there is a new features section where they have a graph  
of the global dialplan where it breaks down partitions, Css, Xlations,  
and Xforms.  Also, see if you can get the advanced dialplan preso from  
Networkers 2009.

Having played with global dialing and taking the voice lab recently  
here is the basic layout that I use which works very well.

Create a global partition which has all E.164 numbers in it.  Also, if  
you have to do any TEHO you can expand your patterns to include the  
pattern with area code and a seperate RL other than Stand Local.   
Example,  \+.! Points to SLRL and \+1212.! which points to a NYC RL  
containing NYC RG first then SLRL second.

Every site will have a device partition which xlates user dialable  
numbers into their global representations.  Set ANI here!  Also I  
create a US partition which houses all Intl and LD patterns that get  
xlated to their global representations.

Each phones device css has the site specific pt first, then US pt,  
then Global pt last.

This will route all global numbers to the right GWY, but you need to  
use called party xforms on the GWY or DP where the GWY is in order to  
step the global number down to PSTN requirements.  This will let you  
change DNIS and called type/plan.

Hope that helps...

Alex

Sent from my iPhone

On Jan 24, 2010, at 4:11 PM, ccie_voice-requ...@onlinestudylist.com  
wrote:

> 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
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: Proper Way to Build Your Dial Plan (Stephen Greszczyszyn)
>   2. Re: Lab 12A - UCCX custom script (Roger K?llberg)
>   3. fair-queue when configuring FRF.12 (sean hurricane)
>   4. SNR and call apperance (LAB 4 Q 3.1) (sean hurricane)
>   5. CUPS Registration issue (sean hurricane)
>
>
> --
>
> Message: 1
> Date: Sun, 24 Jan 2010 19:26:42 +
> From: Stephen Greszczyszyn 
> Subject: Re: [OSL | CCIE_Voice] Proper Way to Build Your Dial Plan
> To: ccie_voice@onlinestudylist.com
> Message-ID:
><71601cd61001241126j3d4727d3nc35c060731013...@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> David,
>
> Thanks for bringing up this topic, as I have been struggling with the
> same question while working through Vol1 Lab5.  I have seen different
> posts or videos about using one route pattern which points to the
> Standard Local Route Group, and then using various translation
> patterns.  Should the single E164 route pattern be \+.! and we strip
> the "+" and route only digits, or should we set the route pattern to
> be \+! and route the number with the prefixed "+"?  Or does it really
> matter which way we do it?
>
> I went through the routing lab fairly well, and got most of the
> results even though I did things quite differently than in the Proctor
> Guide.  I set up the single route group and did most of my
> manipulations using a combination of translation patterns,
> transformation patterns, or route lists.  I'm just not sure that I'm
> setting things up in a way that is non-scalable in real-life networks
> or in way that is hazardous to passing the exam :)
>
> Maybe Vik or Otto can give us some guidance on what is the "best" way
> to organize the dialplan?
>
>
> --
>
> Message: 2
> Date: Sun, 24 Jan 2010 20:27:47 +0100
> From: Roger K?llberg 
> Subject: Re: [OSL | CCIE_Voice] Lab 12A - UCCX custom script
> To: "ccie_voice@onlinestudylist.com" 
> Message-ID:
><79fa99add19eda4c9880d26d736e50ef2cf7fa4...@ex2-sth.domain.root>
> Content-Type: text/plain; charset="windows-1252"
>
> No one that has any thought about this?
>
> Roger K?llberg
> Unified Communication Consultant
> Cygate AB
>
>
> From: Roger K?llberg [mailto:roger.kallb...@cygate.se]
> Sent: den 22 januari 2010 09:24
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Lab 12A - UCCX custom script
>
> Hi guys and girls, (if there are any around, I mean except for  
> Amy :))!
>
> I need to get s

[OSL | CCIE_Voice] CCIE Voice 25853!!!!!!!!!!!!!!!!!!

[Alex Hannah]

Guys,

I recieved the news about an 20 minutes ago, passed the lab which I took on
Friday in RTP!  I cannot thank the people on this list enough (
Johnathan Charles, Amir, Otto, Vic, Mark, Amy, etc).  Also a big thanks to
IPExpert who's material helped me along throughout the years.

I will put out a more detailed debrief/how to study later but I am still
in shock and shaking :)

Alex Hannah
CCIE Voice 25853
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 47, Issue 70

[Alex Hannah]

Matthew,

Depends on what is connected to the other end of the trunk and what  
that devices capabilities are.  Do a show cdp neigh details look at  
the remote sides info,  also look at the routers fa or gig port  
config.  I personally leave it to auto negot, but I know a lot of  
peeps who hard code it.  Do a term mon on the switch and see if you  
are getting speed/duplex mismatches.

Sean,

Your config outbound in Cucm is where you need to look,  goto  
serviceabilty page and run DNA.  Run it from the phones tab NOT the  
analizer and fill in the calling/called num and the CSS.  On the  
gateway you can do a sh isdn stat, and debug mgcp packet.  Make sure  
your discarding predot and your called party xforms are setting the  
type/plan correctly.

HTH

Alex

Sent from my iPhone

On Jan 17, 2010, at 12:00 PM, ccie_voice-requ...@onlinestudylist.com  
wrote:

> Send CCIE_Voice mailing list submissions to
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>
> 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
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Trunk Port Configuration (Berry, Matthew J.)
>   2. MGCP issue (sean hurricane)
>
>
> --
>
> Message: 1
> Date: Sun, 17 Jan 2010 09:22:00 -0600
> From: "Berry, Matthew J." 
> Subject: [OSL | CCIE_Voice] Trunk Port Configuration
> To: "ccie_voice@onlinestudylist.com" 
> Message-ID:
><0d69df41cdb9bf4bb30c481d414a8be70ab9554...@usepx2pmxmbx04.corp.kroll.com 
> >
>
> Content-Type: text/plain; charset="us-ascii"
>
> In workbook 1, lab 1, we are told to configure a trunk port this way:
>
> Standard Catalyst 3750 Configuration for Trunk Port
>
> Vlan 10
>
> Name DATA
>
> State active
>
> Interface FastEthernet 1/0/2
>
> Switchport trunk encapsulation dot1q
>
> Switchport mode trunk
>
> Switchport trunk native vlan 10
>
> Speed 100
>
> Duplex full
>
> Is there a specific reason that you manually set the speed and  
> duplex instead of letting it negotiate automatically?
>
> Thanks,
>
> Matthew Berry, Sr. Unified Communications Engineer, CCVP
> Kroll Ontrack  |  9023 Columbine Road, Eden Prairie, MN 55347
> 952 516 3748  |  Fax 952 516 3646  |  Mobile 952 221 2814|  
> mjbe...@krollontrack.com 
> <mailto:agutz...@krollontrack.com>
> www.krollontrack.com<http://www.krollontrack.com/>
>
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>
> --
>
> Message: 2
> Date: Sun, 17 Jan 2010 11:40:14 -0500
> From: sean hurricane 
> Subject: [OSL | CCIE_Voice] MGCP issue
> To: ccie_voice@onlinestudylist.com
> Message-ID:
><29604bef1001170840u6a8a519v19d3c1194a2bf...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I am experiencing a problem with MGCP where i am not able to make  
> outbound
> pstn calls, i will just get a re-order tone
> and i dont see the call on the gateway... but when i call from the  
> PSTN
> inbound i can see the call using debug isdn 931 but i does not ring  
> UCCM
> .the gateway is registered in ccm and multi frame established..  
> this
> leads me to believe the problem is in UCCM... i have checked all the
> usuals.significant digit, inbound css and funny thing is my HQ  
> and BR2
> router are registered to the same UCCM and working
>
>
> *Router info*
> **
> *2811*
> *ios: 2800nm-adventerprisek9_ivs-mz.124-22.T2.bin*
> **
> **
> *any thoughts? *
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>
> --
>
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>
> End of CCIE_Voice Digest, Vol 47, Issue 70
> **
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 47, Issue 69

[Alex Hannah]

Kill Mill,

Look at the command "auto qos voip cisco-phone". It does everything  
you just did for you regarding SCCP traffic which I think is what you  
are getting at " wink wink".  All you need to do is change the mls qos  
dscp map from 0 to 8.   Also pull up the 3750 configuration guide for  
your version of ios.  As it walks you through all pieces of qos on the  
switch.

HTH

Alex

Sent from my iPhone

On Jan 17, 2010, at 10:15 AM, ccie_voice-requ...@onlinestudylist.com  
wrote:

> 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
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. EZVPN Setup (Arun Kumar)
>   2. Police SCCP traffic on the Switch (kill mill)
>   3. Phones Losing Connection with Rack (Berry, Matthew J.)
>   4. Vol1 lab 9.4 (Ehab Salem)
>
>
> --
>
> Message: 1
> Date: Sat, 16 Jan 2010 23:38:26 +0530
> From: Arun Kumar 
> Subject: [OSL | CCIE_Voice] EZVPN Setup
> To: ccie_voice@onlinestudylist.com
> Cc: Vik Malhi , Michael Ciarfello
>
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Dear All,
>
> tried all possible ways but still not able to get this working  
> talking to
> support also but someone have any suggestion will be appreciated,  
> here is my
> config:
>
> I'm using 2821 router and all my phones are connected to 3560 no  
> vlans.
>
> Router#show running-config
> Building configuration...
>
> Current configuration : 3515 bytes
> !
> version 12.4
> service timestamps debug datetime
> service timestamps log datetime
> service password-encryption
> !
> hostname Router
> !
> boot-start-marker
> boot-end-marker
> !
> ! card type command needed for slot/vwic-slot 0/1
> ! card type command needed for slot/vwic-slot 0/3
> logging message-counter syslog
> logging buffered 512000 informational
> enable secret 5 $1$BbSU$D1GeD44ZhXIIS4wYTDwkF0
> !
> no aaa new-model
> !
> dot11 syslog
> ip source-route
> !
> !
> ip cef
> ip dhcp excluded-address 192.168.7.1 192.168.7.10
> !
> ip dhcp pool DHCP-Pool
>   import all
>   network 192.168.7.0 255.255.255.0
>   option 150 ip 10.10.210.10
>   default-router 192.168.7.1
>   dns-server 209.124.41.100
>   domain-name proctorlabs.com
>   lease 8
> !
> !
> ip inspect name CBAC-FW tcp timeout 3600
> ip inspect name CBAC-FW udp timeout 3600
> ip inspect name CBAC-FW http java-list 1 timeout 3600
> ip inspect name CBAC-FW https timeout 3600
> ip inspect name CBAC-FW icmp
> ip inspect name CBAC-FW ddns-v3
> ip inspect name CBAC-FW smtp
> ip inspect name CBAC-FW pop3
> ip inspect name CBAC-FW pop3s
> ip inspect name CBAC-FW imap
> ip inspect name CBAC-FW ftps
> ip inspect name CBAC-FW ntp
> ip inspect name CBAC-FW ftp timeout 3600
> no ipv6 cef
> !
> multilink bundle-name authenticated
> !
> !
> voice-card 0
> !
> !
> archive
> log config
>  hidekeys
> !
> !
> crypto isakmp policy 10
> encr 3des
> authentication pre-share
> group 2
> !
> !
> crypto ipsec client ezvpn IPx-Voice-vRack
> connect auto / manual also
> group vpodgroup key proctorvoice
> mode client / network-extension also
> peer 74.126.20.247
> xauth userid mode interactive
> !
> !
> !
> interface GigabitEthernet0/0
> description insdie interface
> ip address 192.168.7.1 255.255.255.0
> ip nat inside
> ip virtual-reassembly
> duplex auto
> speed auto
> crypto ipsec client ezvpn IPx-Voice-vRack inside
> !
> interface GigabitEthernet0/1
> description (Outside Public Interface)
> ip address dhcp
> ip access-group FW-IN in
> no ip unreachables
> ip mtu 1300
> ip nat outside
> ip inspect CBAC-FW out
> ip virtual-reassembly
> duplex auto
> speed auto
> no cdp enable
> crypto ipsec client ezvpn IPx-Voice-vRack
> !
> ip forward-protocol nd
> ip route 0.0.0.0 0.0.0.0 dhcp
> ip http server
> no ip http secure-server
> !
> !
> ip nat inside source list 101 interface GigabitEthernet0/1 overload
> !
> ip access-list extended FW-IN
> permit udp any any eq bootpc
> deny   ip 10.0.0.0 0.255.255.255 any log
> deny   ip 172.16.0.0 0.15.

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 47, Issue 40

[Alex Hannah]

Jeff,

What's even better  Check out their extremely "professional"  
website.  There is no phone number, just a "gmail" account and you  
have to pay for their product using paypal, western union, or get this  
a wire transfer!!!

Now I did a whois and the site is hosted/registered in brussels, which  
coincidently is the latest test site for the lab!  So does that mean  
this is a disgruntled Cisco employee leaking the test,  someone taking/ 
memorizing the test, or a complete f'n joke???

Also they are pretty open in listing questions on their site, pretty  
close to nda if you ask me,

I am tempted to send it to Cisco.  Is anyone legitimately using the  
$3k worth of products?

Sent from my iPhone

On Jan 11, 2010, at 6:46 AM, ccie_voice-requ...@onlinestudylist.com  
wrote:

> 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
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: Attempted but it's bad luck again,I was veryclose.
>  (Jeff Garvas)
>   2. RTP availability (Paul Dardinski)
>   3. LAB 1A CME phone not registering (Randall Crumm)
>   4. How to insatll Unity 7.x on the PC platform (bijan arda)
>   5. Re: How to insatll Unity 7.x on the PC platform (Hough, Earl)
>
>
> --
>
> Message: 1
> Date: Sun, 10 Jan 2010 20:29:17 -0500
> From: Jeff Garvas 
> Subject: Re: [OSL | CCIE_Voice] Attempted but it's bad luck  
> again,I
>was veryclose.
> To: ccie_voice@onlinestudylist.com
> Message-ID:
><3f94dfab1001101729o14fe3651ibab2d96f04276...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Sun, Dec 27, 2009 at 5:29 AM, cciemeanstome cciemeanstome <
> cciemeanst...@gmail.com> wrote:
>
>> Hi guys,
>>
>> Sry for comming late attempted lab last week and i was very very  
>> close.
>>
>> Lab was exactly same as ccie-voice-labs"."com,
>>
>
> Am I the only one who finds it funny that someone who continually  
> fails the
> lab wants you to buy his product?
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100110/95a96e46/attachment-0001.htm
>
> --
>
> Message: 2
> Date: Sun, 10 Jan 2010 22:02:59 -0500
> From: "Paul Dardinski" 
> Subject: [OSL | CCIE_Voice] RTP availability
> To: 
> Message-ID:
>
> Content-Type: text/plain; charset="us-ascii"
>
> Anyone heard any word if the "closed til July" RTP availability is a
> temporary phenomena or is it pretty much set in stone that there won't
> be a date in RTP til July 1? I'm asking as my written expires 5/12/10
> and I had planned to attempt in April timeframe, but SJC would mean  
> much
> more travel time...which I can't take at this point.
>
>
>
> Paul (RS/Sec #16842)
>
>
>
>
>
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100110/8d493585/attachment-0001.htm
>
> --
>
> Message: 3
> Date: Sun, 10 Jan 2010 19:59:24 -0800
> From: Randall Crumm 
> Subject: [OSL | CCIE_Voice] LAB 1A CME phone not registering
> To: "ccie_voice@onlinestudylist.com" 
> Message-ID:
><9473270a65ca67458d287f3da3c9f37d0ed5314...@exch-cms.hlit.local>
> Content-Type: text/plain; charset="us-ascii"
>
> HI,
> I am having a hard time getting the 1 phone on CME to register.
>
> I am attaching my config
>
> Thanks
> Randall
>
>
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100110/1352dac2/attachment-0001.htm
> -- next part --
> A non-text attachment was scrubbed...
> Name: lab 1A BR2-router.log
> Type: application/octet-stream
> Size: 4154 bytes
> Desc: lab 1A BR2-router.log
> Url : 
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100110/1352dac2/attachment-0001.obj
>
> --
>
> Message: 4
> Date: Sun, 10 Jan 2010 23:31:52 -0800
> From: bijan arda 
> Subject: [OSL | CCIE_Voice] How to insatll Unity 7.x on the PC
>platform
> To: ccie_voice@onlinestudylist.com
> Message-ID:
><300d709e1001102331m287ae286xf58f510a9d0df...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi ALL
>
> How do i install unity 7.0 on the PC platform with 2 gig of RAM with  
> windows
> 2003 installed.??
>
> Can anyone send a procedure link?
>
> Thanks
> -- 
> Bijan Ardalan
> Senior Network Eng
> --

[OSL | CCIE_Voice] CUCM 7.X Auto Registration issues...

[Alex Hannah]

Guys,

I have a CUCM 7.x cluster ( 7.1(2) ) with a Pub and a Sub with the Sub being
the primary server for phones to register to via the CM Group, and auto
registration is turned on for the Sub.  DHCP is running on the Pub, the
phones get an IP Address, and download the firmware fine, when they attempt
to register to the CUCM Server something very strange occurs.  It seems like
the phone registers for a split second and then I get the "registration
rejected" message.  The phone then attempts to register again and for a
split second I see it register, then I recieve the "registration rejected"
message.

If I click on the Settings > Status > Status Messages, I see the following.

4:19:58a SEPXXX.cnf.xml
4:19:58a No CTL Installed
4:19:58a No IPv4 TFTP Server
4:19:58a File Not Found: CTLFile.tlv
4:19:58a No IPv4 DNS Server

Under the DHCP Scope for each subnet that I have IP Phones in ( HQ in this
case ), I have option 150 pointing to the PUB at 142.100.64.11.

I have reset the TFTP Server and I have since stopped the CUCM Service on
the Pub to force the phones to register via the Sub.

I thought to check the DB Replication to see if that was hosed up somehow,
but it looks okay I think:

admin:utils dbre
admin:utils dbreplication status

  utils dbreplication status 
Replication status check is now running in background.
Use command 'utils dbreplication runtimestate' to check its progress
The final output will be in file
cm/trace/dbl/sdi/ReplicationStatus.2010_01_03_04_23_18.out
Please use "file view activelog
cm/trace/dbl/sdi/ReplicationStatus.2010_01_03_04_23_18.out " command to see
the output
admin:file view activelog
cm/trace/dbl/sdi/ReplicationStatus.2010_01_03_04_23_18.out
SERVER ID STATESTATUS QUEUE  CONNECTION CHANGED
---
g_esx_pub1_ccm7_1_2_2_22 Active   Connected   0 Jan  2 21:22:57
g_esx_sub1_ccm7_1_2_2_24 Active   Local   0
end of the file reached
options: q=quit, n=next, p=prev, b=begin, e=end (lines 1 - 4 of 4) :


Any thoughts?

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

[OSL | CCIE_Voice] CUCM 7.1(2) - Apply Config Button?

[Alex Hannah]

Guys,

I have a very stupid question regarding CUCM 7.1(X) and the new "Apply
Config" button.  In regards to phones... if you configure the device then
the line on a phone, what the heck is the difference in "resetting" vs
"apply config"?  It looks to me after reading the description of the apply
config button that is performs more a restart than a reset.  Under what
circumstances would you use this in regards to a phone?

Thanks,

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

Re: [OSL | CCIE_Voice] 3750 QoS Question

[Alex Hannah]

Michael,

My understanding was older CUCM servers ( 4.x and early 5.x ) sent
signalling out at AF31, also I thought I remembered something about CIPC not
sending traffic out with right markings.  I was trying to do a "catch all"
to match any type of signaling be it either CS3 or AF31.

And the police statement I have verified on my 2811 running 12.4(22) T2 (
Same as v3 lab last month ).  So I believe this to be correct.  What exactly
did you mean by checking it to meet "ONLY" my requirements?  The exceed
action would remark traffic above 32k down to 8k correct?

Thanks again,

Alex

2009/11/11 Michael Ciarfello 

>  That's looking better.  Check your policed-dscp line to ONLY meet your
> requirements.
>
> Check the command reference and 3750 Switch COnfiguration guide - QoS
> chapter on that police command. I haven't looked at that or remember if it's
> correct.
>
> Pay attention to what Farkas said.  Look at other documents to find the
> source of that.  Maybe the document I mentioned above on what he is saying
> is in there.
>
> Why CS3 and AF31?  If you have a home lab or a partial home lab, use a
> sniffer and sniff around.  Let us know what you find.
>  --
>  *From:* ccie_voice-boun...@onlinestudylist.com [
> ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Hannah [
> alex.han...@gmail.com]
> *Sent:* Wednesday, November 11, 2009 6:56 PM
> *To:* Farkas Péter
> *Cc:* ccie_voice@onlinestudylist.com
>
> *Subject:* Re: [OSL | CCIE_Voice] 3750 QoS Question
>
>Michael and Farkas,
>
> Okay, I have thought about what you mentioned.  Here is my revised
> approach.  Let me know what you think about this way:
>
> !
> mls qos map policed-dscp  0 24 to 8
> mls qos map cos-dscp 0 8 16 24 32 46 48 56
> mls qos
> !
> !
> class-map match-any SCCP-Traffic
>   match ip dscp cs3  af31
> !
> !
> policy-map POLICE-MAP
>   class SCCP-Traffic
> police 32 8000 exceed-action policed-dscp-transmit
>set dscp cs3
> !
> !
> interface FastEthernet0/6
>   service-policy input POLICE-MAP
> !
>
> What is the signifigance of matching both ip dscp cs3  af31?  Since I have
> match-any will it match on both?  New CUCM 7.x servers should send SCCP out
> at cs3 correct?
>
> Thanks,
>
> Alex
>
>
> 2009/11/11 "Farkas Péter" 
>
>> AutoQoS cannot be configured until service-policy is attached to the
>> interface so you cannot use it for correction. Also, AutoQos does not work
>> on Eth.
>>
>> - Original Message -
>> From: Michael Ciarfello 
>> Date: Wednesday, November 11, 2009 8:56 pm
>> Subject: Re: [OSL | CCIE_Voice] 3750 QoS Question
>> To: Alex Hannah , "ccie_voice@onlinestudylist.com"
>> 
>>
>>
>> > Here are some hints for you to research:
>> >
>> >  I believe there is an error in one of the class-maps.  See if you can
>> find it or agree.
>> >
>> >  I believe you have too much extra stuff configured, let’s eliminate the
>> unneeded stuff.
>> >
>> >  How about use match IP protocol instead of access-lists?
>> >
>> >  Are you sure your access-list is correct for the inbound / outbound
>> traffic you have?
>> >
>> >  I think the data vlan people are going to be pissed and complain about
>> slowness.  I know it’s
>> > a lab.  I believe you can get the entire config down to a much simplier
>> 10-15 lines instead of
>> > all the stuff you have.
>> >
>> >  From: ccie_voice-boun...@onlinestudylist.com [ On Behalf Of Alex
>> Hannah
>>  >  Sent: Wednesday, November 11, 2009 2:41 PM
>> >  To: ccie_voice@onlinestudylist.com
>> >  Subject: [OSL | CCIE_Voice] 3750 QoS Question
>> >
>> >  Hello everyone.
>> >
>> >  I am attempting to create the following QoS policy on a 3750  port with
>> an IP Phone plugged in
>> > behind it.
>> >
>> >  The policy will police signalling ( SCCP ) 32k down to 8k and remark to
>> DSCP 8.  I have read
>> > through most of the SRND guide for the 3750, the model I am following is
>> the:
>> >
>> >  2970/3560/3750–Conditionally-Trusted IP Phone + PC + Scavenger (Basic)
>> Model Configuration on
>> > page 105 of the 3.3 QoS SRND.
>> >
>> >  Can anyone validate my work below and let me know if you think this
>> meets those requirements?
>> > Also, in this scenerio, Auto Qos would not need to be applied over top
>> of it correct?
>> >
>> >  mls qos map cos-dscp 0 8 16 24 32 46 48 56
>> &g