Re: [OSL | CCIE_Voice] LICENSE FROM FTP on 10.10.210.5
A question mark in the correct place will shine the light on you. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Saturday, November 21, 2009 1:41 PM To: OSL Group Subject: [OSL | CCIE_Voice] LICENSE FROM FTP on 10.10.210.5 Hello, i was trying to install the license to use cue with ccm but got the following error. Sdid i miss the syntax? CUE# $ clean url ftp://10.10.210.5/cue-vm-license_12mbx_ccm_7.0.1.pkg WARNING:: This command will install the necessary software to WARNING:: complete a clean install. It is recommended that a backup be done WARNING:: before installing software. Would you like to continue? [n]y Downloading ftp cue-vm-license_12mbx_ccm_7.0.1.pkg Error: Download error Can not download cue-vm-license_12mbx_ccm_7.0.1.pkg error code 530 : error type 'Access denied: 530' CUE# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Inbound PRI Calls Display ID Unavailable
Check with debug isdn q931. No calling number field, no caller ID. Then you can send them the output when they say must be your equipment. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder [narinder.ku...@uxcg.com.au] Sent: Wednesday, November 18, 2009 3:36 PM To: Berry, Matthew J.; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Inbound PRI Calls Display ID Unavailable Most likely Carrier is not sending the CID, check with the carrier. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry, Matthew J. Sent: Thursday, 19 November 2009 7:24 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Inbound PRI Calls Display ID Unavailable All – I am working on a CUCM 6.2 cluster where all inbound calls display a “ID Unavailable” on the ringing phone. We are running PRIs on MGCP gateways. PSTN -- Internal Extension = Calling number, but no CID (“ID Unavailable”) I have looked through the CUCM 6.x Admin Guide, but could not figure out a place to make a change. Could it be that our carrier is not sending CID? Can someone point me in the right direction so I can figure this out? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 562 2175 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection User Import
Maybe a sync? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Thursday, November 12, 2009 2:50 AM To: 'OSL Group' Subject: Re: [OSL | CCIE_Voice] Unity Connection User Import I am still waiting on response, Anyone got idea ?? Thanks From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Wednesday, 11 November 2009 5:25 PM To: 'OSL Group' Subject: [OSL | CCIE_Voice] Unity Connection User Import All, Integrated Unity connection with UCM and imported the users I wanted to import everything was working as I wanted. I end up rebuilding my unity connection box while the UCM stay as it is, integrated the Unity connection part again with the UCM. When I try to import the users from Unity Connection, I am only seeing the user which I did NOT import last time. Am I missing something ? can't find the reason why ? Any help much appreciated Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] terminal emulator used in lab.
Didn't look at ANY version stuff. Didn't really even notice it was SecurCRT or not. I don't use SecurCRT in every-day workings, but it caused me no troubles, headaches, slowdowns, etc. I clicked on the icon(s), the thing(s) came up and away I went. I guess I am trying to be subtle and say it doesn't matter. Don't sweat the small stuff. I know you are just looking for that comfort blanket, but trust me, it doesn't matter. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brian Valentine Sent: Thursday, November 12, 2009 11:14 AM To: OSL Group Subject: [OSL | CCIE_Voice] terminal emulator used in lab. All, I believe the CCIE lab uses Secure CRT for access to routers and switches and I believe this is public knowledge. Does any one know if the version of Secure CRT used is public knowledge? Can anyone with recent lab experience tell me the version of Secure CRT currently being used in the lab? Let me be clear. I'm not looking for anyone to break NDA. I'm asking if this is public knowledge and, if so, what is the current version. I would like to get a couple days of experience with the tool to make sure that I can work it efficiently in my upcoming lab exam. I do not want to rely on features that will not be available to me during my actual lab attempt. Thanks for any help you can give. Brian ___ 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
Re: [OSL | CCIE_Voice] AXL app user
I just can't stay away. Maybe it's because I don't want to finish my scope of work. It's boring. Only problem is in real life if someone changes or has to change the appadmin as you call it user. You then break presence. Other than that I would say do what the lab requirements say. Maybe they want to test you on creating the correct user. Maybe they want to test you on creating a new group that only has Std AXL role. Etc, etc, etc. Maybe the appadmin user's password expires at 5:11pm and you missed the hint in the test to deal with making sure it doesn't, etc. Short answer: YES. Long answer: Study more (and don't do it for production.) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hawkins Jason L NGA-ES USA CTR Sent: Thursday, November 12, 2009 12:12 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] AXL app user When integrating CUPS and CUC with CUCM via AXL is there any reason I shouldn't just use the default appadmin account that already has the Standard AXL API Access user role? Does using the appadmin account for this purpose break something else? I would like to just the appadmin account for the AXL account for speed reasons in the lab instead of going through the process of creating another application user only for AXL access between servers. Does anyone know of any problems with using the appadmin account like this? Jason ___ 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
Re: [OSL | CCIE_Voice] MOH spoofing at remote branch
Lol. I'm still here. I'm trying to get an IP phone to register to my Commodore 64. I might need help. Lol From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi Sent: Wednesday, November 11, 2009 12:13 AM To: Aamir Panjwani; Kumar, Narinder; Robert McGhee; OSL Group Subject: Re: [OSL | CCIE_Voice] MOH spoofing at remote branch Aamir- good to see you sticking around! It's like the KGB used to say- we will never let you leave (this list)! I'm sure Michael, Jonathan, Daniel, Otto et al are still addicted too. My two cents: no mgcp / mgcp. Also one more: the router is not multicasting and you do not need pim/multicast-routing enabled on the router since the router is the MOH server and we are flooding the multicast packets onto the subnets defined in the multicast moh route statement. -- Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Aamir Panjwani aamir.panjw...@ivision.com.au Date: Wed, 11 Nov 2009 15:50:21 +1100 To: Kumar, Narinder narinder.ku...@uxcg.com.au, Robert McGhee bobwmcg...@verizon.net, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH spoofing at remote branch In addition to what Narinder suggested, by default MOH file name is music-on-hold.au NOT music_on_hold.au unless you manually renamed it. If you are playing from flash there is no need to enable multicast/pim dense mode. Turn on debug ccm-manager music-on-hold events and debug ip igmp then make a test call If you hear tone on hold as opposed to dead silence, that usually means UCM config problem. Double check you mrg/mrgl/moh server/regions etc. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Wednesday, 11 November 2009 3:43 PM To: Robert McGhee; 'OSL Group' Subject: Re: [OSL | CCIE_Voice] MOH spoofing at remote branch Robert, You need to change the command multicast moh 239.1.1.1 port 16384 To multicast moh 239.1.1.1 port 16384 router ( Voice Vlan) ( loopback) Voice vlan for PSTN users and loopback for MOH between ip phones. Also if it is h323 gw you need ccm-manager music-on-hold. You need to no the existing commands and then add again, that's the way of resetting MOH from flash. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Robert McGhee Sent: Wednesday, 11 November 2009 3:36 PM To: 'OSL Group' Subject: [OSL | CCIE_Voice] MOH spoofing at remote branch Hi, Anyone run into issues getting MOH to play from flash for remote sites, spoofing the multicast address? I have MOH setup for multicast on CUCM and MRGL is assigned to remote site phones. Here's remote configuration: call-manager-fallback max-conferences 8 gain -6 transfer-system full-consult ip source-address 10.0.200.1 port 2000 max-ephones 2 max-dn 4 moh music_on_hold.au multicast moh 239.1.1.1 port 16384 ip multicast-routing int f0/0 ip pim dense-mode Nothing shows up with show ip mroute I don't see anything for 239.1.1.1: For giggles I added ccm-manager music-on-hold. Still nothing just the beeps. The MOH file does play when the phones are in SRST mode. I reset the streaming service and nothing, any ideas? Thanks!!! CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email
Re: [OSL | CCIE_Voice] 3750 QoS Question
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 [mailto: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 mls qos map policed-dscp 0 24 to 8 class-map match-all VVLAN-VOICE !Was in SRND but not using match access-group name VVLAN-VOICE match ip dscp ef class-map match-all VVLAN-CALL-SIGNALING !Was in SRND but not using match access-group name VVLAN-CALL-SIGNALLING match ip dscp cs3 af31 class-map match-all VVLAN-ANY match access-group name VVLAN-ANY policy-map IPPHONE+PC-BASIC class VVLAN-VOICE set ip dscp 46 police 128000 8000 exceed-action drop class VVLAN-CALL-SIGNALING set ip dscp 24 police 32000 8000 exceed-action policed-dscp-transmit class VVLAN-ANY set ip dscp 0 police 32000 8000 exceed-action policed-dscp-transmit class class-default set ip dscp 0 police 500 8000 exceed-action policed-dscp-transmit interface FastEthernet0/1 service-policy input IPPHONE+PC-BASIC ip access list extended VVLAN-VOICE permit udp x.x.x.x 0.0.0.255 any range 16384 32767 ip access list extended VVLAN-CALL-SIGNALING permit tcp x.x.x.x 0.0.0.255 any range 2000 2002 ip access list extended VVLAN-ANY permit ip x.x.x.x 0.0.0.255 any Thanks, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 3750 QoS Question
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 wormh...@sch.bme.humailto:wormh...@sch.bme.hu 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 mciarfe...@iplogic.com Date: Wednesday, November 11, 2009 8:56 pm Subject: Re: [OSL | CCIE_Voice] 3750 QoS Question To: Alex Hannah alex.han...@gmail.commailto:alex.han...@gmail.com, ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto: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.commailto:ccie_voice-boun...@onlinestudylist.com [ On Behalf Of Alex Hannah Sent: Wednesday, November 11, 2009 2:41 PM To: ccie_voice@onlinestudylist.commailto: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�CConditionally-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 mls qos map policed-dscp 0 24 to 8 class-map match-all VVLAN-VOICE !Was in SRND but not using match access-group name VVLAN-VOICE match ip dscp ef class-map match-all VVLAN-CALL-SIGNALING !Was in SRND but not using match access-group name VVLAN-CALL-SIGNALLING match ip dscp cs3 af31 class-map match-all VVLAN-ANY match access-group name VVLAN-ANY policy-map IPPHONE+PC-BASIC class VVLAN-VOICE set ip dscp 46 police 128000 8000 exceed-action drop class VVLAN-CALL-SIGNALING set ip dscp 24 police 32000 8000 exceed-action policed-dscp-transmit class VVLAN-ANY set ip dscp 0 police 32000 8000 exceed-action policed-dscp-transmit class class-default set ip dscp 0 police 500 8000 exceed-action policed-dscp-transmit interface FastEthernet0/1 service-policy input IPPHONE+PC-BASIC ip access list extended VVLAN-VOICE permit udp x.x.x.x 0.0.0.255 any range 16384 32767 ip access list extended VVLAN-CALL-SIGNALING permit tcp x.x.x.x 0.0.0.255 any range 2000 2002 ip access list extended VVLAN-ANY permit ip x.x.x.x 0.0.0.255 any Thanks, Alex ___ For more information regarding industry leading CCIE Lab training, please visit
Re: [OSL | CCIE_Voice] mgcp switchover: graceful or immediate
To add a sidenote to this. If ccm-manager fallback-mgcp is configured the call will drop when the GW registers to (or back from) SRST. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani Sent: Friday, October 30, 2009 7:14 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] mgcp switchover: graceful or immediate I have notice in same of the ip expert labs they have configured ccm-manager switchback immediate command and in other case cases it is left at default which is ccm-manager switchback graceful. If I just tested both scenarios 1) Graceful mode: when there is active call and primary UCM comes back online, mgcp gateway waits for the active call to end before switching back to primary UCM as expected 2) Immediate mode: when there is active call and primary UCM comes back online, mgcp gateway straightaway registers with primary UCM and the phone display says Temp Fail but the call doesn't drop which is a bit of surprise for me, I thought call would drop as well. So if the requirement doesn't indicate one way or another would you leave at default (graceful mode) or manually configure ccm-manager switchback immediate ?? thanks __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Firmware issue
Saw this again last night. Are you getting an error message on the IP Phone when it it upgrading? You have to stare at it. It downloads 1, 2, 3, etc files then says error. I think it might have to do with the duplex mismatch issue we get from VMware even though the switch is not reporting one. Eventually, the phone would get all the files downloaded. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of sean hurricane [shurric...@gmail.com] Sent: Saturday, October 31, 2009 7:24 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Firmware issue i am experiencing a firmware issue with the phones in my lab, i have a mix of 7960/7962/7970...but the problem only happens on 7962/7970 and my problem is constant phone reloadi have tried most of firmware on cisco.comhttp://cisco.com but none seems to help .. i am running uccm 7.1.2 and ccme 7.0.1..the problem is affecting both sip and sccp phones..i have provided most of the files it's looking for (debug tftp events and wireshark came in handy) but the phones are still resetting which is very frustrating... Anyone with these phone types i will appreciate if you can tell me what type of firmware you have ... currently i am running the following: 7970 - SIP70.8-5-2SR1S 7962 - SIP42.8-5-2SR1S 7970-SCCP70.8-5-2SR1S thanks On Sat, Oct 31, 2009 at 2:14 AM, ccie_voice-requ...@onlinestudylist.commailto:ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.commailto: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.commailto:ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.commailto: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: mgcp switchover: graceful or immediate (Daniel Rodriguez) 2. Lab 1 5.9 ? In route pattern and no error message on SIP 7961 (Girard, Jeffrey COL MIL USA) 3. Re: Lab 1 5.9 ? In route pattern and no error message on SIP 7961 (Michael Ciarfello) 4. Re: Vol 1 Lab 5.8 Testing failover to MGCP (crazi bug) -- Message: 1 Date: Fri, 30 Oct 2009 23:13:31 -0400 From: Daniel Rodriguez drodrig...@fidelus.commailto:drodrig...@fidelus.com Subject: Re: [OSL | CCIE_Voice] mgcp switchover: graceful or immediate To: 'aamir.panjw...@ivision.com.aumailto:aamir.panjw...@ivision.com.au' aamir.panjw...@ivision.com.aumailto:aamir.panjw...@ivision.com.au Cc: 'ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Message-ID: 0789a7dd55ce174eb6d97f57422151a403e2166...@nyc-exch03.fidelus.commailto:0789a7dd55ce174eb6d97f57422151a403e2166...@nyc-exch03.fidelus.com Content-Type: text/plain; charset=utf-8 I would say the same really. If you're in the habit of configuring it, definitely keep doing it (unless specifically told not to). I'd say one of the most important things to remember is not to break strategy. Hope that helps. Dan - Original Message - From: Aamir Panjwani aamir.panjw...@ivision.com.aumailto:aamir.panjw...@ivision.com.au To: Daniel Rodriguez Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Sent: Fri Oct 30 22:43:13 2009 Subject: RE: [OSL | CCIE_Voice] mgcp switchover: graceful or immediate Thanks Dan...what about h323 call preservation? When there is a active call through h323 gateway and the primary UCM goes down call drops as expected, unless I configure following call preservation command. voice class h323 1 h225 timeout tcp establish 3 call preserve limit-media-detection Now if the requirement doesn't indicate one way or another I would still think that this call preserve command is very important because what if the proctor at the time of marking notice a call drop then we could very easily lose points. Interestingly, I haven't seen this command in any of the ip expert solutions. thanks -Original Message- From: Daniel Rodriguez [mailto:drodrig...@fidelus.commailto:drodrig...@fidelus.com] Sent: Saturday, 31 October 2009 1:23 PM To: Aamir Panjwani; 'ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com' Subject: Re: [OSL | CCIE_Voice] mgcp switchover: graceful or immediate I would say it depends on what I'm being asked to do. If nothing is specified in terms of MGCP failover/switchback behavior, nothing - I would leave
Re: [OSL | CCIE_Voice] Vol 1 Lab 5.11. Using # in Xlation pattern
Works here. I usually strip it out (discard predot, trailing #) before it gets to the route pattern (becasue I've been using a single e164 route pattern as of late. Strip it off somewhere before it gets to the GK and see what happens. Don't forget, if the GK is registered with a tech prefix that has a # in it (remmeber, # is optional--it's just a cosmetic character) it might be getting confused that the digit(s) before the # is the tech prefix. What does the gatek main 10 debug say? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA [jeffrey.gir...@us.army.mil] Sent: Saturday, October 31, 2009 6:28 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 1 Lab 5.11. Using # in Xlation pattern Lab 5.11 is to send international calls from HQ to gatekeeper Labbed it up with no issues Decided to add Q in Xlation pattern to avoid having to wait for interdigit timeout This broke int calls from HQ I ran DNA to see what was happening and it appears that you can't use # in a xlation pattern. DNA shows that with the # sign in, the xlation pattern does not get matched. Take the # out and all is well I went to Ch 10 of the SRND and could not find anything about using # in xlation patterns. Anyone have any insight? Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling Party Number Type
I agree becasue there are no set rules. Depends on what the carrier wants or requires. Some are jsut happy with unknown / unknown. Some send you unknown / unknown (type/plan). From: Aamir Panjwani [aamir.panjw...@ivision.com.au] Sent: Saturday, October 31, 2009 8:19 PM To: Kumar, Narinder; Daniel Rodriguez; ciscod...@live.com; Michael Ciarfello; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Calling Party Number Type Logically I would say it should remain national in either case. But I think question requirement should clearly state as to what is expected by PSTN. If not, I would speak to proctor. -Original Message- From: Kumar, Narinder [mailto:narinder.ku...@uxcg.com.au] Sent: Sunday, 1 November 2009 10:11 AM To: Aamir Panjwani; Daniel Rodriguez; ciscod...@live.com; mciarfe...@iplogic.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Calling Party Number Type What about calling number type ? HQ calling BR1 pstn. 1st Preference call should go via BR1 gateway if BR1 gateway unavailable call go out via HQ gateway. Called number type when call go out via Br1 subscriber and when it go out via HQ type national. BUT what about calling number type, you can't mark the CALLING number type subscriber when it is going out via BR1 gateway CAN YOU ??? I thought the calling number type will be national in both the cases, can some please clarify ? -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani Sent: Saturday, 31 October 2009 12:55 PM To: Daniel Rodriguez; ciscod...@live.com; mciarfe...@iplogic.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Calling Party Number Type Thanks Dan, totally make sense.. -Original Message- From: Daniel Rodriguez [mailto:drodrig...@fidelus.com] Sent: Saturday, 31 October 2009 12:52 PM To: Aamir Panjwani; 'ciscod...@live.com'; 'mciarfe...@iplogic.com'; 'ccie_voice@onlinestudylist.com' Subject: Re: [OSL | CCIE_Voice] Calling Party Number Type You're correct - its from the perspective of the egress gateway. For example, using IPExpert labs as a point of reference, calls from HQ Gw to Spain would be international. That is, you pass the international access code and country code to the PSTN with the called number type as international. But that same call from the BR2 Gw would be considered a local call, no international access code or country code and called party type set to subscriber. It's easier to setup TEHO when you think how would I route this call if it was dialed locally? - by locally I mean the from the same location of the egress gateway. Hope that helps! - Dan - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Cisco Dave ciscod...@live.com; mciarfe...@iplogic.com mciarfe...@iplogic.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Fri Oct 30 21:33:08 2009 Subject: Re: [OSL | CCIE_Voice] Calling Party Number Type It’s easy to set calling/called party number type for calls going out local gateway, however , I just wanted to confirm how it actually works in case of backup gateway and teho Backup GW: Local call goes out HQ gateway calling/called number type set to “subscriber”, if HQ GW goes down, the local call reroute via BR1 GW so in this case it’s a long distance call from the perspective of BR1 GW so called/called party number type should be set to “national” right? TEHO: If HQ user dial BR1 pstn number it should route via BR1 GW first, now in this instance calling/called party type is “subscriber” from the BR1 GW perspective, but “national” from the perspective of HQ user dialing..so not sure which one is correct?? I guess what I am getting at is when setting calling/called party number type, do we look at from the perspective of user initiating the call or from the perspective of the existing GW? I think it would be based on existing GW From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cisco Dave Sent: Tuesday, 27 October 2009 3:45 PM To: mciarfe...@iplogic.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Calling Party Number Type Thanks Michael, I totally agree about asking the proctor, and so far they have been very helpful whenever I had to ask them anything. I checked the SRND again and found the following. Page 10-17 Gateway Calling Party Number Localization +1415.XXX, strip pre-dot, numbering type: subscriber +1.!. strip pre-dot, numbering type: national IPExpert Lab 5 also shows that when a call is made to an international number that the calling number type should be set to international. This seems to indicate that the the calling number type coincide with the called number type. Can anyone confirm
Re: [OSL | CCIE_Voice] VOL 2 Lab 3 - practice OEQs
4) I would have been more precise and said CCM MGCP Backhaul. Only becasue it's a different port (2448 if memory serves.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani [aamir.panjw...@ivision.com.au] Sent: Saturday, October 31, 2009 10:04 PM To: Brian Valentine; OSL Group Subject: Re: [OSL | CCIE_Voice] VOL 2 Lab 3 - practice OEQs 1) Zone local CME no shut 2) Gateway has been assigned directly to route pattern 3) Single line 4) MGCP From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brian Valentine Sent: Sunday, 1 November 2009 12:20 PM To: OSL Group Subject: [OSL | CCIE_Voice] VOL 2 Lab 3 - practice OEQs Looking for the answer key to the OEQ section of VOL 2 Lab 3. They aren't in the PG. I think I know the answer and would have passed this section. Could someone share the answers with me? Brian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] fun with Volume2, Lab 5
Thanks Peter, Done. That was a good twist. If anyone is trying these, here is another set for the same lab. Appendix A config: (just my label, don't look for an appendix anywhere) Make HQ gateway H.323 - source from the loopback0 interface Make Br1 a SIP gateway - your choice for SIP source packets. Keep Br2 as MGCP Use the least number of route patterns as possible. Question 2.8. Must register your GK using the loopback0 interface as asked. Don't invalidate the requirements of appendix A, above. You may not change the PSTN-WAN GK config. MoH must also work. And you may not change the PSTN ephone-dn config to test whether your CAC is working or not. Explain your answer. Change one HQ phone to SIP and one BR1 phone to SIP. Don't invalidate Q3.1. Don't invalidate anything else that was originally asked in that lab. From: Peter Slow [peter.s...@gmail.com] Sent: Wednesday, October 28, 2009 3:59 PM To: Michael Ciarfello Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] fun with Volume2, Lab 5 On Wed, Oct 28, 2009 at 12:47 AM, Michael Ciarfello mciarfe...@iplogic.com wrote: If you are waiting for more labs to come out, I twisted lab5 around to add Do lab 3 but only use a total of 3 route patterns to do it. you can Ignore post dial delay requirements. all your transformations and callback numbers still have to display proper +... format number ignore TEHO requirement with all your + dialing configured properly and with local route group, you shoudl be able to make ALL non-emergency calls use a single RP, \+.! now fix teho / dialing timeout issues by adding no more than (number of sites +1) route patterns. (or some other way if you know one.) -Pete the following. You still have to meet all the requirements of all the questions in that MOC Lab: 1. Optimize the number of configured objects. I think I got route patterns down to 5 or 6. Didn't feel like moving the 3 emergency route patterns. So the number can be less. 2. Then added in AAR for all sites. One route pattern (6 or 7 total now,) one AAR group. Remember to keep everything working that is supposed to work in AAR mode. 3. Then added in TEHO for all sites according to how the SRND discusses. One RP per site. You decide on the TEHO method. Keep in mind 3 sites might be 100. I might move the QoS from the FRF.12 over to MLP and keep RSVP working (putting both serial links in the mlp bundle.) Something is funny with the equal cost paths anyways. Maybe turning off cef would balance it better. Feel free to post your own topics. ___ 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
Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
Does stop routing on unallocated number help? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA [jeffrey.gir...@us.army.mil] Sent: Friday, October 30, 2009 9:06 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP Have completed the config and calls and transformations occur as they should. To test the failover from HQ SIP GW to MGCP, I follow the instructions in the PG and do a shut on the voice port on HQ GW. I retry the call and get reorder tone. If a no shut the voice port and then go and reverse the priority of the GWs in the RL (putting BR1 on top of HQ) and then retry the call - it completes out through BR1 as it should with the proper ANI. So, it does not appear that doing a shut on the HQ voice port is the right way to test failover. Anybody else have this issue or have a better way to test failover? Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vol 2 lab 3
Part 1. (CUE AA) Look around the menus some more. Part 2. Look around the subscriber edit menus some more. Download and read the Audiotext applications docuemnt from the Ciscounitytools.com web site. Part2B (4-digit to BR2) Unity Connection by default only allows you to dial extensions it knows about. So seek a way to learn about BR2's extension or dial arbitrary extensions. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Friday, October 30, 2009 8:58 PM To: Aamir Panjwani Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] vol 2 lab 3 Still on Vol2 lab 3 when you can in from the pstn to 3100, you can only dial the br2 extension. when you dial branch 1 or hq extension you get you have entered an invalid entry also when you call the UC auto attendant and you dial hq and br1 phones. you get the message that they are unavailable even though they are and when you dial the br2 extension . you get the message i did not recognize that as a valid entry in summary you can not call across through the different AA Does any know what i need to do for these to wrk well On Sat, Oct 31, 2009 at 1:49 AM, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, Sorry i mean it is registered now Regards On Sat, Oct 31, 2009 at 1:48 AM, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hi, i have reloaded the cue and its still not registering Thanks On Sat, Oct 31, 2009 at 1:28 AM, Aamir Panjwani aamir.panjw...@ivision.com.aumailto:aamir.panjw...@ivision.com.au wrote: 3000 CUE route point make sure it is associated with cue jtapi users and then reload the cue 5000 unity connection would work without registration as long as you specify voicemail profile and call forward all to voicemail on it . From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Saturday, 31 October 2009 11:25 AM To: OSL Group Subject: [OSL | CCIE_Voice] vol 2 lab 3 Hello, working on the messaging question, the 3600 cti route point registered the 5000 and 3000 CTI route point are not registering. Any idea as to what i might be missing __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 5.9 ? In route pattern and no error message on SIP 7961
I did it with a translation pattern: 91900? with urgent priority will give reorder or the precedence message in your case after you dial 91900 91900XXX with or without urgent priority will give reorder or precedence message after you dial the full 919004522138 number. 91900? without urgent priority will allow you to dial and it will ring the PSTN phone becasue of the 9.1[2-9]XX[2-9]XX translation pattern I have. When you are dealing with ! or ? wildcards, rememebr that the ! or ? gets replaced with X's with the number of Xs being the number of digits you dialed. So 91900! becomes 91900XXX when you dial 919004522138 then you can determine which pattern is the best match. I think wildcards were explained well in the old troubobleshooting book by Cisco Press. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA [jeffrey.gir...@us.army.mil] Sent: Friday, October 30, 2009 11:26 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 1 5.9 ? In route pattern and no error message on SIP 7961 OK - moved on from 5.8 to a pretty simple question. Block 900 calls w/variable length digits and set a specific error message. RP set to 91900!# with do not route and exceeded precedence message set. Then it got interesting. HQ Phone 2 is a 7961 with a SIP load. Dialing 919004522138 gets blocked (as it should) but no error message was heard. Tried varying the error message with no change. So, I tried calling from HQ Phone 1 (my home lab - 5001 is a 7960 running SCCP). Dialed the same number and the call was blocked again AND the error message was played. So, anyone know why a 7961 SIP phone would not play the error message? Second, I checked the PG and it showed a RP of 91900? I have not heard of a ? Wildcard. The PG says it is 0 or more digits. I tried the PG solution RP. It did not work. As soon as I entered the 6th digit, I get message that indicates call can't be completed as dialed. I went back to the SRND Chapter 10 and could not find the ? at all Anybody used the ? and have it work? Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls through gatekeeper
Pretty sure you need an MTP to mitigate the CallManager equivalent of late codec negotiation vs CCME's early offer codec negotiation. You will need inbound fast start on CCM for calls from BR2 to HQ. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Wednesday, October 28, 2009 7:35 PM To: Kumar, Narinder; Daniel Rodriguez Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] calls through gatekeeper HQ is as shown below HQ-RTR(config)#do sh run Building configuration... Current configuration : 4870 bytes ! ! Last configuration change at 17:03:52 pdt Wed Oct 28 2009 ! NVRAM config last updated at 17:33:21 pdt Wed Oct 28 2009 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 clock timezone pst -8 clock summer-time pdt recurring network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! archive log config hidekeys ! ! ! ! controller T1 0/0/0 shutdown framing esf linecode ami pri-group timeslots 1-3,24 ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ip rsvp bandwidth ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id CUBE h323-gateway voip bind srcaddr 10.10.200.3 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! ! ! ! control-plane ! rmon event 3 log trap AutoQoS description AutoQoS SNMP traps for Voice Drops owner AutoQoS ! ! voice-port 0/0/0:23 ! ! ! sccp local FastEthernet0/0.20 sccp ccm 10.10.210.10 identifier 3 version 5.0.1 sccp ccm 10.10.210.11 identifier 2 version 5.0.1 sccp ccm 10.10.200.3 identifier 1 version 5.0.1 sccp ! sccp ccm group 2 associate ccm 2 priority 1 associate ccm 3 priority 2 associate profile 2 register xcoder ! sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate profile 1 register transcode ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 6 associate application SCCP ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 3 associate application SCCP ! ! dial-peer voice 3000 voip media flow-around incoming called-number 3... ! dial-peer voice 3001 voip destination-pattern 3... session target ras codec g711ulaw no vad ! ! gateway ! ! ! gatekeeper zone local UCM proctorlabs.comhttp://proctorlabs.com 10.10.110.1 zone local UCME protorlabs.comhttp://protorlabs.com outvia VIA zone local VIA proctorlabs.comhttp://proctorlabs.com zone prefix UCM 1... gw-priority 10 gk-trunk_2 zone prefix UCM 1... gw-priority 9 gk-trunk_1 zone prefix UCME 3... zone prefix UCM 5... gw-priority 10 gk-trunk_2 zone prefix UCM 5... gw-priority 9 gk-trunk_1 gw-type-prefix 1#* default-technology bandwidth total zone UCM 32 no shutdown ! ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 transcode max-ephones 2 max-dn 2 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6
Re: [OSL | CCIE_Voice] Proctor Labs CCIE Voice 3.0 VPIM
I think IPexpert has stated that they haven't procured the VPIM license yet. So I guess put that question on hold for now. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Andrew MacDonell [andrewmacdon...@lancasterisd.org] Sent: Thursday, October 29, 2009 6:10 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Proctor Labs CCIE Voice 3.0 VPIM Per Proctor Labs support request, I would like to post my question to ccievoicestuygroup in hopes that it might readily be solved by one of the subscribers. I am currently working on a VPIM lab with a Proctor Labs 3.0 voice lab session and I am unable to find the needed license file for activating VPIM. As this license is not needed for initial integration I would have assumed it would have been installed prior to starting the lab but this is not the case. So I can only assume that is located somewhere within the lab. If you have any information that would help please post. Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ccie voice stats
RTP looks pretty full for November-except day before Thanksgiving. hmmm. lol SJC is pretty empty. Don't worry about the statistics or the scheduling. Concentrate on your studies and when you feel you are ready. Don't sweat the small stuff as a co-worker always says. But thanks for the stats. Could be anything that this guy doesn't have access to--netiher do I. They moved, didn't recerfity, got kicked out for selling their number, etc. -- became a tornado chaser instead? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani [aamir.panjw...@ivision.com.au] Sent: Thursday, October 29, 2009 7:51 PM To: OSL Group Subject: [OSL | CCIE_Voice] ccie voice stats FYI all – only 5 new ccie voice in the last 32 days…scary figures :) Is that because it’s too challenging or just not many people attempting at the moment? http://www.networkworld.com/community/node/46893 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Say it isn't so!
Here's a different view. I never have seen the VoDs or audio books. I started watching one and decided to go back to actually configuring the equipment. The VoDs from V2 have been sitting on the shelf somewhere. I never sent in my hard drive for the V3's. Don't know about the quality, it's just that the documentation and the equipment are your biggest resources. The Vod's won't be available in the lab-the documentation will. You guys don't understand. IPexpert is providing a great service just in the MOC labs and workbooks. All the other stuff is just to one-up the competition. It has to be done, but these things are secondary resources. There are no MOC labs after #5, so I started to create my own. We have to do some work on our own too. IPCC *IS* a challenge. The documentation is more a reference than a tutorial. So either spend the hours and hours working it out, take a class, shoulder a project with a co-worker, ask a co-worker for help, participate in the ask-icd discussion groups, etc. I would not rely on the 10 IPCC questions I have seen in the workbooks. There are literally 10,000's they can ask you. If you only memorized 10, those 9990 others will kill ya. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf Sent: Wednesday, October 28, 2009 4:36 AM To: Atlanta CCIE Cc: ccie_voice@onlinestudylist.com; Scott ODonnell Subject: Re: [OSL | CCIE_Voice] Say it isn't so! Atlanta, It is my personal view. People have right to differ and i accept any differences. However,In my humble view,i would not have paid for VoD voice if i had taken a preview beforehand. May be,I had lots of expectations and product fell much shorter.Mainly for 3 reasons: 1) there was no hands-on-labs/demonstration done in VoD. 2) UCCX is an imp topic and it was missing. 3) Overall quality of product was not good. Mr.Mark is a double CCIE and i have great admiration and respect towards him .If he moved,that doesn't mean i have less respect for him.I had conveyed my thoughts before to Ipexpert support team and i think i have right to do so. Thanks. On Tue, Oct 27, 2009 at 11:14 PM, Atlanta CCIE atlantac...@gmail.commailto:atlantac...@gmail.com wrote: This doesnt look good. This VOD was the BEST as of 2 days ago according to IPX Team but as soon as the instructor moved places, the quality of the VoD becomes an issue? Not sure what to say about this. On Tue, Oct 27, 2009 at 2:00 PM, Wayne Lawson groupst...@ipexpert.commailto:groupst...@ipexpert.com wrote: We're very unhappy with the quality of the VOD that Mark created. It will be redone by Vik in early 2010. Also, stay tuned for quite a few announcements. Everything happening fits perfectly into our corporate direction, business plan and strategy. Our clients will not suffer - and will be extremely pleased with what happens at IPexpert throughout the month of November. Things aren't always as they appear! ;-) Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.commailto:wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 701 Live Assistance, Please visit: www.ipexpert.com/chathttp://www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communitieshttp://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp://www.ipexpert.com. On Oct 27, 2009, at 1:42 PM, kamal yousaf lovingprin...@gmail.commailto:lovingprin...@gmail.com wrote: I think Vik Malhi is a great great resource.His labs/even vlectures are always orientated towards practical demonstration while most others just use slides.I wish Vik had produced Voice VoD too.I have great faith in him and Wayne,and i am positive , IPexpert will keep dominating voice track. On Tue, Oct 27, 2009 at 9:39 PM, Thomas Koch koch1...@comcast.netmailto:koch1...@comcast.net wrote: I'm sure Wayne will find a suitable resource to replace Mark. No worries. Thomas J. Koch Owner/Consultant CCNA, CCVP Digitones, LLC Cell: 630-808-4910 E-mail: digito...@comcast.netmailto:digito...@comcast.net From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott ODonnell Sent: Tuesday, October 27, 2009 11:12 AM To: Brian Valentine Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Say it isn't so! Wow. That is a shame for IPExpert and truly a huge loss. Mark is really one of the best instructors I've ever
Re: [OSL | CCIE_Voice] Say it isn't so!
I think the intention is to ask you IPCC questions so you can do some basic setup and configuration in the field. Remember, IPCC 5-license came with 4.x, etc. Now it's in CUWL pro (25 users = 1 IPCC license, etc.) So asking it on the lab prepares us to be able to install it for customers so customers will like it and want to buy more licensing and do more complicated call center scripts. Cisco was telling the partners way back when everyone needs callcenter and it's true. Even simple queuing (and reporting!!) for your 3 person helpdesk is helpful. But they CAN ask anything they want and still make it a fairly short time script. So keep practicing the IPCC stuff or take your chances with the 10 questions that are out there. It's only 3-4 commands to read from a database. No reason an excel file can't be put on the IPCC server desktop you have to create the ODBC connection to and read the first line from. No reason why they can't ask XML files like holiday script (that's a popular script.), etc, etc etc. From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Wednesday, October 28, 2009 10:07 AM To: Michael Ciarfello Cc: kamal yousaf; Atlanta CCIE; ccie_voice@onlinestudylist.com; ODonnell; sc...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Say it isn't so! Michael, I don't think you've made an attempt yet but you raise a question I keep asking myself. How extensive could the UCCx scripting aspect of the exam really be? I remember hearing that in V2 some people skipped it entirely and passed, so if you're a UCCx capable person you should be able to gain some bonus points here. Given the time that you have I can't imagine they want you to write a really complex script. I'm speculating that they're going to have you setup route points, create triggers that point to an app/script and either modify something, fix something, or do something very basic to demonstrate understanding of environment. Do you think I'm under estimating the scope of the UCCx component of the exam? For those of you who may respond, I'm not asking for NDA knowledge, but more thinking out loud about this area of the exam. The level of detail that you could go into in UCCx could almost be a four-hour exam within itself, no? -Jeff On Wed, Oct 28, 2009 at 9:39 AM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Here's a different view. I never have seen the VoDs or audio books. I started watching one and decided to go back to actually configuring the equipment. The VoDs from V2 have been sitting on the shelf somewhere. I never sent in my hard drive for the V3's. Don't know about the quality, it's just that the documentation and the equipment are your biggest resources. The Vod's won't be available in the lab-the documentation will. You guys don't understand. IPexpert is providing a great service just in the MOC labs and workbooks. All the other stuff is just to one-up the competition. It has to be done, but these things are secondary resources. There are no MOC labs after #5, so I started to create my own. We have to do some work on our own too. IPCC *IS* a challenge. The documentation is more a reference than a tutorial. So either spend the hours and hours working it out, take a class, shoulder a project with a co-worker, ask a co-worker for help, participate in the ask-icd discussion groups, etc. I would not rely on the 10 IPCC questions I have seen in the workbooks. There are literally 10,000's they can ask you. If you only memorized 10, those 9990 others will kill ya. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf Sent: Wednesday, October 28, 2009 4:36 AM To: Atlanta CCIE Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; Scott ODonnell Subject: Re: [OSL | CCIE_Voice] Say it isn't so! Atlanta, It is my personal view. People have right to differ and i accept any differences. However,In my humble view,i would not have paid for VoD voice if i had taken a preview beforehand. May be,I had lots of expectations and product fell much shorter.Mainly for 3 reasons: 1) there was no hands-on-labs/demonstration done in VoD. 2) UCCX is an imp topic and it was missing. 3) Overall quality of product was not good. Mr.Mark is a double CCIE and i have great admiration and respect towards him .If he moved,that doesn't mean i have less respect for him.I had conveyed my thoughts before to Ipexpert support team and i think i have right to do so. Thanks. On Tue, Oct 27, 2009 at 11:14 PM, Atlanta CCIE atlantac...@gmail.commailto:atlantac...@gmail.com wrote: This doesnt look good. This VOD was the BEST as of 2 days ago according to IPX Team but as soon as the instructor moved places, the quality of the VoD becomes
Re: [OSL | CCIE_Voice] Say it isn't so!
Good point. I get hung up and set on the wrong course on the first sentence sometimes with the way people ask things. Well explained answer. From: cpar...@cparker.us [mailto:cpar...@cparker.us] Sent: Wednesday, October 28, 2009 10:03 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com; ODonnell; sc...@onlinestudylist.com; kamal yousaf; Atlanta CCIE Subject: RE: [OSL | CCIE_Voice] Say it isn't so! Michael, I don't think anyone here is operating under the delusion watching the VoD will transform them into a CCIE. Just like doing a bunch of mock labs wont make you one either. People learn in different ways, reading, listening, watching and doing. I think CCIE prpearation requires intense preparation and people need to do what works for them. No one method is superior over another. What matters is that the result is a well rounded capable person who can pass the lab. Chris Original Message Subject: Re: [OSL | CCIE_Voice] Say it isn't so! From: Michael Ciarfello mciarfe...@iplogic.com Date: Wed, October 28, 2009 6:39 am To: kamal yousaf lovingprin...@gmail.com, Atlanta CCIE atlantac...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, ODonnell scott.odonn...@gmail.com, sc...@onlinestudylist.com Here’s a different view. I never have seen the VoDs or audio books. I started watching one and decided to go back to actually configuring the equipment. The VoDs from V2 have been sitting on the shelf somewhere. I never sent in my hard drive for the V3’s. Don’t know about the quality, it’s just that the documentation and the equipment are your biggest resources. The Vod’s won’t be available in the lab—the documentation will. You guys don’t understand. IPexpert is providing a great service just in the MOC labs and workbooks. All the other stuff is just to one-up the competition. It has to be done, but these things are secondary resources. There are no MOC labs after #5, so I started to create my own. We have to do some work on our own too. IPCC *IS* a challenge. The documentation is more a reference than a tutorial. So either spend the hours and hours working it out, take a class, shoulder a project with a co-worker, ask a co-worker for help, participate in the ask-icd discussion groups, etc. I would not rely on the 10 IPCC questions I have seen in the workbooks. There are literally 10,000’s they can ask you. If you only memorized 10, those 9990 others will kill ya. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf Sent: Wednesday, October 28, 2009 4:36 AM To: Atlanta CCIE Cc: ccie_voice@onlinestudylist.com; Scott ODonnell Subject: Re: [OSL | CCIE_Voice] Say it isn't so! Atlanta, It is my personal view. People have right to differ and i accept any differences. However,In my humble view,i would not have paid for VoD voice if i had taken a preview beforehand. May be,I had lots of expectations and product fell much shorter.Mainly for 3 reasons: 1) there was no hands-on-labs/demonstration done in VoD. 2) UCCX is an imp topic and it was missing. 3) Overall quality of product was not good. Mr.Mark is a double CCIE and i have great admiration and respect towards him .If he moved,that doesn't mean i have less respect for him.I had conveyed my thoughts before to Ipexpert support team and i think i have right to do so. Thanks. On Tue, Oct 27, 2009 at 11:14 PM, Atlanta CCIE atlantac...@gmail.commailto:atlantac...@gmail.com wrote: This doesnt look good. This VOD was the BEST as of 2 days ago according to IPX Team but as soon as the instructor moved places, the quality of the VoD becomes an issue? Not sure what to say about this. On Tue, Oct 27, 2009 at 2:00 PM, Wayne Lawson groupst...@ipexpert.commailto:groupst...@ipexpert.com wrote: We're very unhappy with the quality of the VOD that Mark created. It will be redone by Vik in early 2010. Also, stay tuned for quite a few announcements. Everything happening fits perfectly into our corporate direction, business plan and strategy. Our clients will not suffer - and will be extremely pleased with what happens at IPexpert throughout the month of November. Things aren't always as they appear! ;-) Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.commailto:wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 701 Live Assistance, Please visit: www.ipexpert.com/chathttp://www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communitieshttp://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp
Re: [OSL | CCIE_Voice] Instructor Led Training Lab 1 Question 4.4 - Gatekeeper configuration problems
The key here is to look at the quetion and decide what they are looking for. I think you know, but you are not 100% sure. So get to the 100%. Kumar just hinted at the WAY it works. Take a leap, what else would IPExpert be looking for? How many correct possibilities are there in answering this question? Once you know that you can make it work as it, or correct it so it does work. Hopefully that was clear. There are lots of these little errors in the solutions and questions. Find them, correct them and you will be better prepared. Hope that helped. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.com] Sent: Tuesday, October 27, 2009 7:54 PM To: OSL Group Subject: [OSL | CCIE_Voice] Instructor Led Training Lab 1 Question 4.4 - Gatekeeper configuration problems I'm pretty sure there are mistakes in the solution for this lab. BTW, this module is worked on in the IPexpert bootcamps. In the question, 4 digit dialing needs to be handled between HQ and BR2 in g729. HQ; zone local US ipexpert.comhttp://ipexpert.com zone local SPAIN ipexpert.comhttp://ipexpert.com no shutdown BR2; int loopback 0 h323-gateway voip interfaces h323-gateway voip id PL ipaddr 10.10.110.1 1719 h323-gateway voip h323-id BR2-RTR h323-gateway voip tech-prefix 3 So, my first problem is figuring out why a voip id of PL is set on BR2. I'm PRETTY sure that it should be Spain. Is there any reason why this is set differently? In ths problem, we are not allowed to use a default prefix on the gateway. I believe that if the voip ID on BR2 is changed to SPAIN, it SHOULD work after the prefixes are configured. However, I'm still getting busing signals. Can someone see the problem? -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Instructor Led Training Lab 1 Question 4.4 - Gatekeeper configuration problems
sorry. off topic. So what's the crazi bug you're carrying around? How about the one from today that puts the linux filesystem in read-only mode and gets constant journal errors? taking CCM or UC down. Make sure your and your customer's backups are working every night!! From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of crazi bug [crazi...@gmail.com] Sent: Tuesday, October 27, 2009 11:28 PM To: Nara Shikamaru Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Instructor Led Training Lab 1 Question 4.4 - Gatekeeper configuration problems Each issue you face is actually different and it is hard to tell what the problem is by just knowing you get a busy signal. I'd say if you think your config is good, then make sure your gateway is registered with the gatekeeper, without which you're just standing out there in the blue when the gk is not even aware of you. :) On Tue, Oct 27, 2009 at 7:54 PM, Nara Shikamaru shikam...@kagadis.commailto:shikam...@kagadis.com wrote: I'm pretty sure there are mistakes in the solution for this lab. BTW, this module is worked on in the IPexpert bootcamps. In the question, 4 digit dialing needs to be handled between HQ and BR2 in g729. HQ; zone local US ipexpert.comhttp://ipexpert.com/ zone local SPAIN ipexpert.comhttp://ipexpert.com/ no shutdown BR2; int loopback 0 h323-gateway voip interfaces h323-gateway voip id PL ipaddr 10.10.110.1 1719 h323-gateway voip h323-id BR2-RTR h323-gateway voip tech-prefix 3 So, my first problem is figuring out why a voip id of PL is set on BR2. I'm PRETTY sure that it should be Spain. Is there any reason why this is set differently? In ths problem, we are not allowed to use a default prefix on the gateway. I believe that if the voip ID on BR2 is changed to SPAIN, it SHOULD work after the prefixes are configured. However, I'm still getting busing signals. Can someone see the problem? -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] fun with Volume2, Lab 5
If you are waiting for more labs to come out, I twisted lab5 around to add the following. You still have to meet all the requirements of all the questions in that MOC Lab: 1. Optimize the number of configured objects. I think I got route patterns down to 5 or 6. Didn't feel like moving the 3 emergency route patterns. So the number can be less. 2. Then added in AAR for all sites. One route pattern (6 or 7 total now,) one AAR group. Remember to keep everything working that is supposed to work in AAR mode. 3. Then added in TEHO for all sites according to how the SRND discusses. One RP per site. You decide on the TEHO method. Keep in mind 3 sites might be 100. I might move the QoS from the FRF.12 over to MLP and keep RSVP working (putting both serial links in the mlp bundle.) Something is funny with the equal cost paths anyways. Maybe turning off cef would balance it better. Feel free to post your own topics. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets
Don't know. Stick with what works. Seems because you have the first wildcard (which would cover the entire string in itself) then another one, the second one will take the rightmost two explicit dots and consider the third dot with the star is 0 or more occurrances, but decides on 0 occurrances for unknown reason. So you have the last two dots. If my explanation makes sense. Try it with 4 dots and a star. Should end up with 3 rightmost digits. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Monday, October 26, 2009 12:28 PM To: OSL Group Subject: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets I found something interesting when testing voice translation rule sets... I have a voice translation rule that strips everything but the last 4 digits: /^.*\(\)$/ /\1/ It's very useful and allows me to strip the last 4 digits on inbound called party without having to know the full DNIS... but then I started to mess around with my sets and inserted a wildcard: /^.*\(...*\)$/ /\1/ Passing a number through this rule results in only the last TWO digits, yet my set contains THREE . followed by a *. Does the * cancel out one of the . when inside of a set? Thanks ahead of time. - Dan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets
The regular expression language came from Unix SED, so check out that documentation. You should only need a handful of translation patterns on the lab. I can't see them asking weird stuff like take the 3, 4, 5th digits in the DNIS or the DNIS is coming into the PSTN backwards, reorder it. I had a document I was using and put in my own notes when I was studying the written 2+ years ago. I'll see if I can find it again. It had a table with a LOT of examples. If not, I'll paste it out of my notes for you's. You strip a 9, add a 9, strip a 011, add a 011, take the last 4 digits, etc. From: Wilson Bolanos [mailto:wbola...@pvt.com] Sent: Monday, October 26, 2009 2:45 PM To: Daniel Rodriguez; Michael Ciarfello; OSL Group Subject: RE: Xlate Rules: Wildcard * in Sets Does anyone know of a great whitepaper or Cisco document that explains the translation rules very well for the CCIE voice lab? Or should the SRNDs be the main source? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Monday, October 26, 2009 11:43 AM To: Michael Ciarfello; OSL Group Subject: Re: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets Thanks for the reply Mike. It seems that no matter how many dots I have in the set, the asterisk will always cancel the last dot out. I read the voice translations rule doc again and noticed this: Wildcard Combination .*Any digit followed by none or more occurrences, effectively anything including null. I understand that wildcard combination and used it before my 1st set.. But maybe I'm thinking of it the wrong way... I'm thinking of my set as dot, dot, dot, asterisk. Four separate wildcards. But if it's treated as a combination (as mentioned above) it's more like dot, dot, dot-asterisk. In other words, the first two dots are anything from 0-9 twice, then dot-asterisk is treated as a wildcard combination matching anything including null.. and for some reason goes with null. I hope that theory makes some sense. Seems like it follows your idea below. I tried with more dots and got the same result - all but the last dot was used to match the last digits. I also replaced the * with a ? and got the same result. Both characters could possible match null. Thanks again. Dan From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Monday, October 26, 2009 1:22 PM To: Daniel Rodriguez; OSL Group Subject: RE: Xlate Rules: Wildcard * in Sets Don't know. Stick with what works. Seems because you have the first wildcard (which would cover the entire string in itself) then another one, the second one will take the rightmost two explicit dots and consider the third dot with the star is 0 or more occurrances, but decides on 0 occurrances for unknown reason. So you have the last two dots. If my explanation makes sense. Try it with 4 dots and a star. Should end up with 3 rightmost digits. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Monday, October 26, 2009 12:28 PM To: OSL Group Subject: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets I found something interesting when testing voice translation rule sets... I have a voice translation rule that strips everything but the last 4 digits: /^.*\(\)$/ /\1/ It's very useful and allows me to strip the last 4 digits on inbound called party without having to know the full DNIS... but then I started to mess around with my sets and inserted a wildcard: /^.*\(...*\)$/ /\1/ Passing a number through this rule results in only the last TWO digits, yet my set contains THREE . followed by a *. Does the * cancel out one of the . when inside of a set? Thanks ahead of time. - Dan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets
That would be too mean. I don't think it's the intention of the OEQ masters to decipher the entire internal workings of the SED processing language. But I bet the source code is out there if anyone wants to tackle it. From: Aamir Panjwani [aamir.panjw...@ivision.com.au] Sent: Monday, October 26, 2009 6:44 PM To: Daniel Rodriguez; Michael Ciarfello; OSL Group Subject: RE: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets Hmm looks like a interesting OEQ :) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Tuesday, 27 October 2009 4:43 AM To: Michael Ciarfello; OSL Group Subject: Re: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets Thanks for the reply Mike. It seems that no matter how many dots I have in the set, the asterisk will always cancel the last dot out. I read the voice translations rule doc again and noticed this: Wildcard Combination .*Any digit followed by none or more occurrences, effectively anything including null. I understand that wildcard combination and used it before my 1st set.. But maybe I’m thinking of it the wrong way… I’m thinking of my set as dot, dot, dot, asterisk. Four separate wildcards. But if it’s treated as a combination (as mentioned above) it’s more like dot, dot, dot-asterisk. In other words, the first two dots are anything from 0-9 twice, then dot-asterisk is treated as a wildcard combination matching anything including null.. and for some reason goes with null. I hope that theory makes some sense. Seems like it follows your idea below. I tried with more dots and got the same result – all but the last dot was used to match the last digits. I also replaced the * with a ? and got the same result. Both characters could possible match null. Thanks again. Dan From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Monday, October 26, 2009 1:22 PM To: Daniel Rodriguez; OSL Group Subject: RE: Xlate Rules: Wildcard * in Sets Don’t know. Stick with what works. Seems because you have the first wildcard (which would cover the entire string in itself) then another one, the second one will take the rightmost two explicit dots and consider the third dot with the star is 0 or more occurrances, but decides on 0 occurrances for unknown reason. So you have the last two dots. If my explanation makes sense. Try it with 4 dots and a star. Should end up with 3 rightmost digits. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Monday, October 26, 2009 12:28 PM To: OSL Group Subject: [OSL | CCIE_Voice] Xlate Rules: Wildcard * in Sets I found something interesting when testing voice translation rule sets… I have a voice translation rule that strips everything but the last 4 digits: /^.*\(….\)$/ /\1/ It’s very useful and allows me to strip the last 4 digits on inbound called party without having to know the full DNIS… but then I started to mess around with my sets and inserted a wildcard: /^.*\(…*\)$/ /\1/ Passing a number through this rule results in only the last TWO digits, yet my set contains THREE “.” followed by a “*”. Does the * cancel out one of the “.” when inside of a set? Thanks ahead of time. - Dan __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling Party Number Type
hmmm. I never cared enough to ask. I always ask what the carrier wants (or get it off the paperwork) or just don't worry about it until callerID is not showing up somewhere. Some don't seem to care what anything is set to. Some need a specific type and plan in order to display your calling number on the destination device. Then there is the crazy reality of your calling number shows up properly on one provider's network and doesn't show up on another provider's network. Then there's ATT. humph. I would think if you set it to subscriber and your call ends up going international, the carrier would (might) modify it. Couldn't find anything on Google, so ask your carrier. If this is a test question, I would think they would tell you what they expect or ask the proctor. I'm guessing you were asking a real-life based question. lol Take care From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cisco Dave [ciscod...@live.com] Sent: Monday, October 26, 2009 11:02 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Calling Party Number Type Does anyone know of good information that details how the calling and called number type should be set? For called party I would expect it to be very straight forward: US 7 or 10 digit dialing subscriber 11 digit national 011 International Non-US (may vary) 8 digits - National 00 - International Calling party number type on the other hand seems to be a bit more of a mystery since it, at least in my mind, it is not dependent on the called party (or is it?). If I dial (US) a seven digit number (303-) that is understandably marked as subscriber. But if I dial the same number as an 11 (1714-303-) digit number is it still marked as subscriber? There are two variations, I dial 1714-303- and send those digits out to the PSTN, and secondly I strip off the 1714 before sending out the PSTN. Should one or both be set to calling party type of subscriber? So the calling party type scenarios are restated below for subscriber calls: A) 303- B) 1714-303- C) 1714-303- (1714 stripped before sending to PSTN) I will lump both international and national calling number type into this question. How does the calling number type get set in this situation? Should it be set to national in both cases? Or does it change based on the number called? Thank you, cd Windows 7: Simplify your PC. Learn more.http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volume2, lab4, 6.1, BACD
You only have two hunt groups. 3 and 10. Anytime you make a change, I like to do a no service queue, no service aa, then paste it back in. I don't trust the call application stop. From: Aamir Panjwani [aamir.panjw...@ivision.com.au] Sent: Sunday, October 25, 2009 11:09 PM To: Michael Ciarfello; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Volume2, lab4, 6.1, BACD I still can’t get this to work fully. BACD answers ok, but as soon as I press 3 to go to hunt group or 0 to go to operator the call just drops. Interestingly, if I select option 1 to dial by extension and then dial hunt group 3210 it rings on both 3001 and 3002. voice hunt-group 1 parallel list 3001,3002 pilot 3210 application service queue flash:app-b-acd-2.1.2.2.tcl param aa-hunt3 3210 param aa-hunt10 3002 param queue-len 15 param queue-manager-debugs 1 param number-of-hunt-grps 3 ! service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param number-of-hunt-grps 3 param handoff-string aa param dial-by-extension-option 1 paramspace english language en param max-time-vm-retry 2 param aa-pilot 3500 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 3600 param max-time-call-retry 90 param service-name queue dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:10.10.102.1 (voice vlan - h323 bind interface, have tried loopback as well) incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad thanks From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Friday, 23 October 2009 2:09 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Volume2, lab4, 6.1, BACD A while ago someone had troubles getting BACD working with voice hunt-group. It's interesting since the documentation doesn't mention it works with voice hunt-group, but it seems to work. I'm sharpening up on some topics and had trouble getting this to work again. BACD would answer but not send the call to the hunt-group. It would just queue the call. It was also not sending the call to an ephone-hunt group. My problem was my loopback dial-peer was pointing to the wrong IP address. It seems it must point to the IP address (on the same router) that is bound to the h323 voip bind command. I was bound to the Lo0. I was putting the session target as the phone interface. Weird, but might save someone a big headache. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Question about lab reference material
Qos is correct. old. But applicable--just needs a lot of corrections. CME SRND is There's no date on it. Nice. If you were in the SRND section, they should be the latest CME Admin is 7.1. Ignore stuff that says requires 7.1. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wilson Bolanos [wbola...@pvt.com] Sent: Sunday, October 25, 2009 10:11 PM To: Nara Shikamaru; Mark Snow Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Question about lab reference material I downloaded the QOS SRND ver 3.3 Nov. 2005. Is that the right one? Which one is the lattest for CME Express SRND and Admin Guide for the Lab? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Friday, October 23, 2009 12:25 AM To: Mark Snow Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Question about lab reference material Mark, I think someone mentioned that the following resources are available on the desktop for the exam. Do these look right? * http://www.cisco.com/cisco/web/psa/default.html * Cisco Unified Communications Solution Reference Network Design (SRND) Based on Cisco Unified Communications Manager Release 7.x * Enterprise QoS Solution Reference Network Design Guide.pdf * Cisco Unified Communications Manager Express System Administrator Guide On Thu, Oct 22, 2009 at 7:08 PM, Mark Snow ms...@ipexpert.commailto:ms...@ipexpert.com wrote: The QoS SRND, and the CUCM 7 SRND are both available on the candidate's desktop in the actual lab. Others may make their way on or off of there, but those are the only two official ones posted. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.commailto:ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communitieshttp://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 22, 2009, at 20:27, Nara Shikamaru shikam...@kagadis.commailto:shikam...@kagadis.com wrote: Can someone tell me which SRNDs are available in the lab? Also, it's my understanding that the link http://www.cisco.com/cisco/web/psa/default.html is accessible via desktop on the lab PC. Can someone confirm? -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab2 Lab5 - PQ Calculations
Smoke and mirrors no one can seem to get straight. I would ask the proctor. He might say use best practices, so you say which best practices? The out-dated QoS SRND where it tells you two different values in two different places in the same document or the CCM 7 SRND? Some say FRF.12 is not supposed to fragement the voice packets becasue they are already sized properly and won't add the FRF.12 header, so 4 bytes. Others say the header is always added. Same smoke for MLP. MLP is 6 but it's MLP over Frame Relay, so add 4 more then there is 3 more magic bytes from somewhere. It's a terrible question I hope they don't ask on the lab. Need a serial sniffer to see what's going on. I don't know anyone who has one. Maybe someone in the Wireshark discussion group or something can help. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez [drodrig...@fidelus.com] Sent: Saturday, October 24, 2009 11:26 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab2 Lab5 - PQ Calculations I'm looking at the PG for lab 2 and noticed the layer-2 overhead is different from the SRND. The PG says FR header= 4bytes... when in this lab FRF.12 LFI is configured. The SRND states that FR with FRF.12 adds an overhead of 8 bytes. The same goes for lab 5 - in this case we're using MLP LFI and the PG states that layer-2 overhead is 6 bytes, but the SRND shows MLP LFI adding 13 bytes of overhead. That's quite a difference after calculating codec bandwidth for the PQ. Is there something I'm missing? Thanks ahead of time. - Dan ___ 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
Re: [OSL | CCIE_Voice] Issues with calls through gatekeeper
So that says your call routing is working and you have a media problem. Codec problem, transcoder problem, MTP problem, fast start inbound, no wait h245, don't use intercluster trunk, etc. Spend the time and document your findings in your personal notes. Then do other scenarios--make your own up too, document, etc. Plan on spending MANY hours on this set of topics. Debugs, CCM traces, etc. Don't forget SIP to SCCP, SCCP to SIP, SIP to SIP, etc over gk, sip-trunk, etc. See what works and what doesn't. Come up with your own documentation or configuration guide and troubleshooting guide. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Friday, October 23, 2009 6:24 PM To: OSL Group Subject: [OSL | CCIE_Voice] Issues with calls through gatekeeper Hello, When i call from hq through the gatekeeper it shows connected on the br2 phones but still continues to ring and disconnect after few seconds Any ne with an idea on how to fix this thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question
An idea just popped into my head. What if you used outbound fast start? What is the well known bug? And what service parameter? “The workaround, if memory serves, is to set the service parameter mentioned above to g729” From: Brett [mailto:brett.sal...@gmail.com] Sent: Wednesday, October 21, 2009 9:08 PM To: Michael Ciarfello Cc: Mark Snow; ccie_voice@onlinestudylist.com; Dave Wong Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Mark - No critical intentions inferred :) Simply that, and to add to Michael's reply inline, if you absolutely positively have an 'everywhere' g711 region than you most likely undid your 'everywhere' g729 region in the process. I had this same issue occur with three different customers over the last 2 months and everytime it was a region issue; however, none of them were running 7.0 code. And being in the field more than the lab, I forget about lab specific code. So, if we're indeed running 7.0 code (Dave?) as the blueprint dictates then you may be hitting a well known bug, assuming our regions are spot on, wherein the ARQ neglects the region pairings and lets the IntraRegion Audio Default codec Service Parameter override its decision thereby requesting 1280 up front, since if you haven't messed with this service parameter it defaults to g711. Without the BRQ Enabled param enabled the sh gatek calls will report 128k, but both endpoints will claim g729. With it enabled, of course you'll see a BRQ in the h225 asn1 debug and sh gatek calls will then display 16k. Not a big deal as long we don't introduce session bandwidth limits that don't account for that extra 1280 ARQ. The workaround, if memory serves, is to set the service parameter mentioned above to g729 after which you should see the initial ARQ on the HQ side be 160 (or upgrade to a fixed version :) hth, Brett On Wed, Oct 21, 2009 at 8:58 AM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: If you put the GK-Trunk in a g729 only DP/region then the “it should work this way” should be a non-issue since all calls will be g729 over the GK for the original question? I think that’s what he’s getting at. And I think that is a correct statement. What if the GK trunk has MTP checked and the MTP is in a G711 only region or is getting selected as g711 because of a misconfiguration on a phone, etc? Or if the MTPs are not in any MRG so are trying to use the default. So many combinations. You have to get basic scenarios working first. Document it FULLY. Do more scenarios, document it fully, do more scenarios etc, etc ,etc. Until the experience level is comfortable enough for you. Because of all the possible call scenario types that can be asked, I think there is always a degree of playing around with settings (MTP, xcoder, fast-start, etc) but the experience and configuration confidence should greatly reduce the play around time. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Wednesday, October 21, 2009 11:44 AM To: Brett Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; Dave Wong Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Not trying to be critical, only curious - where are you going with that question? Just trying to understand the relevance provided that the this H225 GK-Controlled Trunk and the HQ Phone are both in different Regions, and that the matrix between those regions is G729. Only asking because maybe you have a thought process I haven't thought of yet :) Dave - can you in fact real quick confirm (by way of a quick table below) what *all* Regions you have defined in CUCM, and then specifically what Region you have applied to your HQ Phone and what Region you have applied to your H225-Trunk to HQ-GK, and the matrix'd BW/Codec between them? Cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.commailto:ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communitieshttp://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 21, 2009, at 10:43 AM, Brett wrote: Do you have any regions defined that are g711 everywhere? On Wed, Oct 21, 2009 at 5:17 AM, Dave Wong dwch...@gmail.commailto:dwch...@gmail.com wrote: Hi all Here's the debug h225 asn1 on the HQ GK and PSTN GK. The first set of debugs is taken when PSTN GK calls HQ GK and the second set of debugs is for a call the other way round. It clearly shows that a bandwidth
Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question
HI Brett. Looks like that bug should be fixed in 7.0(1) and newer (well, fixed in 7.0.0.whatever.) So we should be lab safe for that one according to Networkers. My intraregion was get to g711/g729 and my ARQ said 160 so I think we are good. Outbound FS: Now that I know what you were talking about, it didn't sound like it would have applied or fixed that bug. Cancel that brainstorm. So Dave, how are we doing? What version are you running? From: Brett Saling [brett.sal...@gmail.com] Sent: Friday, October 23, 2009 12:07 PM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Bug CSCsl74701 and the default intraregion audio codec service parameter, ie, the parameter that is in effect if you leave the dropdown at default in region config when specifying the intraregion codec. So even if you manually set every region to g729 everywhere, including intraregion, you will still see an ARQ of 1280 under this bug. Outbound fast start - would we see the same behavior on calls to the pstn re bw as we did inbound with fast start checked? Not sure I follow. I can lab it up when I get back in the states tomorrow unless Dave is still working through this and could test. Mobile. iPhone. GoBig. On Oct 23, 2009, at 7:37 AM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: An idea just popped into my head. What if you used outbound fast start? What is the well known bug? And what service parameter? “The workaround, if memory serves, is to set the service parameter mentioned above to g729” From: Brett [mailto:brett.sal...@gmail.com] Sent: Wednesday, October 21, 2009 9:08 PM To: Michael Ciarfello Cc: Mark Snow; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; Dave Wong Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Mark - No critical intentions inferred :) Simply that, and to add to Michael's reply inline, if you absolutely positively have an 'everywhere' g711 region than you most likely undid your 'everywhere' g729 region in the process. I had this same issue occur with three different customers over the last 2 months and everytime it was a region issue; however, none of them were running 7.0 code. And being in the field more than the lab, I forget about lab specific code. So, if we're indeed running 7.0 code (Dave?) as the blueprint dictates then you may be hitting a well known bug, assuming our regions are spot on, wherein the ARQ neglects the region pairings and lets the IntraRegion Audio Default codec Service Parameter override its decision thereby requesting 1280 up front, since if you haven't messed with this service parameter it defaults to g711. Without the BRQ Enabled param enabled the sh gatek calls will report 128k, but both endpoints will claim g729. With it enabled, of course you'll see a BRQ in the h225 asn1 debug and sh gatek calls will then display 16k. Not a big deal as long we don't introduce session bandwidth limits that don't account for that extra 1280 ARQ. The workaround, if memory serves, is to set the service parameter mentioned above to g729 after which you should see the initial ARQ on the HQ side be 160 (or upgrade to a fixed version :) hth, Brett On Wed, Oct 21, 2009 at 8:58 AM, Michael Ciarfello mailto:mciarfe...@iplogic.commciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: If you put the GK-Trunk in a g729 only DP/region then the “it should work this way” should be a non-issue since all calls will be g729 over the GK for the original question? I think that’s what he’s getting at. And I think that is a correct statement. What if the GK trunk has MTP checked and the MTP is in a G711 only region or is getting selected as g711 because of a misconfiguration on a phone, etc? Or if the MTPs are not in any MRG so are trying to use the default. So many combinations. You have to get basic scenarios working first. Document it FULLY. Do more scenarios, document it fully, do more scenarios etc, etc ,etc. Until the experience level is comfortable enough for you. Because of all the possible call scenario types that can be asked, I think there is always a degree of playing around with settings (MTP, xcoder, fast-start, etc) but the experience and configuration confidence should greatly reduce the play around time. From: mailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Wednesday, October 21, 2009 11:44 AM To: Brett Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; Dave Wong Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Not trying
Re: [OSL | CCIE_Voice] PSTN Configuration
If you are an IPexpert customer, you get them from the My Configs section on their web site. If you are not, make your own. Good practice. Make your own dial-plan. Search the Internet for how others have done it, equipment used, etc. Search the Internet for real telephone numbers--make it realistic and fun. I made a dial-plan with New York (of course), San Jose and Tokyo. Looked up real Cisco numbers on the internet. There is also a web site you can search for (forgot what it was) on the digits you need to dial for national access code, international access code, etc from each country. Search for international dialing or something like that. Or read the SRND. They have dial-plan examples and typical national / international access codes in there. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of IQBAL JAMEEL [aeenyiq...@gmail.com] Sent: Friday, October 23, 2009 12:44 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] PSTN Configuration Hello, I m going to build my home LAB. Can anyone send me the PSTN Configuration. Thanks for your help. Iqbal ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question
Nevermind, I take that back. 7.0.1.11000-2 seems to experience the same thing. Set the intraregion to g711/g722 and put bandwidth total default 16 on the gatekeeper and fast busy. Change it to g729 and the call goes through. Fails (or works) in both directions though. Just as Brett described with the ARQ, etc. So turn on BRQ to show the proper values in show gk calls and set the GK BW to (number of calls*16) + one worst case (128). Sounds like RSVP!! Or set the service parameter and define all explicit regions. Example HQ to HQ explicitly set to G711 and make sure it shows up on the top portion of the region page. Otherwise you get g729 HQ to HQ. Seems no need to reset phones or press the reset button at the top of the region page. Good one. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello [mciarfe...@iplogic.com] Sent: Friday, October 23, 2009 9:51 PM To: Brett Saling Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question HI Brett. Looks like that bug should be fixed in 7.0(1) and newer (well, fixed in 7.0.0.whatever.) So we should be lab safe for that one according to Networkers. My intraregion was get to g711/g729 and my ARQ said 160 so I think we are good. Outbound FS: Now that I know what you were talking about, it didn't sound like it would have applied or fixed that bug. Cancel that brainstorm. So Dave, how are we doing? What version are you running? From: Brett Saling [brett.sal...@gmail.com] Sent: Friday, October 23, 2009 12:07 PM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Bug CSCsl74701 and the default intraregion audio codec service parameter, ie, the parameter that is in effect if you leave the dropdown at default in region config when specifying the intraregion codec. So even if you manually set every region to g729 everywhere, including intraregion, you will still see an ARQ of 1280 under this bug. Outbound fast start - would we see the same behavior on calls to the pstn re bw as we did inbound with fast start checked? Not sure I follow. I can lab it up when I get back in the states tomorrow unless Dave is still working through this and could test. Mobile. iPhone. GoBig. On Oct 23, 2009, at 7:37 AM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: An idea just popped into my head. What if you used outbound fast start? What is the well known bug? And what service parameter? “The workaround, if memory serves, is to set the service parameter mentioned above to g729” From: Brett [mailto:brett.sal...@gmail.com] Sent: Wednesday, October 21, 2009 9:08 PM To: Michael Ciarfello Cc: Mark Snow; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; Dave Wong Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Mark - No critical intentions inferred :) Simply that, and to add to Michael's reply inline, if you absolutely positively have an 'everywhere' g711 region than you most likely undid your 'everywhere' g729 region in the process. I had this same issue occur with three different customers over the last 2 months and everytime it was a region issue; however, none of them were running 7.0 code. And being in the field more than the lab, I forget about lab specific code. So, if we're indeed running 7.0 code (Dave?) as the blueprint dictates then you may be hitting a well known bug, assuming our regions are spot on, wherein the ARQ neglects the region pairings and lets the IntraRegion Audio Default codec Service Parameter override its decision thereby requesting 1280 up front, since if you haven't messed with this service parameter it defaults to g711. Without the BRQ Enabled param enabled the sh gatek calls will report 128k, but both endpoints will claim g729. With it enabled, of course you'll see a BRQ in the h225 asn1 debug and sh gatek calls will then display 16k. Not a big deal as long we don't introduce session bandwidth limits that don't account for that extra 1280 ARQ. The workaround, if memory serves, is to set the service parameter mentioned above to g729 after which you should see the initial ARQ on the HQ side be 160 (or upgrade to a fixed version :) hth, Brett On Wed, Oct 21, 2009 at 8:58 AM, Michael Ciarfello mailto:mciarfe...@iplogic.commciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: If you put the GK-Trunk in a g729 only DP/region then the “it should work this way” should be a non-issue since all calls will be g729 over the GK for the original question? I think that’s what he’s getting at. And I think that is a correct statement. What if the GK trunk has MTP checked and the MTP is in a G711 only region or is getting selected as g711 because of a misconfiguration on a phone, etc? Or if the MTPs
Re: [OSL | CCIE_Voice] Transcoder not engaging
Glad you got it working. You are welcome. I don't think I totally understood when the xcode gets invoked until you asked your question. Seems only on outgoing dial-peers or voice register pool's, etc. Not incoming stuff. For incoming, you have to let the people in. Once in, then you can make them put on a clean-room suit (codec) Were you using outcall MWI (and were you using CUE?) That's why Ipexpert likes to put an incoming called-number of the MWI's on the same dial-peer as the CUE dial-peer. DP0 is ok, as long as you want to apply the settings that are contained in DP0. I can't rememebr what they were. Easier to debug if you just make your own since you can see the config. Take care. From: Jeff Cotter [jcot...@voxns.com] Sent: Wednesday, October 21, 2009 8:15 PM To: vccie2010; Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Transcoder not engaging Thanks for the replies. DP 1 is the incoming DP I did not include the other with session target of RAS. The problem turned out to be the lack of voice-class codec command on DP 1. Interestingly enough this also broke MW notification!! To summarize- remote site (another CME) has dial-peer 10 session target ras with codec g711 hardcoded in DP. Terminating Site had a DP 1 defined with incoming called number of (.) and NO codec or Voice-Class Codec defined. I assumed it would default to g729 and since there would now be a codec mismatch my transcoder would be invoked. Not the case….. As soon as I configured the voice class with codecs g711 and g729 and applied to dial-peer 1 on terminating CME everything started working (Big Thanks to Michael Ciarfello for pointing this out!) including MW notification! I can now hardcode the remote DP to g729 or g711 and the call completes. If I hard code the remote DP to g729 and then make a call and let the call FNA to CUE than my transcoder is invoked. Confirmed all the above with show and debugs. I can also duplicate the problem including breaking the MW by removing the voice-class command. One other point on this is if I remove DP 1 all together and let the Default DP handle the incoming leg everything works….ARGH!! Always thought having the default DP involved was a big no no….! Thanks again for the replies and support. Jeff From: vccie2010 [mailto:vccie2...@gmail.com] Sent: Wednesday, October 21, 2009 4:26 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Transcoder not engaging I don't see session target ras on DP voip 1 Not sure how you are still getting calls working. from UCM to CME via GK. On Wed, Oct 21, 2009 at 12:29 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Having problems getting my txcoders to work on new CME. Shows registered but all g729 to g711 calls fail. Configs included. DSP farm shows as registered and enabled. Any help would be appreciated. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip voice-card 0 dsp services dspfarm sccp local FastEthernet0/0 sccp ccm 192.168.1.7 identifier 1 priority 1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 1 register localtxc ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 2 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 localtxc max-ephones 2 max-dn 20 ip source-address 192.168.1.7 port 2000 auto assign 1 to 2 url services http://192.168.1.8/voiceview/common/login.do url authentication http://192.168.1.8/voiceview/authentication/authenticate.do voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp Jan 01 2002 00:00:00 R2801#sh sccp SCCP Admin State: UP Gateway IP Address: 192.168.1.7, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.7, Port Number: 2000 Priority: 1, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.7, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 4, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf
[OSL | CCIE_Voice] Volume2, lab4, 6.1, BACD
A while ago someone had troubles getting BACD working with voice hunt-group. It's interesting since the documentation doesn't mention it works with voice hunt-group, but it seems to work. I'm sharpening up on some topics and had trouble getting this to work again. BACD would answer but not send the call to the hunt-group. It would just queue the call. It was also not sending the call to an ephone-hunt group. My problem was my loopback dial-peer was pointing to the wrong IP address. It seems it must point to the IP address (on the same router) that is bound to the h323 voip bind command. I was bound to the Lo0. I was putting the session target as the phone interface. Weird, but might save someone a big headache. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question
If you put the GK-Trunk in a g729 only DP/region then the it should work this way should be a non-issue since all calls will be g729 over the GK for the original question? I think that's what he's getting at. And I think that is a correct statement. What if the GK trunk has MTP checked and the MTP is in a G711 only region or is getting selected as g711 because of a misconfiguration on a phone, etc? Or if the MTPs are not in any MRG so are trying to use the default. So many combinations. You have to get basic scenarios working first. Document it FULLY. Do more scenarios, document it fully, do more scenarios etc, etc ,etc. Until the experience level is comfortable enough for you. Because of all the possible call scenario types that can be asked, I think there is always a degree of playing around with settings (MTP, xcoder, fast-start, etc) but the experience and configuration confidence should greatly reduce the play around time. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Wednesday, October 21, 2009 11:44 AM To: Brett Cc: ccie_voice@onlinestudylist.com; Dave Wong Subject: Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question Not trying to be critical, only curious - where are you going with that question? Just trying to understand the relevance provided that the this H225 GK-Controlled Trunk and the HQ Phone are both in different Regions, and that the matrix between those regions is G729. Only asking because maybe you have a thought process I haven't thought of yet :) Dave - can you in fact real quick confirm (by way of a quick table below) what *all* Regions you have defined in CUCM, and then specifically what Region you have applied to your HQ Phone and what Region you have applied to your H225-Trunk to HQ-GK, and the matrix'd BW/Codec between them? Cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.commailto:ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 21, 2009, at 10:43 AM, Brett wrote: Do you have any regions defined that are g711 everywhere? On Wed, Oct 21, 2009 at 5:17 AM, Dave Wong dwch...@gmail.commailto:dwch...@gmail.com wrote: Hi all Here's the debug h225 asn1 on the HQ GK and PSTN GK. The first set of debugs is taken when PSTN GK calls HQ GK and the second set of debugs is for a call the other way round. It clearly shows that a bandwidth of 128K is requested on the HQ GK probably by CUCM when PSTN calls HQ, but show gatekeeper calls show 16K. When HQ calls PSTN, the bandwidth requested was 16K. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from br1 not getting to the contact center
Are you using RSVP? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Wednesday, October 21, 2009 2:27 AM To: anil batra Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] calls from br1 not getting to the contact center will check and get back thanks On Wed, Oct 21, 2009 at 6:54 AM, anil batra anil...@yahoo.commailto:anil...@yahoo.com wrote: Seems like some other issue than transcoding. Could you please set g711 codec end to end to verify. Also you may use Cisco Dialed Number Analyzer to verify where it getting to... --- On Wed, 10/21/09, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: From: Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com Subject: Re: [OSL | CCIE_Voice] calls from br1 not getting to the contact center To: anil batra anil...@yahoo.commailto:anil...@yahoo.com Cc: OSL Group ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Date: Wednesday, October 21, 2009, 11:12 AM hello, when its dialed, it appeared it will ring for a second then it goes busy with the msg on the phone--can not reach unknown number Regards On Wed, Oct 21, 2009 at 6:35 AM, anil batra anil...@yahoo.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=anil...@yahoo.com wrote: are you getting busy or fast busy when you dial into trigger from BR1, also do you see any tnascoding sessions on HQ at that time. The next step will be to look at CUCM traces --- On Wed, 10/21/09, Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com wrote: From: Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com Subject: Re: [OSL | CCIE_Voice] calls from br1 not getting to the contact center To: anil batra anil...@yahoo.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=anil...@yahoo.com Cc: OSL Group ccie_voice@onlinestudylist.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com Date: Wednesday, October 21, 2009, 11:01 AM yes On Wed, Oct 21, 2009 at 5:05 AM, anil batra anil...@yahoo.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=anil...@yahoo.com wrote: what's the status of transcoder...is it showing registered on HQ. --- On Wed, 10/21/09, Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com wrote: From: Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com Subject: Re: [OSL | CCIE_Voice] calls from br1 not getting to the contact center To: anil batra anil...@yahoo.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=anil...@yahoo.com Cc: OSL Group ccie_voice@onlinestudylist.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com Date: Wednesday, October 21, 2009, 9:33 AM Hello, i applied the HQ device pool to the ccx and the cti rp hq device pool has 9729 to br1 that is the reson i confiured the transcoder On Wed, Oct 21, 2009 at 5:02 AM, Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com wrote: On Wed, Oct 21, 2009 at 4:57 AM, anil batra anil...@yahoo.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=anil...@yahoo.com wrote: - I hope you don't have any location related issue, try location as hub_none and see. - What's the DP applied to UCCX related CTI RP, Ports and on UCCX - OR simply test with the codec as G711 between HQ to BR1. --- On Wed, 10/21/09, Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com wrote: From: Omotayo adefilabi...@gmail.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=adefilabi...@gmail.com Subject: [OSL | CCIE_Voice] calls from br1 not getting to the contact center To: OSL Group ccie_voice@onlinestudylist.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com Date: Wednesday, October 21, 2009, 8:58 AM Hello, i have addded a transcoder to the hq router but when i call the trigger DN from a br1 phone it gives a bust tone Anyone with an idea on how to fix it Regards -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] uccx editor 7.0x
Well, what error do you get? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anupam TYAGI Sent: Wednesday, October 21, 2009 12:04 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] uccx editor 7.0x I have download uccx editor 7.0 on my xp machine , but i am not able to launch it .Can some one help me out what can be the reason .. Thanks Anupam ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Transcoder not engaging
Where are you calling from and to? I don't like, but don't know if it makes a difference. You have sdspfarm session 4, but only 2 sessions in your dspfarm profile. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Wednesday, October 21, 2009 3:29 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Transcoder not engaging Having problems getting my txcoders to work on new CME. Shows registered but all g729 to g711 calls fail. Configs included. DSP farm shows as registered and enabled. Any help would be appreciated. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip voice-card 0 dsp services dspfarm sccp local FastEthernet0/0 sccp ccm 192.168.1.7 identifier 1 priority 1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 1 register localtxc ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 2 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 localtxc max-ephones 2 max-dn 20 ip source-address 192.168.1.7 port 2000 auto assign 1 to 2 url services http://192.168.1.8/voiceview/common/login.do url authentication http://192.168.1.8/voiceview/authentication/authenticate.do voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp Jan 01 2002 00:00:00 R2801#sh sccp SCCP Admin State: UP Gateway IP Address: 192.168.1.7, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.7, Port Number: 2000 Priority: 1, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.7, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 4, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period R2801#sh sdspfarm units mtp-1 Device:localtxc TCP socket:[2] REGISTERED in SCCP ver 0/10 actual_stream:4 max_stream 4 IP:192.168.1.7 58193 MTP YOKO keepalive 24 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab max-mtps:1, max-streams:8, alloc-streams:4, act-streams:0 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to br2 through gatekeeper
Look at the CCM trace file and see what digits are coming into CCM. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Tuesday, October 20, 2009 9:23 PM To: OSL Group Subject: [OSL | CCIE_Voice] calls from hq to br2 through gatekeeper Hello, I can call from HQ to br2 through gatekeeper but i can not call from br2 to hq belowis the output f my debug Oct 21 02:21:39.068: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Oct 21 02:21:39.068: ////GK/gk_rassrv_arq: arqp=0x48F4DF10,crv=0x17, answerCall=0 Oct 21 02:21:39.068: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Oct 21 02:21:39.068: //4BC4D26E80A1/4BC56E9680A3/GK/gk_dns_query: No Name servers Oct 21 02:21:39.068: //4BC4D26E80A1/4BC56E9680A3/GK/rassrv_get_addrinfo: (1#1002) Matched tech-prefix 1# Oct 21 02:21:39.068: //4BC4D26E80A1/4BC56E9680A3/GK/rassrv_get_addrinfo: (1#1002) Matched zone prefix 1 and remainder 002 Oct 21 02:21:39.068: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Oct 21 02:21:39.068: //4BC4D26E80A1/4BC56E9680A3/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x48E89F24 Oct 21 02:21:39.068: //4BC4D26E80A1/4BC56E9680A3/GK/rassrv_arq_select_viazone: matched zone is HQ, and z_invianamelen=0 Oct 21 02:21:39 HQ-RTR#.068: //4BC4D26E80A1/4BC56E9680A3/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x48E89F24 Oct 21 02:21:39.068: //4BC4D26E80A1/4BC56E9680A3/GK/rassrv_arq_select_viazone: matched zone is HQ, and z_outvianamelen=0 Oct 21 02:21:39.068: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Oct 21 02:21:39.092: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Oct 21 02:21:39.092: ////GK/gk_rassrv_arq: arqp=0x48F4DF10,crv=0x8017, answerCall=1 Oct 21 02:21:39.092: //4BC4D26E80A1/4BC56E9680A3/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Oct 21 02:21:39.116: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Oct 21 02:21:39.116: ////GK/gk_rassrv_arq: arqp=0x48F4DF10,crv=0x8018, answerCall=1 Oct 21 02:21:39.116: //4BC4D26E80A1/4BC56E9680A3/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Oct 21 02:21:39.132: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-RTR# HQ-RTR# Oct 21 02:21:48.764: ////GK/gk_process: got a TIMER event Oct 21 02:21:48.764: ////GK/gk_handle_timers thanks ___ 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
Re: [OSL | CCIE_Voice] Cisco Software Download site.
What did you purchase with your router? For example, if you purchased a 2811-CCME bundle, then it comes with SP Services by default (which is usually 12.4.15Tsomething). If at the time of ordering you wanted security, you upgraded the bundle SP Services to Advanced IP Services. In this case, you SHOULD choose factory upgrade for bundles to upgrade your advanced ip from the factory installed 12.4.15 to your required version, say 12.4.22t2. They are the same but one is more correct than the other. Maybe someone else's account would only display the factory upgrade for bundles if that is the only router they ever ordered and is associated with their account. Maybe Cisco is just tracking internally. REMEMBER, that the download page says you may be liable for downloading software you are not licensed for. Never heard of Cisco going after people, but they might start to enforce it some day. TAC is getting MUCH stricter in opening cases. Need the serial number now most of the time. Software downloads might someday too. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Monday, October 19, 2009 1:20 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco Software Download site. Going to upgrade the IOS on my new router, Cisco has two packages for each IOS. Can someone explain the difference between The Feature Set Factory Upgrade For Bundles and just the straight IOS feature set? Thanks. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Software Download site.
Yes, that would be the safest. They are both the same IOS bit-bit, it's just how Cisco wants you to order it for book-keeping. They don't seem to be enforcing which one, but might someday. Might as well learn the correct method. From: Jeff Cotter [mailto:jcot...@voxns.com] Sent: Monday, October 19, 2009 1:47 PM To: Michael Ciarfello; ccie_voice@onlinestudylist.com Subject: RE: Cisco Software Download site. Yes I ordered the 2801 which came with the SP services 12.4.15 T(10) as you mentioned. It also came with CME 4.0.1. I would like to upgrade the IOS as to support CME 7.X and then download and install the correct TAR file for CME 7.X. If I understand you correctly I would choose the Feature Set Factory Upgrade for Bundles for SP service 12.4.22T? Correct. From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Monday, October 19, 2009 10:42 AM To: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: RE: Cisco Software Download site. What did you purchase with your router? For example, if you purchased a 2811-CCME bundle, then it comes with SP Services by default (which is usually 12.4.15Tsomething). If at the time of ordering you wanted security, you upgraded the bundle SP Services to Advanced IP Services. In this case, you SHOULD choose factory upgrade for bundles to upgrade your advanced ip from the factory installed 12.4.15 to your required version, say 12.4.22t2. They are the same but one is more correct than the other. Maybe someone else's account would only display the factory upgrade for bundles if that is the only router they ever ordered and is associated with their account. Maybe Cisco is just tracking internally. REMEMBER, that the download page says you may be liable for downloading software you are not licensed for. Never heard of Cisco going after people, but they might start to enforce it some day. TAC is getting MUCH stricter in opening cases. Need the serial number now most of the time. Software downloads might someday too. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Monday, October 19, 2009 1:20 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco Software Download site. Going to upgrade the IOS on my new router, Cisco has two packages for each IOS. Can someone explain the difference between The Feature Set Factory Upgrade For Bundles and just the straight IOS feature set? Thanks. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] rtp-nte vs sip-notify
I found a great dtmf-relay interoperability chart in the CUE Design Guide. See if that helps. I think the design guide is a little old and says CUE only supports SIP-Notify, but it now seems to work for rtp-nte according to my testing. Try some combinations and see what happens. Try rtp-nte everywhere (cue DP, sip trunk to CCM DPs, etc.) and see if it works. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Monday, October 19, 2009 1:51 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] rtp-nte vs sip-notify I see in many labs in PG : rtp-nte and sometime it has sip-notify under voice-register pool. Which is corect please ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 10A - priority 88
I don't remember the question, but I think there were a couple of typos in those types of questions. What do you think the value should be and how did you arrive at that value? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Monday, October 19, 2009 1:55 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 10A - priority 88 under policy-map LLQ-BR1 it has priority 88 for 2 allowing calls in RSVP enabled infa. I am not able to follow the calculation (24x2)+40 =88 Anyone can shed some light on this please. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 10A - priority 88
Looks good. 2 calls is 64. Add in what Daniel says about the priority LLQ command in needing to add in layer 2 to that value. (24 + l2 * 2 calls) Worst case shouldn't go in the priority because the call is established (and is now 24) and RTP is streaming by the time the priority LLQ is hit. There is no RTP to prioritize when the call still think's it's 40. From: vccie2010 [mailto:vccie2...@gmail.com] Sent: Monday, October 19, 2009 2:05 PM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 10A - priority 88 it says Nth value be th eworst case BW to ensure that Nth call gets admitted so for 2 calls I think it should be for 1st call =24 for Nth (2nd call_ = 40 so total shd be 64 am I missing somehting here ? On Mon, Oct 19, 2009 at 10:57 AM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: I don't remember the question, but I think there were a couple of typos in those types of questions. What do you think the value should be and how did you arrive at that value? From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Monday, October 19, 2009 1:55 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 10A - priority 88 under policy-map LLQ-BR1 it has priority 88 for 2 allowing calls in RSVP enabled infa. I am not able to follow the calculation (24x2)+40 =88 Anyone can shed some light on this please. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection working without creating AXL user
Were you thinking Business Edition? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Monday, October 19, 2009 2:31 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity Connection working without creating AXL user I am surprised that Unity Connection working without creating AXL user. I have not created AXL user on CUCM and thus did not define it on Unity Connectionbut strangely VM is working fine. I can leave and rerteive VMs. I am stumped :) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection working without creating AXLuser
No, were you thinking business edition when you asked the question thinking that AXL was required and that's why you were confused why VM was working? Doesn't matter. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Monday, October 19, 2009 3:18 PM To: Daniel Rodriguez Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity Connection working without creating AXLuser Mike , No it;s not BE, it's Unity Connection on Proctors lab. On Mon, Oct 19, 2009 at 11:37 AM, Daniel Rodriguez drodrig...@fidelus.commailto:drodrig...@fidelus.com wrote: You should only need the AXL user if you plan on importing your users from CUCM. If you're importing users, you'll need to create your AXL user in CUCM, define it in UC, and associate a new AXL server with your phone system in UC. Otherwise you can manually create users in UC, completely disregard AXL, and voicemail functionality should work just fine assuming the rest of your integration is configured correctly. Hope that helps. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Monday, October 19, 2009 2:31 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity Connection working without creating AXLuser I am surprised that Unity Connection working without creating AXL user. I have not created AXL user on CUCM and thus did not define it on Unity Connectionbut strangely VM is working fine. I can leave and rerteive VMs. I am stumped :) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cant register to CME BR2 for SIP phones.
option 150 ip 10.10.110.3 Also get rid of the authentication requirements. Not needed for CCME phones on the same CCME router. Start with basic. When you get the phones registered, you can put it back in if you want to play with that or if it was a requirement. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashfaaq Poonawala [ashfaaq.poonaw...@gmail.com] Sent: Monday, October 19, 2009 11:28 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cant register to CME BR2 for SIP phones. Hi, I am not able to register my SIP phone. It does not pull over the config files properly for SIP and tries to register using SCCP and hence it rejects the registration. Can you please help me get past this point? I am attaching the traces for the config files if that might help. The traces are for SIPmac.cnf , running config and a trace for registration. Thanks, Ash ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] issues with incoming and outgoing calls to hq
CallManager received 2123945002 but doesn't have a route entry that matches it. Check your dial-peers going to CallManager. Check your gateway css. Check your gateway sig digits. Make sure gw can ping callmanager. Make sure gateway object has proper IP address. Check your h323 voip bind command. Must match ccm gw object ip address Where'd the other output come from after Plan:ISDN, Type:National From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Monday, October 19, 2009 7:18 PM To: OSL Group Subject: [OSL | CCIE_Voice] issues with incoming and outgoing calls to hq Hello, Working on lab 2. i have configured hq router as h323 gateway. calls in does not go through. the debug output gives. the cause code means from cisc documentation is ''The channel or service that the user requests is unavailable for an unknown reason. This problem is usually temporary'' Any ne with an idea on how to resolve this tahnks HQ-RTR(config)# Oct 19 23:10:07.529: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0088 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '2123942123' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '2123945002' Plan:ISDN, Type:National Oct 19 23:10:07.529: ISDN Se0/0/0:23 EVENT: process_rxstate: ces/callid 1/0x5 calltype 2 CALL_INCOMING Oct 19 23:10:07.529: ISDN Se0/0/0:23 EVENTd: isdn_get_guid: Got Guid 60786C578006 Oct 19 23:10:07.529: ISDN Se0/0/0:23 EVENT: call_incoming: call_id 0x0005, Guid = 60786C578006 Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: calltrkr_incoming_call: call_id=0x5 Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: calltrkr_setup_received: isdn_info=1200999100l, call_id=0x5 ANSWER Oct 19 23:10:07 HQ-RTR(config).533: ISDN Se0/0/0:23 EVENTd: calltrkr_setup_received: calltracker disabled Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: calltrkr_setup_received: isdn_info=1216675976l, call_id=0x5 ANSWER Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: call_incoming: b channel 0, call type is VOICE ULAW Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: call_incoming: Received a VOICE call from 2123942123 on b channel 0 at 64 Kb/s Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: sys_cons[0].clng_name=0 Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: call_incoming: Sending event to RM. Callid 5 Oct 19 23:10:07.533: ISDN Se0/0/0:23 EVENTd: call_incoming: RM returned Oct 19 23:10:07.533: ISDN CDAPI: cdapi_find_tsm found a GTD message IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,04,,1,2123945002 CGN,02,,1,y,1,2123942123 CPC,09 FCI,,,y, GCI,60786c57bc3b11de800600179421b680 : end of gtd length is 176 Oct 19 23:10:07.537: ISDN Se0/0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x48850088, call_id=0x5 Oct 19 23:10:07.537: ISDN Se0/0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x4795CABC, call_id=0x5 Oct 19 23:10:07.537: ISDN EVENTd: cc_clear_free_list freeing 0x47A764A8 Oct 19 23:10:07.537: ISDN Se0/0/0:23 EVENT: process_rxstate: ces/callid 1/0x5 calltype 2 CALL_CLEARED Oct 19 23:10:07.537: ISDN Se0/0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x4795CABC, call_id=0x5 Oct 19 23:10:07.537: ISDN Se0/0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x48850088, call_id=0x5 Oct 19 23:10:07.537: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8088 Cause i = 0x80AF - Resource unavailable, unspecified# HQ-RTR(config)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS without AD guide?
No guide. Search the OSL archives for keywords such as CUPS, Presence, ad, or IPPM From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mike O [mik...@msn.com] Sent: Monday, October 19, 2009 11:56 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUPS without AD guide? I know this has been talked about before but I can't find the thread that talks about it. Anyone have a guide? Thanks, Mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Moving IP Telephony Setup - Completed
Glad to hear it went well. You are welcome. From: Arun Kumar [arunv...@gmail.com] Sent: Sunday, October 18, 2009 4:16 AM To: Michael Ciarfello Cc: Nara Shikamaru; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Moving IP Telephony Setup - Completed Hi All, Just to give you all an update on this: With everyone's valuable info and other info, migrated whole setup of IP Telephony (6 Servers + 3 VG Transcoders + Couples of GW MGCP / H323) to new site and downtime was very less 15-20 min. In new site I kept the same subnet and ip addressing scheme as earlier so it made my life really easy and yesterday completed this whole migration and everything is seems to be working fine. So again thanks very much everyone for your valuable feedback and suggestion. Thanks Arun On Thu, Sep 24, 2009 at 5:55 AM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: IPCC SRND. 4.5, but it hasn't changed. See page 3.1 for the IPCC / CTI requirement. http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_4_5/srnd/crs4.5srnd.pdf Other than that, change the JTAPI and RMJTapi groups so that the primary CTI server ponts to the CCM server that is moved and were you will move the IPCC server to. You may have to change the device pool all the CTI ports are pointing to so that the CallManager group has listed first the CCM server that already moved. You might want to create a new DP that has only the CCM that already moved. You don't want it (CTI / JTAPI / RMJTAPI) failing over to the distant CCM server. I think that's it. Let me (us) know if you have additional questions. From: Arun Kumar [arunv...@gmail.commailto:arunv...@gmail.com] Sent: Wednesday, September 23, 2009 10:42 AM To: Michael Ciarfello Cc: Nara Shikamaru; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Moving IP Telephony Setup Hi Michael, Can you please provide me any link what re-configuration I need to do in IPCC to make the setup successful. Thanks Arun On Wed, Sep 23, 2009 at 7:23 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: IPCC must be co-located on the same LAN (simplicity. Read the IPCC SRND for full details, I think 45MB would be enough) as the CTI server it connectes to. SO move the IPCC and reconfig if necessary to connect it to the CCM that moved. From: Arun Kumar [arunv...@gmail.commailto:arunv...@gmail.com] Sent: Wednesday, September 23, 2009 7:46 AM To: Nara Shikamaru Cc: Michael Ciarfello; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Moving IP Telephony Setup Hi All, I've started my move and so far moved successfully, one Unity 5 and CCM 5 but I've to move my IPCC Servers also (version 5) and IP's are changed now. I've moved Unity and CCM and tested with New IP Changes works fine but I'm not sure on IPCC Move, what needs to be done on IPCC Servers, CCM Side ? so if anyone can give some guidance or link to some doc on this will be really helpful as without this I'll not be able to move my CCM PUB. Thanks for you all your valuable suggestions. Regards Arun On Tue, Sep 15, 2009 at 11:32 AM, Arun Kumar arunv...@gmail.commailto:arunv...@gmail.com wrote: Nara thanks very much I'll surely take these things into the consideration before going for any move. Cheers On Tue, Sep 15, 2009 at 11:14 AM, Nara Shikamaru shikam...@kagadis.commailto:shikam...@kagadis.com wrote: I would register the phones and gateways at both sites to a single host, move the free host to the new site, move its phones and gateway to the new site and have them register to it (at this point, only site-to-site dialing and voicemail will be down.), then move the other host and its phones/gateway to the new site. Anyway, there are a dozen ways to do it and all of them involve some kind of outage. Don't get tricky and try to change IP addresses between the old and new sites and you'll be fine - move your phone network and get it up and operational so that the sites have telephony. If you work with someone who knows everything and starts insisting that the subnets need to be changed at the new sites, tell them to recreate the subnets only for telephony and you can work on changing the voice network addressing scheme some other time. 99.99 percent of the time moves like these turn into disasters when the engineer decide to get fancy at the wrong time. Keep it simple. And no, do NOT rebuild the cluster. Shut hosts down gracefully, move them, and make sure the new sites can ping each other when you put the hosts in the new networks. On Mon, Sep 14, 2009 at 9:23 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Yea, that works. You don't even have to stop the servivces, just
Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question
== PSTN#sh gatekeep call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 2-3091121 16(Kbps) Endpt(s): Alias E.164Addr src EP: pstn-gw 6745738932 CallSignalAddr Port RASSignalAddr Port 192.168.145.14 1720 192.168.145.14 52507 Endpt(s): Alias E.164Addr dst EP: 1#5002 CallSignalAddr Port RASSignalAddr Port 192.168.233.10 32996 192.168.233.10 32996 HQ#sh gatekeep call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 2-309117 128(Kbps) shows 128K instead Endpt(s): Alias E.164Addr src EP: 6745738932 Endpt(s): Alias E.164Addr dst EP: hqgk_15002 CallSignalAddr Port RASSignalAddr Port 192.168.233.10 32996 192.168.233.10 32784 Case 2 - using CUCM phone with ext 5002 to dial India PSTN 9011916745738932 === HQ#sh gatekeep call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 3-328659 16(Kbps) Endpt(s): Alias E.164Addr src EP: hqgk_112123945002 CallSignalAddr Port RASSignalAddr Port 192.168.233.10 32996 192.168.233.10 32784 Endpt(s): Alias E.164Addr dst EP: 2#011916745738932 CallSignalAddr Port RASSignalAddr Port 192.168.145.14 1720 192.168.145.14 1720 PSTN#sh gatekeep call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 3-328656 16(Kbps) Endpt(s): Alias E.164Addr src EP: hqph2-sccp12123945002 Endpt(s): Alias E.164Addr dst EP: pstn-gw 2#011916745738932 CallSignalAddr Port RASSignalAddr Port 192.168.145.14 1720 192.168.145.14 52507 On Sun, Oct 18, 2009 at 12:29 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Please post configs. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Dave Wong [dwch...@gmail.commailto:dwch...@gmail.com] Sent: Saturday, October 17, 2009 11:16 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper bandwidth question Hi I have the following setup IP phone A CUCM GW -- HQ GK --- PSTN GK - PSTN GW - IP phone B The call manager is registered as a GW to the HQ GK and the PSTN GW is registered to the PSTN GK. The HQ GK trunk is placed into a region that is supposed to use g729 to all other regions on the CUCM. When IP phone A calls IP phone B, the show gatekeeper call on both HQ and PSTN GK show that 16Kbps is used. However, when IP phone B calls IP phone A, show gatekeeper call on HQ GK shows 128K of bandwidth used but the same command on PSTN GW shows 16K being used. Does anyone know the reason for this difference? Thanks in advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] How to copy CUCMA speed dials
You're actually USING IPMA in production? That's a first for me. Every customer I've dealt with said it's too difficult to use for the features it gives. lol Sorry, I don't know how. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch [kevin.dami...@vitalsite.com] Sent: Sunday, October 18, 2009 9:34 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] How to copy CUCMA speed dials Anyone know how to copy the speed dials from one assistant console to another? I believe in the old Attendant Console you could copy a text file, but not in the Manager Assistant client. We looked in every folder on the client, but can't find anything. This is on CUCMA 6.x, and will be upgrading to 7.x this week. Not sure if it will be possible in 7.x or not. Thanks, Kevin 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. ___ 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
Re: [OSL | CCIE_Voice] RSVP bandwidth values - scalability issues
Thanks Brett. I think this is going to be the answer. I have a myciscocommunity account, just haven't gotten a chance to look at any of the content. I'll look for the RSVP replay. From: Brett [brett.sal...@gmail.com] Sent: Saturday, October 17, 2009 4:07 PM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com; Otto Sanchez Subject: Re: [OSL | CCIE_Voice] RSVP bandwidth values - scalability issues SIP Preconditions afford this functionality to an extent and will be available with UCM in the 8.x release. It is available today with CME and SIP trunks in 12.4(22)T or later if memory serves. Herein, contraints to the session are included in the initial offer and the receiver will generate an answer to said offer and not alert the target until session establishment. For more 'lite' reading, check out RFC 3312 and 4032. There's also a VoE on RSVP on myciscocommunity.comhttp://myciscocommunity.com. Regards, B On Thu, Oct 15, 2009 at 7:44 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Started to look at the RSVP values again. Volume2, lab 5 had us configure 64 for two calls (on each of the frame-relay PVC's) for a total of 4 calls. We use the SRND recommended calculation. Ignore the dual-links and just say 2 calls which is 64k. (24 for one call) + 40 for worst case call. Problem is the call fails if two phones call at about the same time and the first call didn't answer yet. So it's reserving two worst case calls which exceeds the 64k configured. This represents a scalability issue becasue real-world we won't be able to control when the destination decides to pickup the call--there's the risk of running out of bandwidth. Seems callmanager SCCP needs to move to a early offer type codec advertising model like SIP. Thoughts? Real-world experiences? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] how to config video call over isdn
You can try your question over at Cisco netpro (www.cisco.com/go/netprohttp://www.cisco.com/go/netpro) or see your Cisco account manager becasue you might need different equipment than just a router. You also need to give more requirements to decide on the proper solution such as: 1. What kind of video endpoints? 2. What codecs are those video endpoints using? For example, if using a 384K codec, you will need more ISDN lines or downconvert it which the router won't do unless your endpoint can autonegotiate the lower codec. 3. As far as I know, the routers will not act as h323 to isdn gateways for video. I don';t even know if Cisco has a product for ISDN video anymore. Maybe they brought one back. We are a CCIE study group. While you might find your answers here, you are better served over at Netpro or your Cisco AM. Good luck From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of A Tommy [cokh...@gmail.com] Sent: Saturday, October 17, 2009 9:24 PM To: OnlineStudyList Subject: [OSL | CCIE_Voice] how to config video call over isdn Hii all, does anyone know how to configure router to process video call over isdn.. what command must be add in the voice gw 3800 series...? video call is already success over lan, but over pstn / isdn..it didn't work anyone can help thanks...before ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2, Question 5.1 on GK Call Routing
yea, confusing, isn't it? All transcoding and conferencing and mtp services get configured under telephony-service and are available for SCCP and SIP. As you go through the CCME Administration guide, you see other features that have a similar confusing config concept. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of P N [png_sanj...@yahoo.com] Sent: Saturday, October 17, 2009 9:51 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 2 Lab 2, Question 5.1 on GK Call Routing Hi All, The solution mentioned there is a need to config a transcoder to convert g729 to g711 which the SIP phone is using on BR2. The sccp ccm 10.10.110.3 is the IP for telephony-service for SCCP phone. Shouldn't be this transcoder should be registering to voice register global which control the SIP phone? However, can't find similar sdspfarm command for voice register global' to use The call thru GK to BR2 SCCP phone is working, but will hit fast busy when reaching BR2 SIP phone, sounds like a transcoder issue. Which debug command will show whether transcoder is invoked? And what should be the proper config? Incoming voip dialpeer on BR2 allow both g729 and g711 and BR2 SIP phone only use g711 codec. Thanks for your time! Patrick Ng ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper bandwidth question
Please post configs. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Dave Wong [dwch...@gmail.com] Sent: Saturday, October 17, 2009 11:16 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper bandwidth question Hi I have the following setup IP phone A CUCM GW -- HQ GK --- PSTN GK - PSTN GW - IP phone B The call manager is registered as a GW to the HQ GK and the PSTN GW is registered to the PSTN GK. The HQ GK trunk is placed into a region that is supposed to use g729 to all other regions on the CUCM. When IP phone A calls IP phone B, the show gatekeeper call on both HQ and PSTN GK show that 16Kbps is used. However, when IP phone B calls IP phone A, show gatekeeper call on HQ GK shows 128K of bandwidth used but the same command on PSTN GW shows 16K being used. Does anyone know the reason for this difference? Thanks in advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] b-channel order in H323
Same timeslots match on the other side? Does slot12 EXIST on the other side? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of P N Sent: Friday, October 16, 2009 12:52 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] b-channel order in H323 Hi All, I've a problem on b-channel usage order on a fractional H323, if I only turned on 12 b channels, and use it as default - descending, I will hit fast busy and get Cause i = 0x82AC18 - Requested circuit/channel not available. But if I use it as ascending, then call can go thru. Shouldn't be H323 takes care of fractional PRI in pri-group already? Config: controller E1 0/0/0 pri-group timeslots 1-12,16 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn outgoing display-ie no cdp enable ! interface GigabitEthernet0/0.103 encapsulation dot1Q 103 ip address 142.103.66.254 255.255.255.0 h323-gateway voip interface h323-gateway voip bind srcaddr 142.103.66.254 ! Debug: Oct 16 04:49:01.973: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x008C Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9838C Exclusive, Channel 12 Progress Ind i = 0x8183 - Origination address is non-ISDN Calling Party Number i = 0x0180, '32143001' Plan:ISDN, Type:Unknown Called Party Number i = 0x81, '999' Plan:ISDN, Type:Unknown Oct 16 04:49:01.985: ISDN Se0/0/0:15 Q931: RX - RELEASE_COMP pd = 8 callref = 0x808C Cause i = 0x82AC18 - Requested circuit/channel not available Thanks Patrick Ng ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volumme 2 lab 5
Normal now in later IOS versions. Reminding you the rsvp keyword has not been configured. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Friday, October 16, 2009 5:38 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Volumme 2 lab 5 Hello, When i try to configure the mtp i keep getting the information below dspfarm profile 1 mtp rpm_user_create_profile_entry :: Resource provider not registered Any ideas THANKS On Fri, Oct 16, 2009 at 3:38 AM, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, i configured the lab as required, but when i try to cal from HQ to any branch. i get Not enough Bandwidth even when i press the message button Any one with an idea of how to fix this Also, on the branch two router i have TRANSCODER and MTP as configured. the transcoder registered with the UCM but the MTP has failed to sccp local Loopback0 sccp ccm 10.10.210.10 identifier 2 sccp ccm 10.10.210.11 identifier 1 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register br2-rsvp-agent associate profile 1 register xcoder ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP ! dspfarm profile 2 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 associate application SCCP Thanks for the antcipated support ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 - Lab5 - Call Routing
#3 again. I think Phil is correct. There is no called transform for 999 anywhere in the PG I saw. So when we dial 999 we still seek out the longest match called xform because of the gw setting. The longest match (the only match--there is only 1 in that PT) is 9.! in pt-dnis-hq-gw. So we have to add a called xform for 999 and don't manipulate it. No big deal. But again, there are a lot of small mistakes in the PG. And there are many solutions that arrive at the same result. I use the proctor guide to see if I missed any requirements. The testing, especially for dial-plan is done by just dialing and seeing if they meet the workbook question requirements. The PG to make sure I didn't create extra information (route patterns, etc.) that the question didn't want. If you can't get a particular technology to work (say CUE to CCM, CUPS, etc) and the proctor guide missed something which means you can't get it to work (the famous sip.app config,) there is always the cisco documentation. Should be reading it anyways. So the PGs are nice for some things, but not an authority. I have written MANY technical documents similar to this and have a huge appreciation for the challenge in formatting, accuracy and completeness of content. It is not an easy thing to put together. But WE have the ability and knowledge to find the mistakes and correct them. Or ask / validate our findings in the forum and learn something new. Yes, there should be a centralized errata list, but you will still get repeat questions from new people on the list that may ask their question a different way or not understand it the same way (require an alternate explanation.) The above doesn't mean you shouldn't post. POST! Validate what you see. Phil thought there were some problems and suggested fixes, others validated this thoughts. But understand the between the lines also (written above.) -Original Message- From: Mark Snow [mailto:ms...@ipexpert.com] Sent: Thursday, October 15, 2009 12:38 PM To: Phil G Cc: Michael Ciarfello; CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Vol2 - Lab5 - Call Routing Phil, 1) You are correct that the voice translation-rule needed a change (actually all 3 needed that same change). (Not the ephone-dn - but just the Voice Translation-Rules 1, 2 and 3) All of the changes have been made and the new configs are now uploaded on the scripting (Load Lab Configs) server, as well as the Initial/ Final Labs configs have been change that you can find in your ipexpert.com -- Member's Area -- My Configs section. 2) Read the wording of the requirement of the 3rd bullet point for Task 2.8 and 5th bullet point for Task 2.9, and you will notice that the requirements are not as you state below - in so much that Task 2.8 does not in fact ask you to mark the calling number as National (at least if it does I am missing it :). snippit Task 2.8 In all cases, ANI should be displayed to the PSTN phone as the caller's full E164 number (this includes country code and a preceding +). You are not permitted to perform any digit manipulation at the Route Pattern or Route List Details level. /snippit Task 2.8 snippit Task 2.9 In all cases, ANI should be displayed to the PSTN phone as the caller's full E164 number (this includes country code and a preceding +), however mark it as Type Subscriber. DNIS should be marked as Type Subscriber as well. You are not permitted to perform any digit manipulation at the Route Pattern or Route List Details level /snippit Task 2.9 Task 2.9 does ask you to mark the calling number in the same fashion as 2.8 does, but adds on the requirement of marking the call as plan Subscriber. This doesn't violate what is asked of you in 2.8, only adds onto it. This means you could mark all calls with Full E164 including the + and plan of Subscriber, and you wouldn't be breaking any requirements. 3) On pg.50 of the Detailed Solutions Guide, we show creating 3 new RPs - one of which is 999 and the other of which is 911 - both of these are a more specific longer match than 9.!, and therefore would be chosen. HTH! -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 15, 2009, at 9:45 AM, Michael Ciarfello wrote: 1. Correct. Good job. I changed the translation pattern at first then ran into the outgoing call issue and went the long way around changing the num-exp's. Then realized I could have just changed the ephone-dn. Oh well. I haven't taken a look at the PG yet so can't comment on what
[OSL | CCIE_Voice] Volume2, Lab5, Questions 5.3 - srr input bandwidth
Just want to make sure my understanding is correct: srr-queue input bandwidth 4 4 That we can use any two equal values since the ratio will be the same for two equal values. First 4/(4+4) = 0.5 and second 4/(4+4) = 0.5 so that splits up the remaining bandwidth equally. 4 is default (for some reason) so maybe that's why the PG and all other documentation I've seen uses it. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RSVP bandwidth values - scalability issues
Started to look at the RSVP values again. Volume2, lab 5 had us configure 64 for two calls (on each of the frame-relay PVC's) for a total of 4 calls. We use the SRND recommended calculation. Ignore the dual-links and just say 2 calls which is 64k. (24 for one call) + 40 for worst case call. Problem is the call fails if two phones call at about the same time and the first call didn't answer yet. So it's reserving two worst case calls which exceeds the 64k configured. This represents a scalability issue becasue real-world we won't be able to control when the destination decides to pickup the call--there's the risk of running out of bandwidth. Seems callmanager SCCP needs to move to a early offer type codec advertising model like SIP. Thoughts? Real-world experiences? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volumme 2 lab 5
Make sure your MTP's are in the correct device pools. I just went through this and I can't believe I messed this up. The CCM traces had the phone devices requesting the proper 24k, but the MTPs were requesting 64k (if I was reading that mess correctly.) I was calling from BR1 to BR2. My Br1_Rsvp_MTP was in BR1_DP. My BR2_Rsvp_MTP was in Br1_DP (i'm with stupid.) So the MTP was in the same region and requesitng 64k. So the debug ip rsvp resv was requesting 96. The MTP should also be last in the MRGL. Look at the trace files why. You will see it try the transcoder first and state it's not RSVP capable, so it moves on to the next MRG in the list. This is why the PG says to not put RSVP MTP before other MTPs. Something needing a regular MTP will grab that first and incorrectly steal your allocation. You can see that in the trace files also. I think you need to turn on Locations based tracing in the service parameters to see this. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Thursday, October 15, 2009 10:38 PM To: OSL Group Subject: [OSL | CCIE_Voice] Volumme 2 lab 5 Hello, i configured the lab as required, but when i try to cal from HQ to any branch. i get Not enough Bandwidth even when i press the message button Any one with an idea of how to fix this Also, on the branch two router i have TRANSCODER and MTP as configured. the transcoder registered with the UCM but the MTP has failed to sccp local Loopback0 sccp ccm 10.10.210.10 identifier 2 sccp ccm 10.10.210.11 identifier 1 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register br2-rsvp-agent associate profile 1 register xcoder ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP ! dspfarm profile 2 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 associate application SCCP Thanks for the antcipated support ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME auto-answer
Looks like only headset auto answer and maybe you can use a variation of intercom without the intercom DN that is only accessible via the speed-dial button. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashfaaq Poonawala [ashfaaq.poonaw...@gmail.com] Sent: Thursday, October 15, 2009 3:25 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME auto-answer Hi All, Can we setup an auto-answer for a phone in CME through the router? I am looking to set the BR2 phones to auto-answer, so that i can have test calls running. Thanks, -Ash ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 911 ANI
Both places will tell you. (actually the telco will set that up for you, no one really talks to the 911 PSAP people) Metro areas are usually 10 digits because the next street over can be a different area code but report to the same PSAP. Non metro areas accept 7 or 10 because you will always be in the same area code (US). If using emergency responder, YOU will be responsible for sending the ANI out properly. It's possible no ANI may reach the PSAP or an incorrectly formatted ANI may reach the PSAP. I had a project with a county government to add on emergency responder. Was pretty unique making test 911 calls from within the PSAP building!!! YEAH, it's me. Got any digits yet? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of R Sam Sent: Wednesday, October 14, 2009 12:58 PM To: OSL Group Subject: [OSL | CCIE_Voice] 911 ANI Hi, When we call 911 , how many ANI digits are required to be sent both in the real world and the exam ? Is it 10 or 7 digits. Any help appreciated. Thanks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] interdigit Timeout Vol1_LAB 4A 4.1
Are you doing on-hook dialing? If so, no way around needing to press the dial softkey. That's the intended behavior. Try picking up the phone (handset,) dialing the number and see if there is still a timeout. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of oliver rodrigues [oliro...@gmail.com] Sent: Tuesday, October 13, 2009 8:55 PM To: Daniel Rodriguez Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] interdigit Timeout Vol1_LAB 4A 4.1 I'm having 7945 for SCCP 7911 for SIP, Both are giving me the same results. I have to manully press Dial Key. On Wed, Oct 14, 2009 at 4:39 AM, Daniel Rodriguez drodrig...@fidelus.commailto:drodrig...@fidelus.com wrote: Is this regarding SIP IP phones? If so, I believe this task was meant mainly for the phone models that don't support KPML. The older models required you to either send the digits en bloc. You would create dial plan patterns and push them to the IP phone. It's great for emergency services - your 7960 SIP endpoint wouldn't require 9-9-1-DIAL. Simply dial 911, SIP phone matches your dialed digits with a local dial pattern, then automatically sends the dialed digits to your call agent. Hope this is what you were referring to... if not, disregard every single word I said :) - Dan From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of oliver rodrigues [oliro...@gmail.commailto:oliro...@gmail.com] Sent: Tuesday, October 13, 2009 8:04 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] interdigit Timeout Vol1_LAB 4A 4.1 Hi , Its been asked There should be no inter-digit timeout when users dial this no there should be no need to press the Dial softkey. I have configured the BR2 CME Router as mentioned but this criteria No need to press the dial softkey is not being met. If someone has achieved this task, can you guide me what should be configured. Thanks, Oliver ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VOICEMAIL ICON ON THE CUPC
Specifying the voicemail username and password (web application password not voicemail pin) in the CUPC client? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo [adefilabi...@gmail.com] Sent: Tuesday, October 13, 2009 9:24 PM To: OSL Group Subject: [OSL | CCIE_Voice] VOICEMAIL ICON ON THE CUPC Hello, After sending voicemail to hq phone 2, i do not see any voice mail icon on the CUPC Any one with an idea of what am missing out Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCMfails
What does the voice class codec 1 look like? And try replacing the voice class codec with just codec g711u. Just to see what happens. From: Girard, Jeffrey COL MIL USA [mailto:jeffrey.gir...@us.army.mil] Sent: Sunday, October 11, 2009 11:35 PM To: Michael Ciarfello; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCMfails Michael - Thanks for responding Below is my voip dial peer Dial-peer voice 15 voip Dest pattern [15]... Session tar ras Voice class codec 1 Tech-prefix 1# No vad I did a debug voice dialpeer and then placed calls from CME SCCP and SIP Bloth calss selected the correct outbound dial peer Jeff - Original Message - From: Michael Ciarfello mciarfe...@iplogic.com To: Girard, Jeffrey COL MIL USA; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sun Oct 11 01:24:06 2009 Subject: RE: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCMfails What is your incoming voip dial-peer on BR2 and are you sure you are matching the correct one for each call type (incoming SIP vs incomig SCCP) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA [jeffrey.gir...@us.army.mil] Sent: Sunday, October 11, 2009 1:50 AM To: Girard, Jeffrey COL MIL USA; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCM fails I have done some more testing and have the following to add Calls from 3002 (SCCP phone at BR2) to 5002 (SIP phone at HQ) complete. So, while the call was active, I did a show call active voice command at BR2. The call is completing as G729r8. I also used the ? button on the 5002 phone to see that the call was G729. Calls from 5002 at HQ to 3006 (SIP at BR2) also complete. So, while the call was active, I did a show call active voice command at BR2. This call was completing as G711ulaw. Calls from 5002 at HQ to 3002 (SCCP at BR2) also complete. So, while the call was active, I did a show call active voice command at BR2. This call was completing as G711ulaw. I think what I am seeing is a codec mismatch problem. Though the RAS dial peer is using voice-class codec 1, something is forcing the negotiation down to 729 in one direction, but maintaining 711 in the other. What is puzzling is: what is forcing the negotiation to G729? There are no BW restrictions in the PL zone and the trunk is in the HQ region with a default codec of G711. The other thing that is puzzling is that while both the telephony-service ephones and the voice register pools both have G711ulaw as the preferred codec, apparently the SCCP phones can negotiate down to G729 while the SIP phones cannot. Im at a loss and still looking for assistance. Jeff --- Jeffrey T. Girard (Jeff) COL, 53 Future Forces Integration Directorate (FFID), Deputy - Networks office: (915)568-1240 DSN 978 Mobile: (915)727-4222 reply to: jeffrey.gir...@us.army.mil -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Saturday, October 10, 2009 7:10 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCMfails All - I have spent the last 6 hours on this issue with no success. I have gone back through the archives and found a thread from the beginning of June with Aamir Panjwani who had the same issue (as well as Otto Sanchez who piped up in the middle of the thread). Aamir's problem was solved when he enabled Inbound Faststart on the H323 GW Then in July, Jonathan Charles had the same issue. Inbound Faststart was suggested to him, and he never came back on - don't know if it worked for him. Then in late Sep Jason Hawkins had the same issue and Michael Ciarfello suggested the usual (inbound fast start) and to check G711 codecs. Don't know if his was fixed. Sometime in there, Michael also had the same/similar problem and solved it by creating a xcoder and requiring an MTP on the trunk. At the end of Aamir's thread, Vik suggested a series of solutions - 1) inbound faststart 2) G711 throughout 3) add a dspfarm xcoder. Well, I have tried all of them and none work. I have checked the PG and I believe that my configuration is correct as compared to the PG. I have voice-class codec 1 on all the dial-peers. I have tried enabling Faststart on both the trunk and GW - each individually and then together. I have also created an IOS xcoder on HQ, built a MRG MRGL on the CUCM, added that to the MRGL of the trunk, and then checked MTP required. None of these solutions worked for me. I have dismantled the dspfarm xcoder
Re: [OSL | CCIE_Voice] Volume 2 Lab 4 Question 1.2 - CUPS integration
Something that has some feedback such as resolved, etc. Netpro has a good concept on that. That way we could hopefully sort on resolved vs unresolved. Perform stronger searches than what's offered on OSL archives, etc. CertificationTalk was kind of too stratified. Also need the ability to add small attachments and in-line pictures. I never liked mailing lists, but I found a way to make this one work. I can move messages to category folders (CUP, BACD, UC, CUCM, Dial-plan, etc. or just delete the ones I don't think I'll need. Keeping my own custom archive. Anything else, I can look in the OSL archive. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi Sent: Monday, October 12, 2009 4:50 PM To: Matthew Berry; OSL Group Subject: Re: [OSL | CCIE_Voice] Volume 2 Lab 4 Question 1.2 - CUPS integration Matthew, You raise a good point- we used to have exactly what you describe (CertificationTalk). There were a few problems associated with this but I'll raise the question. Vik -- Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Matthew Berry ciscovoiceg...@gmail.com Date: Mon, 12 Oct 2009 08:47:57 -0500 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Volume 2 Lab 4 Question 1.2 - CUPS integration This is a housekeeping, organization suggestion for IP Expert. There are so many email threads that go back and forth about specific labs in the guide. Have you ever thought about creating a forum, with a different folder for each lab? That way, things would be better organized for future reference. Thanks! Matthew Berry Minneapolis, MN 55438 Mobile: 651-424-5044 www.rawreligion.com http://www.rawreligion.com Distraction: Doing an incredible job at an insignificant activity. Don't waste your life is not a catchphrase for me; it's a cliff I walk beside every day with trembling. - John Piper Contact Me http://www.linkedin.com/in/matthewjberry http://www.facebook.com/ciscovoiceguru http://twitter.com/ciscovoiceguru http://ciscovoiceguru.com http://rawreligion.com This is just a housekeeping, organization request for IP Expert. It seems that a lot of these discussions focus around specifc Thanks! Matthew Berry Minneapolis, MN 55438 Mobile: 651-424-5044 www.rawreligion.com http://www.rawreligion.com Distraction: Doing an incredible job at an insignificant activity. Don't waste your life is not a catchphrase for me; it's a cliff I walk beside every day with trembling. - John Piper Contact Me http://www.linkedin.com/in/matthewjberry http://www.facebook.com/ciscovoiceguru http://twitter.com/ciscovoiceguru http://ciscovoiceguru.com http://rawreligion.com On Mon, Oct 12, 2009 at 7:56 AM, Nara Shikamaru shikam...@kagadis.com wrote: Thanks, Phil. Much appreciated. On Mon, Oct 12, 2009 at 1:03 AM, Phil G pgciscov...@gmx.net wrote: There are 2 steps missing: Under Application/Deskphone Control/User Assingments give the user the permission to use Deskphone Control. Under Application/CUPC/CTI Profile assing user to appropriate profile. Nara Shikamaru wrote: In working though this section and doublechecking my work in the Proctor Guide, there seem to be a few steps missing in the solution (like associating the line 5002 to the end user gwashington, not just the device.) So, I'm wondering if my problem is related to anything else not mentioned in the PG. The CUPS client works fine and the question doesn't ask for Desk Phone Mode to be working, but the screenshot in the PG shows that it should be working. My client only shows Softphone, not Desk Phone. Has anyone else run into this issue? I can't seem to find the source of the problem. -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ 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
Re: [OSL | CCIE_Voice] How to approach CME basic install
How are they going to know that? And the CCM version will probably be different than the CCME version so you will be upgrading again. We know all the phones are 7965's. Is there the upgrade issues that were described before? The 65's are new enough that they should be able to have a decent firmware on them and not have the two-step conversion, etc problems? I do a show flash. Copy and paste to notepad and be quick with the highlights and deletes. Shut down the ports, config everything then turn it back up as was stated. What I keep forgetting to do is check the CCM auto registration protocol!! But if the phones upgrade twice, there is plenty of other stuff to do while the phones are doing their things. It might keep me out of the phone configs so I don't have to go back in TOO many times. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani [aamir.panjw...@ivision.com.au] Sent: Monday, October 12, 2009 7:02 PM To: Nara Shikamaru; Vik Malhi Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] How to approach CME basic install Yes we can do that UNLESS question specifically says to perform firmware conversion locally on CME From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Tuesday, 13 October 2009 9:58 AM To: Vik Malhi Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] How to approach CME basic install WHAT?! YOU CAN DO THAT?!?!?!?!?! My brain hurts. On Mon, Oct 12, 2009 at 3:43 PM, Vik Malhi vma...@ipexpert.commailto:vma...@ipexpert.com wrote: I think by far and away the quickest and best way is to not do firmware uploads on CME- do it on UCM. That means you set your TFTP to be the PUB and add the device in the UCM db wth the correct protocol. When registered point option 150 back to CME. Do not have any TFTP statements in your CME config. Do not put the “load” command within voice register global/telephony-s since you do not want to change the firmware during the registration back to CME. -- Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.comhttp://vma...@ipexpert.com/ Join our free online support and peer group communities: http://www.IPexpert.com/communitieshttp://www.ipexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Nara Shikamaru shikam...@kagadis.comhttp://shikam...@kagadis.com/ Date: Mon, 12 Oct 2009 14:52:03 -0700 To: OSL Group ccie_voice@onlinestudylist.comhttp://ccie_voice@onlinestudylist.com/ Subject: [OSL | CCIE_Voice] How to approach CME basic install I've gotten into the habit of copying the tftp-server syntax from CME-7-0-full-readme-v.1.0.txt, but this doesn't solve everything since the example syntax doesn't seem to have SIP loads. So, those have to be added manually. I still find the process lengthy and prone to error. If there's a mistake along the way, an SCCP phone could download a SIP load and you'll lose valuable time unscrewing it. Does anyone have a better way of doing this, or is it a matter of practive and getting REALLY good with notepad? -- -Shikamaru -- -Shikamaru __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCM fails
What is your incoming voip dial-peer on BR2 and are you sure you are matching the correct one for each call type (incoming SIP vs incomig SCCP) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA [jeffrey.gir...@us.army.mil] Sent: Sunday, October 11, 2009 1:50 AM To: Girard, Jeffrey COL MIL USA; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCM fails I have done some more testing and have the following to add Calls from 3002 (SCCP phone at BR2) to 5002 (SIP phone at HQ) complete. So, while the call was active, I did a show call active voice command at BR2. The call is completing as G729r8. I also used the ? button on the 5002 phone to see that the call was G729. Calls from 5002 at HQ to 3006 (SIP at BR2) also complete. So, while the call was active, I did a show call active voice command at BR2. This call was completing as G711ulaw. Calls from 5002 at HQ to 3002 (SCCP at BR2) also complete. So, while the call was active, I did a show call active voice command at BR2. This call was completing as G711ulaw. I think what I am seeing is a codec mismatch problem. Though the RAS dial peer is using voice-class codec 1, something is forcing the negotiation down to 729 in one direction, but maintaining 711 in the other. What is puzzling is: what is forcing the negotiation to G729? There are no BW restrictions in the PL zone and the trunk is in the HQ region with a default codec of G711. The other thing that is puzzling is that while both the telephony-service ephones and the voice register pools both have G711ulaw as the preferred codec, apparently the SCCP phones can negotiate down to G729 while the SIP phones cannot. Im at a loss and still looking for assistance. Jeff --- Jeffrey T. Girard (Jeff) COL, 53 Future Forces Integration Directorate (FFID), Deputy - Networks office: (915)568-1240 DSN 978 Mobile: (915)727-4222 reply to: jeffrey.gir...@us.army.mil -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Saturday, October 10, 2009 7:10 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCMfails All - I have spent the last 6 hours on this issue with no success. I have gone back through the archives and found a thread from the beginning of June with Aamir Panjwani who had the same issue (as well as Otto Sanchez who piped up in the middle of the thread). Aamir's problem was solved when he enabled Inbound Faststart on the H323 GW Then in July, Jonathan Charles had the same issue. Inbound Faststart was suggested to him, and he never came back on - don't know if it worked for him. Then in late Sep Jason Hawkins had the same issue and Michael Ciarfello suggested the usual (inbound fast start) and to check G711 codecs. Don't know if his was fixed. Sometime in there, Michael also had the same/similar problem and solved it by creating a xcoder and requiring an MTP on the trunk. At the end of Aamir's thread, Vik suggested a series of solutions - 1) inbound faststart 2) G711 throughout 3) add a dspfarm xcoder. Well, I have tried all of them and none work. I have checked the PG and I believe that my configuration is correct as compared to the PG. I have voice-class codec 1 on all the dial-peers. I have tried enabling Faststart on both the trunk and GW - each individually and then together. I have also created an IOS xcoder on HQ, built a MRG MRGL on the CUCM, added that to the MRGL of the trunk, and then checked MTP required. None of these solutions worked for me. I have dismantled the dspfarm xcoder and have reverted back to the PG solution. I don't want to move on until I am sure that my config is correct/works and that I understand what was wrong and why it was not working as per the PG. Looking for help from those smarter than I Jeff --- Jeffrey T. Girard (Jeff) COL, 53 Future Forces Integration Directorate (FFID), Deputy - Networks office: (915)568-1240 DSN 978 Mobile: (915)727-4222 reply to: jeffrey.gir...@us.army.mil -Original Message- From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Friday, October 09, 2009 10:36 PM To: Girard, Jeffrey COL MIL USA; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCM fails Yea, go back and find my post and see if that works. I had to to a g729 MTP on the HQ router to get it to work properly. Others have said it worked fine without my wacko config. I think I detailed
[OSL | CCIE_Voice] Volume2, Lab5, Initial PSTN Configs
Initial configs from 9-18-2009 for Lab5 seem have some errors? (I hope, otherwise I am committing a captain Kirk on his kobayshi maroo) or whatever it was. For example, Call from PSTN line 2 (London) to London HQ site Q931 comes in as 2059434002, type national. Question 2.3 seems to want us to remove the leading 0 and there is none coming into the HQ gateway for a local call. The PSTN-WAN config voice translation-rule 1 I think should be changed to rule 2 /^2059432785$/ /0\0/ type any subscriber plan any unknown There was no other calling manipulation on the PSTN-WAN router to match the original rule2 Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UnityConnection AA with custome .wav file
Make sure your pre-made files are in g.711 format. Go to the greeting menu and type of greeting you want to upload (standard, closed, alternate, etc.) Make sure you are not on the Call handler Basics. That is just for a recorded name that is normally not needed. Click on the Play/Record button to open up the Java application In the media master applet, click the options menu, then open file. Select your pre-made file. Don't forget to set the Callers Hear to my personal recording Don't forget to press save or your uploaded file won't save and will revert back to the old one. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan [j.jho...@gmail.com] Sent: Sunday, October 11, 2009 10:40 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UnityConnection AA with custome .wav file I asked this question once before but nw I can not find the email that answered my question. So I am sking again I want to use custom pre-made greetings for the auto-attendant. I know I can use the phone or my PC to record greetings but I want to send custom made .wav files to my greetings thanks in advance -- J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI Yahoo ID: jhogan552000 AIM ID: jhogan55 MSN ID: jhogan55 ICQ ID: 257599283 Work hard and get a check, Work smart and earn a living ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UnityConnection AA with custome .wav file
Lol. One minute too late. Do #2 like Michael Says. lol I just provided a little more detail. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch [kevin.dami...@vitalsite.com] Sent: Sunday, October 11, 2009 10:46 PM To: J Hogan; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UnityConnection AA with custome .wav file Here you go, I still have it in Outlook (from August 3rd)…..But, Mr. Ciarfello gets the credit. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Monday, August 03, 2009 11:09 PM To: J Hogan Cc: ccie list Subject: Re: [OSL | CCIE_Voice] [cisco-voip] unityConn Uploading VM files Sure. 1. Use the TRAP. That's the recorder bar you see in Greetings page, profile page to record the name, etc. IN Unity Connection you have to press the play/record button, it will download the java app (I think it is) and present the recorder bar. Then set the settings to the Unity connection IP address and the extension you want the phone to ring. Then set the record and playback to PHONE. 2. Record a file in 8 bit G711u file format using your favorite sound recorder. (the extension doesn't matter. WAV or AU or g711u) Don't let that confuse you. WAV files can encode many different formats, g711u is one of a long list of them. Then use the recorder bar menu to (what was it) paste from file? 3. Use the greeting administrator. Search Cisco web site for that. Thanks, Kevin From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan [j.jho...@gmail.com] Sent: Monday, August 03, 2009 11:44 PM Cc: ccie list Subject: Re: [OSL | CCIE_Voice] [cisco-voip] unityConn Uploading VM files Forgive me if this is documented somewhere,,,(please point me to the docs if the exist) is there a way to Upload AA files to UnityConnection? not the sript files but the audio files? so I can have personized AA with the voice of my choosing? thanks From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan Sent: Sunday, October 11, 2009 9:40 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UnityConnection AA with custome .wav file I asked this question once before but nw I can not find the email that answered my question. So I am sking again I want to use custom pre-made greetings for the auto-attendant. I know I can use the phone or my PC to record greetings but I want to send custom made .wav files to my greetings thanks in advance -- J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI Yahoo ID: jhogan552000 AIM ID: jhogan55 MSN ID: jhogan55 ICQ ID: 257599283 Work hard and get a check, Work smart and earn a living 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. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPPA in IPCCexpress HA scenario
See if a DNS SRV record works in an IP Phone service URL. Report back, because I'm curious. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.com] Sent: Saturday, October 10, 2009 4:01 PM To: OSL Group Subject: [OSL | CCIE_Voice] IPPA in IPCCexpress HA scenario This isn't addressed in the labs, but COULD be eventually so why not ask (the Proctor Labs IPCC Express servers are licensed for Premium HA.) I've configured IPCCexpress and IPPA in a scenario to test its feasibility in an upgrade at work but realized as I set up one-button login for IPPA that the URL referers to, obiously, one IP address. In a failover situation, how is this addressed? To do this properly, DNS be needed? Please forgive the question, HA is newer territory for me. -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling XForm on GW - No Transformation Being Applied
I've had good luck sending all patterns to translation patterns, manipulating there and needing no transformation pattern objects. That way, no matter what convoluted question they ask on the lab, you are covered. And you don't have to worry about SIP vs mgcp vs h323 trunks where digit manipulation on a route pattern a route list member or a xformation pattern gets overwritten, forgotten due to bugs, is inconsistent, etc. Might take a few minutes longer, but (so far on Ipexpert labs 3 and 4) I haven't said oops, didn't' see that requirement, now I have to go back and do this and that... I've been able to charge through the dial-plan questions one-by-one. Configure test and move on. At the end of the test, go back and quickly dial everything again to make sure nothing changed or was forgotten. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Friday, October 09, 2009 4:40 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Calling XForm on GW - No Transformation Being Applied Folks: I'm facing an issue with calling party transforms on my gateway that I can't seem to figure out. It's related to lab 3 section regarding calls between sites during WAN failure (ex. HQ call BR1 when BR1 in SRST). Here's my configuration... I'll stick to my example between HQ -- BR1 Phone1 x1001 for simplicity purposes. 1. x1001 CFwdUnreg: +16178631001 - CFwdUnreg CSS: SRST_CSS 2. SRST_CSS contains one partition: SRST_PT 3. Route Pattern created: \+.! - Partition: SRST_PT - Use external number mask - Strip PreDot Called Number 4. Gateway assigned Calling Party XForm CSS: HQ_GW_XFORM_CSS - CSS contains one partition: HQ_GW_XFORM_PT 5. Calling Party XForm Patterm created: 212394500X - Strip PreDot - Prefix +1 6. HQ x1001 External Number Mask: 2123945XXX Call flow: HQ calls BR1 x1001 -- CUCM routes call using CFwdUnreg destination and matches RP above -- RP sends call to assigned RL which uses Standard Local Route Group -- *At this point my calling number is 2123945001, called is 16178631001 --HQ Gateway chosen for call routing --HQ GW Calling XForm CSS has access to XForm pattern 212394500X (matching my ANI), but no transformations applied -- ISDN Q931 SETUP sends ANI of 2123945001. I've messed around by trying to match calling number in different ways while also manipulating my external number mask (just for troubleshooting purposes) but with no luck. Is there something simple that I'm missing? Has anyone else come across this issue where the Gateway Calling Party CSS clearly has a match but doesn't use it for transformation? Thanks ahead of time. - Dan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [NEWSENDER] - RE: Calling XForm on GW - No Transformation Being Applied - Message is from an unknown sender
Thanks for the clarification. I see what you are trying to do now. Correct, need the called transform. You can still go through the translation pattern for the calling party manipulation, globalize it and match the route pattern. I'm trying to remember that lab. I think I had zero calling transformations and one or two called transformations for these weird questions. I'd have to load that lab back up and look, but that's why I try to do as much as I can through translations.. IN case one of these weird things pop up. If you decided to do everythng with calling and called transformations and use device pool. then you get stuck on this question and have to re-do a lot of work because the xform patterns start to get jumbled up and confusing--unless you can make (more) specific matches to route to your weird questions. Ahh, a mess. From: Daniel Rodriguez [drodrig...@fidelus.com] Sent: Friday, October 09, 2009 5:26 PM To: Michael Ciarfello; ccie_voice@onlinestudylist.com Subject: RE: [NEWSENDER] - RE: [OSL | CCIE_Voice] Calling XForm on GW - No Transformation Being Applied - Message is from an unknown sender Thanks for the reply. I also prefer using translations over transformations whenever possible, but I don’t believe a translation would suffice for this particular task (please correct me if I’m wrong). Since a CFwdUnreg destination is a single, fixed number with one CSS, both HQ and BR2 will have to match the same pattern for digit analysis. In this case, both would hit a translation pattern that could either change the DNIS to 1617-863-1001 or 00-1-617-863-1001, but not both in order to accommodate HQ and BR2 dialing requirements. I gave this some thought too… but it could only work if it didn’t matter what GW was used for the outbound call (ex. BR2 calls BR1 unreg -- HQ GW used for outbound call). In this case, I would skip a translation and just go straight for a RP -- StndLocalRG.. but then there’s still the issue of manipulating ANI for BR2 and HQ (011num and 1-areacode and num). Dan From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Friday, October 09, 2009 5:04 PM To: Daniel Rodriguez; ccie_voice@onlinestudylist.com Subject: [NEWSENDER] - RE: [OSL | CCIE_Voice] Calling XForm on GW - No Transformation Being Applied - Message is from an unknown sender I’ve had good luck sending all patterns to translation patterns, manipulating there and needing no transformation pattern objects. That way, no matter what convoluted question they ask on the lab, you are covered. And you don’t have to worry about SIP vs mgcp vs h323 trunks where digit manipulation on a route pattern a route list member or a xformation pattern gets overwritten, forgotten due to bugs, is inconsistent, etc. Might take a few minutes longer, but (so far on Ipexpert labs 3 and 4) I haven’t said “oops, didn’t’ see that requirement, now I have to go back and do this and that…” I’ve been able to charge through the dial-plan questions one-by-one. Configure test and move on. At the end of the test, go back and quickly dial everything again to make sure nothing changed or was forgotten. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Rodriguez Sent: Friday, October 09, 2009 4:40 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Calling XForm on GW - No Transformation Being Applied Folks: I’m facing an issue with calling party transforms on my gateway that I can’t seem to figure out. It’s related to lab 3 section regarding calls between sites during WAN failure (ex. HQ call BR1 when BR1 in SRST). Here’s my configuration… I’ll stick to my example between HQ -- BR1 Phone1 x1001 for simplicity purposes. 1. x1001 CFwdUnreg: +16178631001 - CFwdUnreg CSS: SRST_CSS 2. SRST_CSS contains one partition: SRST_PT 3. Route Pattern created: \+.! - Partition: SRST_PT - Use external number mask - Strip PreDot Called Number 4. Gateway assigned Calling Party XForm CSS: HQ_GW_XFORM_CSS - CSS contains one partition: HQ_GW_XFORM_PT 5. Calling Party XForm Patterm created: 212394500X - Strip PreDot – Prefix +1 6. HQ x1001 External Number Mask: 2123945XXX Call flow: HQ calls BR1 x1001 -- CUCM routes call using CFwdUnreg destination and matches RP above -- RP sends call to assigned RL which uses Standard Local Route Group -- *At this point my calling number is 2123945001, called is 16178631001 --HQ Gateway chosen for call routing --HQ GW Calling XForm CSS has access to XForm pattern 212394500X (matching my ANI), but no transformations applied -- ISDN Q931 SETUP sends ANI of 2123945001. I’ve messed around by trying to match calling number in different ways while also manipulating my external number mask (just for troubleshooting purposes) but with no luck
Re: [OSL | CCIE_Voice] Starting cupc
Point to the CUPS server. Not to CCM. Deployment guide, chapter 1 used confusing wording. And make sure you changed the topology server name to an IP address and reboot otherwise when you login, the next time it will replace your IP address with the DNS server name again and you won't be able to login again (or have to type the server ip again.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of ABIOLA ADEFILA [adefilabi...@gmail.com] Sent: Friday, October 09, 2009 6:54 PM To: OSL Group Subject: [OSL | CCIE_Voice] Starting cupc Hello, After configuring cupc on the callmanager. Starting the cupc. it asks for usename and password i entered the userid and password configured on the end user page with the login server is the publisher-10.10.210.10 it always return login failed any one with an idea thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Redirecting Numbers
Looks like your carrier is not accepting the redirecting number or another thing in that message. Oct 9 17:51:56.552: ISDN Se0/0/0:23 Q931: RX - STATUS pd = 8 callref = 0x8001 Cause i = 0x82E373 - Information element not implemented Try unchecking outbound redirecting and try again. Or call the carrier, send them that message and see what their switch doesn't like. I'm just making a best guess. (if they can read it the Cisco message. They may setup a trap and have you call so they can view their captured data in the format they understand which is probably hex. Some of those old carrier guys are bit heads.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego [cristobalpri...@gmail.com] Sent: Friday, October 09, 2009 8:52 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Redirecting Numbers Hello I have a question on how the redirecting number works this is my problem i one of my users do a CFA to his Cell Phone number on the primary line. if I try to call this number internally. I get a fast busy this is what i see on the pri tcstkrt1(config)# Oct 9 17:51:56.508: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0001 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Display i = 0xB1, 'Large Conf Rm' Calling Party Number i = 0x0081, '2096646275' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '19162966846' Plan:ISDN, Type:National Original Called Number i = 0x0F81, '2302' Plan:Unknown, Type:Unknown Redirecting Number i = 0x8F, '2302' Plan:Unknown, Type:Unknown Oct 9 17:51:56.552: ISDN Se0/0/0:23 Q931: RX - STATUS pd = 8 callref = 0x8001 Cause i = 0x82E373 - Information element not implemented Call State i = 0x01 Oct 9 17:51:56.652: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8001 Channel ID i = 0xA98397 Exclusive, Channel 23 Oct 9 17:51:56.736: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x0001 Cause i = 0x80AF - Resource unavailable, unspecified Oct 9 17:51:56.804: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8001 Oct 9 17:51:56.852: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0001 what do i need to change to get this to work thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCM fails
Yea, go back and find my post and see if that works. I had to to a g729 MTP on the HQ router to get it to work properly. Others have said it worked fine without my wacko config. I think I detailed the config in that post. I know everything worked. Calls, supp services, Moh was playing, etc. I know when working on that, I forgot to do codec g711 on the sip voice-regi-pool's so things were getting confused. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA [jeffrey.gir...@us.army.mil] Sent: Friday, October 09, 2009 5:56 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 1 Lab 4 - CME SIP - GK - trunk - CUCM fails I have scanned the archives and it appears that several folks have had this issue. I have already tried enabling inbound faststart on the H.323 GW and the GK trunk. As with other posters who have had the same issue, CUCME SCCP calls work fine Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] AAR CSS on the GW
Backwards. AAR CSS on the device (phone or gateway). AAR group on the line. Need AAR group on the calling and called phones so the proper AAR group prefixes can be determined if you are using multiple AAR groups. If only using one AAR group like for E164 aar dialing, still need the same aar group on both sides. The phone device aar css determines where or if the calling phone can dial the destination number during an aar situation.Maybe during wan congestion, a phone that could not normally dial long distance numbers is allowed to call LD numbers to make 4-digit calls to remote sites transparent to that user. The most popular use of AAR css on the gateway is for call forward to voicemail. A call comes into a remote gateway and to a remote site phone. NO one answers and call forward noan to voicemail kicks in. Voicemail exists at another CAC location, say HQ. Since the remote site gateway was the original calling device (it called the remote site phone), **IT's** AAR CSS determines where and if it can call (to HQ voicemail DID number.) hope that helps. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of So Gwaai [sogw...@gmail.com] Sent: Friday, October 09, 2009 12:43 PM To: ccievoice Subject: [OSL | CCIE_Voice] AAR CSS on the GW For the AAR configuration, we need to set AAR CSS in the line only. How about the function in the GW since we can set this in the incoming call setting? I feel quite confusing that aar is use in between ip-phone, not between ip-phone and GW. Thanks for the help ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] FW: Vol2 Lab4 6.1 -- BACD ?
I've gotten voice hunt parallel to work with this lab with call-manager-fallback (regular SRST.) The handoff from PSTN to CUE AA to BACD to GDM works fine. Also HQ 4 digit to CUE AA to BACD to GDM., etc. Try unconfiguring your service commands then paste them back in. Try a separate lab with ephone-hunts to see if it works to get any phone to ring then go to GDM, etc. Then convert to voice hunt. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mick Vaites Sent: Thursday, October 08, 2009 3:21 AM To: Aamir Panjwani Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] FW: Vol2 Lab4 6.1 -- BACD ? With an ephone-hunt yes it works fine - and it makes sense as BACD takes control of the hunt group. Look at the CME documentation in particular the one on BACD it's quite something. The problem is as I pointed out that you cannot use voice hunt-groups which removes the option for broadcast HG's. Vik/Mark can you advise ? Best regards Mick E: m...@pobox.net.uk From: Aamir Panjwani aamir.panjw...@ivision.com.au Date: Thu, 8 Oct 2009 17:15:32 +1100 To: Mick Vaites m...@pobox.net.uk Subject: RE: [OSL | CCIE_Voice] FW: Vol2 Lab4 6.1 -- BACD ? Mick, Did you ever manage to get this working? as soon as I press 3 to go to hunt group the call drops. Thanks Aamir From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mick Vaites Sent: Thursday, 10 September 2009 10:03 AM To: ccie list Subject: [OSL | CCIE_Voice] FW: Vol2 Lab4 6.1 -- BACD ? Hi All, I'm stuggling with Vol2 Lab4 6.1 -- it looks like CUE AA -- BACD -- VoiceHunt-group (you need parallel). However I cannot get the last steps to work and re-reading the CME admin BACD docs - it appears that BACD doesn't support voice hunt-groups ? Best regards Mick E: m...@pobox.net.uk __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] uccx editor issue
Never seen that. I would guess uninstall and reinstall the editor on the IPCC server. Or reintall IPCC. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Juliana Xu [juliana...@hotmail.com] Sent: Wednesday, October 07, 2009 5:48 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] uccx editor issue Hi, On admin GUI, a script is loaded successfully, while UCCX log shows failed to load the script, invalid script. I'm pretty sure that it's the editor (coming with the UCCX installation) issue. If I opened a default script, e.g. icd.aef and didn't change anything just clicked on the save button, I would get the same error message. If I used another editor (installed on my desktop) to modify a script, then the script would be working fine. Any suggestion how to fix this problem on uccx server? Thanks in advance Juliana Windows Live: Make it easier for your friends to see what you’re up to on Facebook.http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PHONES IN SRST
That's fUnKy Are you doing stuff over a VPN? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of ABIOLA ADEFILA [adefilabi...@gmail.com] Sent: Tuesday, October 06, 2009 11:43 PM To: OSL Group Subject: [OSL | CCIE_Voice] PHONES IN SRST Hello, Working on question 3 of lab 3 when the phones goes into SRST, and they are dialled from the pstn. it shows connected with a long ring and disconnects Below is the log on the console .Oct 7 04:15:41.765: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 8632683 N/A .Oct 7 04:15:41.765: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 8632683 N/A .Oct 7 04:15:41.765: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 8632683 N/A BR1-RTR(config-subif)# Anyone with an idea why that is Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Globalization and localization
Works on mine. Make sure the firmware is at least 8-4-1. I think I had a stuck phone once on a slightly older firmware and localization wasn't working. Put it back to CCM7's 8-4-1 and it works. Try upgrading the firmware to 8-4-1 or newer. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow [ms...@ipexpert.com] Sent: Tuesday, October 06, 2009 8:41 PM To: ABIOLA ADEFILA Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Globalization and localization 7970 does not support Globalization. If the phone model ends with a 1,2,4 or 5 it support Globalization. Models ending in a 0 do not. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: mailto:ms...@ipexpert.com ms...@ipexpert.commailto:ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 6, 2009, at 20:30, ABIOLA ADEFILA adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, Has globalization worked for anyone? Thanks On Tue, Oct 6, 2009 at 11:55 PM, ABIOLA ADEFILA mailto:adefilabi...@gmail.comadefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, Am working on lab 3 question 2.4 After creating the globalization, when i call i see the full E.164, when the localization is applied i see 4 digits in displa as well as the received calls. Am using 7970 phones, which i guess is supported fr globalization Any one with any idea is what is wrong Thanks ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling Party Manipulation on globalized Outbound via MGCP Gateway
1. Depends on your methodology and what the questions are asking. Go through the dial-plan chapter in the SRND. Spend a lot of time there trying the things they talk about. I spent a week in that chapter reading, configuring then documenting in my notes and diagrams. You may be able to get away with route patterns. MIght be quicker to configure. You may find it easier / quicker to use translation patterns. You may NEED a translation pattern or two to deal with avoiding or forcing a calling / called transformation pattern. Too many scenarios to list. No one best method. 2. Sounds like translation pattern for your specific example, but see #1 above. I don't have all the details on all the other tasks. 3. Globalization / Localization on the PSTN phone? Will CCME (PSTN Phone) do that? Or are you talking about Placed calls directory on the calling IP Phone? Don't know if you can change that. There is some discussion in the SRND in a couple sections about placed calls directories and why the numbers show up there the way they do. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cisco Dave [ciscod...@live.com] Sent: Saturday, October 03, 2009 9:49 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Calling Party Manipulation on globalized Outbound via MGCP Gateway I have a few questions about manipulating CLID and number type. 1) What is the best method for setting called party local calls to subscriber? Transformation Pattern, Called number Transformation Pattern (DP or GW), or Route Pattern? 2) With the calling party ID set to 10 digits (ex. 3033441000) what method is best for setting the calling party for local calls to subscriber? The calling party transformation mask of 2XXX has been set to national and applied to the GW. I would like to override this, or use a different method in order to set it to subscriber. 3) When making a local outbound call to the PSTN phone, with the calling party ID set to 10 digits, I would like to make a distinction between the displayed number on the phone and what is left in the directory. No matter what manipulation I use they both stay the same. Thank you, cd Hotmail: Trusted email with powerful SPAM protection. Sign up now.http://clk.atdmt.com/GBL/go/177141665/direct/01/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] just wanted to let you all know
Hi Brett, I will have to go back to my Globalization lab and try it again. I took some notes, but not as detailed as your explanation, so I don't remember exactly what I saw in all instances. Seemed whatever number showed up in the popup box while the CUPC softphone was ringing got stored in the missed / received calls history. I'm pretty sure I manipulated and think I remember the popup and the call lists displayed the same number. Pretty sure the deskphone mode seemed to always show E164 format due to the globalization at the gateway. Now I'm curious if I can change that. My globalization lab was to study the topic more and to see how viable this is for customers. To see how much I can save on route patterns, route lists, etc as well as the new device / line approach to PT and CSS's. Definately learned a lot and you bring up good points. I'll give it a try and let you know. Going to finish up volume 2, lab 4 before doing anything else. Thanks From: Brett [brett.sal...@gmail.com] Sent: Saturday, October 03, 2009 10:38 PM To: Michael Ciarfello Cc: Mark Snow; OSL Group Subject: Re: [OSL | CCIE_Voice] just wanted to let you all know It's my understanding that the older gen phones simply do not support display of the + character so UCM strips it before handing the call to them. Regardless of where the logs were to be stored, the content is xml (data is data) in either case is it not? H. Anywho, re: CUPC, it supports + dialing as well as globalization per my testing to date. Depending upon which mode you are in, as you noted, what displays upon ring-in may vary as the CNG Xforms at the Device level will, or seem to be, in effect with respect to device control. For example, I globalize all CNG numbers inbound so they are stored in the logs as such; however, using the CNG Xforms I can adjust what I call the Secondary CNG (the phone display) and localize it, prepend access code so it 'appears' like the actual digits I would dial from the phone given my location, etc. This is only realized on CUPC if in Softphone mode as the CNG Party Transformation at the Device level plays in its favor. In Deskphone mode, it weighs heavier for the hard phone and affects the Secondary CNG (phone display) there, but not on CUPC. Regardless of Secondary display characteristics, the CUPC histories pane retains the number in its Globalized form in this scenario and I can dial from there directly. Application Dial Rules can come into play depending on how the calls are logged in CUPC as well, but ADRs are not 'location aware', so I find it easier to Globalize to the call histories and deal with + dialing outbound per dialing region. Were your findings similar in nature? -b On Sat, Oct 3, 2009 at 1:39 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Can you (or anyone else) verify if CUPC suffers the same fate? Seemed to in my testing. Just want to make sure I am correct. Interesting that there is a difference between softphone mode and deskphone mode. SO I would watch out for that!!! From: Mark Snow [ms...@ipexpert.commailto:ms...@ipexpert.com] Sent: Thursday, October 01, 2009 2:23 PM To: Michael Ciarfello Cc: Peter Slow; OSL Group Subject: Re: [OSL | CCIE_Voice] just wanted to let you all know Neither CIPC, IPBlue (even as a 7961), nor any Gen 2 or older phone (7960, 7940, etc) support Globalization for Call History lists. This is due to the fact that all of these devices store the Call History local on the phone (or computer in case of CIPC/IPBlue software) - rather than with the Gen 3 phones, all of the Call History lists are stored in the DB on the CUCM server - rather than on the local phone. And since the number is globalized on the CUCM, it is always preserved there as the globalized number. It is Localized *before* being handed off to the actual IP Phone, and therefore the devices that store that information locally - cannot inherently support globalization in any form. HTH, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.commailto:ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communitieshttp://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 1, 2009, at 9:50 AM, Michael Ciarfello wrote: While you're all playing with CIPC and engaging in bad touch can you see if localization works? Reports from two people so far that the number will show localized on the phone display, but not show the E164 number in the call lists (like the restriction on the 40/60
Re: [OSL | CCIE_Voice] MVA Issue
Seemed if I didn't have the h323 gateway voip bind command on one of the interfaces, I got the same symptoms. The lab (volume2, lab 3) was using all MGCP (I think) and didn't put the h323 bind command on. That seemed to make a difference. Other than that, what does ccapi show? And what is your design look like. What question, what lab? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder [narinder.ku...@uxcg.com.au] Sent: Saturday, October 03, 2009 11:51 PM To: OSL Group Subject: [OSL | CCIE_Voice] MVA Issue I am having some issues with MVA . When I dial in after keying in my pin number I can turn on or off mobility that is working fine but I can’t dial out any number I try dialling internal numbers i.e 3001, 5001 and well as PSTN number as soon as finish dialling the number after pressing # the call drops. I tried changing the inbound CSS in the service parameter to “ Remote destination profile + line calling search space” but no luck. Anyone have any pointers what I am doing wrong ? Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volume 2 Lab 2 Question 9.3 - SIP SRST?
Don't need it. I think a PG typo. According to the SRST SIP documentation it's for invoking an IVR type application, like BACD but not BACD. Have no idea where the app is. It's not built-in either. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.com] Sent: Saturday, October 03, 2009 11:45 AM To: OSL Group Subject: [OSL | CCIE_Voice] Volume 2 Lab 2 Question 9.3 - SIP SRST? SIP SRST is new to me, having trouble figuring out part of the syntax (the Proctor Guide doesn't really discuss it). On BR1, when configuring voice register pool 1, there's a line saying application sip.app, I'm assuming refering to a file on flash that needs to be invoked in an SRST scenario. Can someone tell me where I can get this app? -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Test Email
Didn't work. You are shooting blanks. lol From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of anil kumar [vccie2...@gmail.com] Sent: Sunday, October 04, 2009 2:28 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Test Email ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA Issue
Convert it back to MGCP and practice the hairpinning mode also. I got it to work by putting MVA on the MGCP router. Little tricky, but the SRND or FS guide gives great hints. Also practice it by putting MVA on a separate router that runs H323, maybe PSTN-WAN router if you have your own equipment. Not sure if you have access to it in the PLs. From: Kumar, Narinder [narinder.ku...@uxcg.com.au] Sent: Sunday, October 04, 2009 5:16 AM To: Aamir Panjwani; Michael Ciarfello; OSL Group Subject: RE: [OSL | CCIE_Voice] MVA Issue Just fixed it, had issues with one of the dialpeer , as this gateway was earlier MGCP and I converted into H323 for MVA. Thanks all for help. From: Aamir Panjwani [mailto:aamir.panjw...@ivision.com.au] Sent: Sunday, 4 October 2009 8:09 PM To: Kumar, Narinder; Michael Ciarfello; OSL Group Subject: RE: [OSL | CCIE_Voice] MVA Issue Make sure CSS (not rerouting CSS which is used for SNR) on the remote destination profile has visibility of internal DN’s partition as well as PSTN route pattern partitions From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Sunday, 4 October 2009 7:29 PM To: Michael Ciarfello; OSL Group Subject: Re: [OSL | CCIE_Voice] MVA Issue It is question 5.12 in Lab 5 volume 1. I have bind command and all Dial Peer configured, I will check the CCAPI output. From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Sunday, 4 October 2009 5:27 PM To: Kumar, Narinder; OSL Group Subject: RE: [OSL | CCIE_Voice] MVA Issue Seemed if I didn't have the h323 gateway voip bind command on one of the interfaces, I got the same symptoms. The lab (volume2, lab 3) was using all MGCP (I think) and didn't put the h323 bind command on. That seemed to make a difference. Other than that, what does ccapi show? And what is your design look like. What question, what lab? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder [narinder.ku...@uxcg.com.au] Sent: Saturday, October 03, 2009 11:51 PM To: OSL Group Subject: [OSL | CCIE_Voice] MVA Issue I am having some issues with MVA . When I dial in after keying in my pin number I can turn on or off mobility that is working fine but I can’t dial out any number I try dialling internal numbers i.e 3001, 5001 and well as PSTN number as soon as finish dialling the number after pressing # the call drops. I tried changing the inbound CSS in the service parameter to “ Remote destination profile + line calling search space” but no luck. Anyone have any pointers what I am doing wrong ? Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact UXC Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ CONFIDENTIALITY - The information contained
Re: [OSL | CCIE_Voice] Lab blueprint question
One (or more, I haven't done them all yet.) have BACD. (I'm pretty sure. The lab I'm working on seems to imply it's looking for BACD. I just quickly read the test and am working on other sections. I just put on my quick notes to prepare for BACD. Looks like they snuck in some implied AAR which I didn't pickup on with my initial read and I've had to waste time to go back and reconfigure or visit objects again.) For other topics not covered, I think anything is game. Run through the CCME Admin guide, SRND, etc and pickup other topics (MLPP, E911, voiceview, etc.) NO ONE ever talked about MLPP for instance in the v2 days. Doesn't mean it won't be on the test. Ipexpert doesn't seem to have call monitoring and recording. Security was said to be a testable topic by Ben Ng during the last Ask the Expert. I would run through at least once topics that SEEM to have a low probability of being on the exam and know how to quickly find the documentation. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.com] Sent: Sunday, October 04, 2009 11:05 AM To: OSL Group Subject: [OSL | CCIE_Voice] Lab blueprint question I'm looking through the mock labs to find topics that weren't covered in labs 1 2 (Presence and BACD, namely). I noticed a couple of things and need a reality check; - There's a problem in mock lab 5, it looks like it's combining lab 4 and 5 in the lab 5 pdf. Is it just me? (Other than that, looks like lab 5 has all six sections.) - The Cisco lab blueprint doesn't seem to mention BACD on CME and the mock labs don't have any BACD sections either. The only instance of BACD seems to be in the volume 1 material. Does this necessarily mean that it's not a covered topic in the lab? - In preparation for my upcoming boot camp, I've been working through mock labs and taking extensive notes so that I can work on speed and memorization in the last 3 1/2 months of preparation. I think mock labs 1,2 and 4 should cover all of the known topics (1, 2, and 3 don't cover Presence.) Are there any sections that cover CME integration with CUCM? I couldn't find any. -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls between hq and br2
What tech prefix did the CCM server register as? You have to strip off the tech prefix on CCM. Use a translation pattern or significant digits. There's a difference between a GATEWAY registering with a tech prefix, and putting the tech prefix in the dial-peer which becomes part of the dialed string. You have to do some H323 audio and video conferencing to really get a full understanding of tech prefixes. But we can fill in the gaps here. We are just using tricks to steer the calls around the gatekeeper routing mechanism with different tech prefixes. The Cisco documentation is a little advanced in the fact they they think you know about h323 already. You have to find an h323 concepts book or perhaps whatever study material they use for the Gatekeeper CCVP, etc exams. From: ABIOLA ADEFILA [adefilabi...@gmail.com] Sent: Sunday, October 04, 2009 3:05 PM To: Michael Ciarfello Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] calls between hq and br2 hello, i added the h323-gateway voip tech-prefix 1# command to the CME and was able to call from HQ to br2 but i can not call from BR2 TO hq Below is my gateway config gatekeeper zone local HQ ipexpert.comhttp://ipexpert.com 10.10.200.3 zone prefix HQ 1... gw-priority 10 gk_trunk_2 zone prefix HQ 1... gw-priority 9 gk_trunk_1 zone prefix HQ 3... gw-priority 10 BR2-RTR zone prefix HQ 5... gw-priority 10 gk_trunk_2 zone prefix HQ 5... gw-priority 9 gk_trunk_1 gw-type-prefix 1#* default-technology bandwidth total zone HQ 128 bandwidth session zone HQ 16 no shutdown On the CME interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id HQ ipaddr 10.10.200.3 1719 h323-gateway voip h323-id BR2-RTR h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.10.110.3 dial-peer voice 400 voip incoming called-number 3... dtmf-relay h245-alphanumeric ! dial-peer voice 900 voip destination-pattern [15]... session target ras incoming called-number . Thanks On Sat, Oct 3, 2009 at 9:33 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: OK, let's take one direction at a time. CCM to CCME. (5002 to 3002) It looks like from the GK output, your got your ARQ. Let's debug voip ccapi on the CCME router and see what we get. Does the phone ring on the other side? Where are you getting that message? A voice message from the annunciator? And your're getting that message when CCME is calling CCM? For your CCME to CCM call, do you have the proper significant digits configured on the gatewa or stripping off the tech prefix? (1#5002) Matched tech-prefix 1# From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ABIOLA ADEFILA [adefilabi...@gmail.commailto:adefilabi...@gmail.com] Sent: Saturday, October 03, 2009 1:23 PM To: OSL Group Subject: [OSL | CCIE_Voice] calls between hq and br2 Hello, i have configured gatekeeper and trunk to call between the hq (5002) and br2(3002) calls from either side will give you the message '' the person you are trying to call is not available'' below is the debug anyone with an idea? rgd Q-RTR#debug gateke HQ-RTR#debug gatekeeper main 10 HQ-RTR# HQ-RTR# Oct 3 18:29:47.027: ////GK/gk_process: got a TIMER event Oct 3 18:29:47.027: ////GK/gk_handle_timers Oct 3 18:29:47.027: ////GK/gk_handle_timers: managed timer expired 0x467BDFF8 HQ-RTR# Oct 3 18:29:52.787: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-RTR# Oct 3 18:29:58.567: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Oct 3 18:29:58.567: ////GK/gk_rassrv_arq: arqp=0x48FFA648,crv=0x4, answerCall=0 Oct 3 18:29:58.567: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Oct 3 18:29:58.567: //809CCBBF0400/809CCBBF0400/GK/gk_dns_query: No Name servers Oct 3 18:29:58.567: //809CCBBF0400/809CCBBF0400/GK/rassrv_get_addrinfo: (3001) Tech-prefix match failed. Oct 3 18:29:58.567: //809CCBBF0400/809CCBBF0400/GK/rassrv_get_addrinfo: (3001) Matched zone prefix 3 and remainder 001 Oct 3 18:29:58.567: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Oct 3 18:29:58.567: //809CCBBF0400/809CCBBF0400/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49005C44 Oct 3 18:29:58.567: //809CCBBF0400/809CCBBF0400/GK/rassrv_arq_select_viazone: matched zone is HQ, and z_invianamelen=0 Oct 3 18:2 HQ-RTR#9:58.567: //809CCBBF0400/809CCBBF0400/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49005C44 Oct 3 18:29:58.567: //809CCBBF0400/809CCBBF0400/GK
Re: [OSL | CCIE_Voice] meet me
What question are you working on? From: ABIOLA ADEFILA [adefilabi...@gmail.com] Sent: Sunday, October 04, 2009 3:10 PM To: Michael Ciarfello Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] meet me Hello, the meet me number is in partion --PT-meetme the br1 phone has in it CSS pt-phone and pt-meetme Created a CTI with DN forwarded all to in css-meetme( that has d partion pt-meetme) for other phones Neither br1 phone nor other phnes can call the , they get Number can not be reached Thanks On Sat, Oct 3, 2009 at 9:36 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: So the meet-me pattern is in a partition that only the person who is allowed to initiate the conference can reach? That person can't initiate the meet-me conference? The other people dial a different number than the meet-me (in a partition they can see) that matches a CTI RP that does call forward all to the meet-me number? From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ABIOLA ADEFILA [adefilabi...@gmail.commailto:adefilabi...@gmail.com] Sent: Saturday, October 03, 2009 11:46 AM To: Brett Hillman Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] meet me Hello, i did just that but it did not work Regards On Sat, Oct 3, 2009 at 4:23 PM, Brett Hillman bghill...@ventech.commailto:bghill...@ventech.com wrote: Did you initiate the conf? 1)Choose line (go off hook) 2) Choose meetme softkey 3) Dial meetme number Then other phone just dial meetme number - Original Message - From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Sent: Sat Oct 03 10:14:21 2009 Subject: CCIE_Voice Digest, Vol 44, Issue 11 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.commailto: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.commailto:ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.commailto: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: Question regarding MVA (Kumar, Narinder) 2. Meet me conference (ABIOLA ADEFILA) -- Message: 1 Date: Sun, 4 Oct 2009 01:06:13 +1000 From: Kumar, Narinder narinder.ku...@uxcg.com.aumailto:narinder.ku...@uxcg.com.au Subject: Re: [OSL | CCIE_Voice] Question regarding MVA To: Darren Manners dmann...@me.vccs.edumailto:dmann...@me.vccs.edu, ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Message-ID: 9e7dd48644dd594da5ff12ffa0d2dbe23a5cb02...@exmsyd01.aus.local Content-Type: text/plain; charset=us-ascii Darren, I didn't understand your topology, try to put on a piece of paper but didn't make sense, can you pls try to explain again? Thanks Narinder From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Darren Manners Sent: Wednesday, 30 September 2009 10:04 PM To: Darren Manners; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Question regarding MVA I guess the device pool option wont work eitherIt changes the ani of course, but does not change the history... Information Security Officer Mountain Empire Community College CCIE (SEC) # 18929, CISSP (#85782), CCSP, GIAC GCIA (#0849) GCIH (#1348) GCWN (#0467) CCVP, MCSE, ASE HP, CCA Tel: 276 523 2400 ext 226 Email: dmann...@me.vccs.edumailto:dmann...@me.vccs.edublocked::mailto:dmann...@me.vccs.edumailto:dmann...@me.vccs.edu www.mecc.eduhttp://www.mecc.edu/blocked::http://www.mecc.edu/ From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Darren Manners Sent: Tuesday, September 29, 2009 11:41 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Question regarding MVA At present Im testing Mobile Voice Access. I have a mgcp PSTN gateway so
Re: [OSL | CCIE_Voice] Lab blueprint question
Bingo. That was the one. And they intentionally made you determine if you need to use it or not. Customer's don't tell you I want to do this, please use BACD and blah blah to do it.. Great question. I'm pissed becasue I ran out of time and didn't have a chance to set the VM timer back to 90 seconds. I was using 20 seconds for testing and thought I was going to have enough time during review to set it back and test again. Oops. I stopped RIGHT at the 8:00 mark. Gotta be a harsh grader on myself. From: Nara Shikamaru [shikam...@kagadis.com] Sent: Sunday, October 04, 2009 9:21 PM To: Michael Ciarfello Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Lab blueprint question Someone just mentioned that BACD is on Vol 2 Lab 4 (which I'm getting ready to do), I must have missed it when I reviewed the mock labs this morning. I hope mock labs 6 - 10 more accurately reflect the material on the exam, not that I mind spending training time on 40 things that won't be on the lab. And, honestly, if it's not on the blueprint I don't know how someone can say that it might still be on the test. No mention of BACD (maybe it's called something else.) On Sun, Oct 4, 2009 at 5:26 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: One (or more, I haven't done them all yet.) have BACD. (I'm pretty sure. The lab I'm working on seems to imply it's looking for BACD. I just quickly read the test and am working on other sections. I just put on my quick notes to prepare for BACD. Looks like they snuck in some implied AAR which I didn't pickup on with my initial read and I've had to waste time to go back and reconfigure or visit objects again.) For other topics not covered, I think anything is game. Run through the CCME Admin guide, SRND, etc and pickup other topics (MLPP, E911, voiceview, etc.) NO ONE ever talked about MLPP for instance in the v2 days. Doesn't mean it won't be on the test. Ipexpert doesn't seem to have call monitoring and recording. Security was said to be a testable topic by Ben Ng during the last Ask the Expert. I would run through at least once topics that SEEM to have a low probability of being on the exam and know how to quickly find the documentation. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.commailto:shikam...@kagadis.com] Sent: Sunday, October 04, 2009 11:05 AM To: OSL Group Subject: [OSL | CCIE_Voice] Lab blueprint question I'm looking through the mock labs to find topics that weren't covered in labs 1 2 (Presence and BACD, namely). I noticed a couple of things and need a reality check; - There's a problem in mock lab 5, it looks like it's combining lab 4 and 5 in the lab 5 pdf. Is it just me? (Other than that, looks like lab 5 has all six sections.) - The Cisco lab blueprint doesn't seem to mention BACD on CME and the mock labs don't have any BACD sections either. The only instance of BACD seems to be in the volume 1 material. Does this necessarily mean that it's not a covered topic in the lab? - In preparation for my upcoming boot camp, I've been working through mock labs and taking extensive notes so that I can work on speed and memorization in the last 3 1/2 months of preparation. I think mock labs 1,2 and 4 should cover all of the known topics (1, 2, and 3 don't cover Presence.) Are there any sections that cover CME integration with CUCM? I couldn't find any. -- -Shikamaru -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
