Re: [OSL | CCIE_Voice] 7960 CUCM SIP Registration
Sip phones require an extension to register. Also make sure your replication is working... maybe your phone is programmed in the pub, but the sub doesn't have it in its database. Good luck. On Jul 17, 2013 9:35 PM, Alex Pishko alexpis...@yahoo.com wrote: All, Having an issue with a 7960 registering to CUCM using SIP. On the actual phone itself keep getting the error registration rejected. Within CUCM I also see rejected on the phone page. I've verifed multiple times that the MAC address is correct and my SIP profile is pretty basic as there is no security applied to the phone itself. This should be something that's really simple as I have other SIP phones registered to the cluster (not type 7960 though). If I register the same phone via SCCP it works fine. Also, have tried to convert from SCCP to SIP using the BAT tool; all with the same results. Has anyone else seen this or have some additional input into what may be happening? Thank you, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cue call failed in srst mode
I just ran into this today. You have jtapi triggers built in cue. They need to be there for ccm integration, but also make sure you set the ccn trigger under sip if using CLI configuration. If you are using web interface under the triggers, check the box for srst. Brian On Jul 7, 2013 2:34 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: Hi folks, My cue is working fine in normal mode. If i switch to srst mode it failed and busy tone. Dialpeer and codec g711 , sip-ua ip show correct. any idea what to tshoot and command to check? K -- * From: * sanity insanity networksanitytoinsan...@gmail.com; * To: * ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; * Subject: * [OSL | CCIE_Voice] CME srst best practices.. * Sent: * Sat, Jul 6, 2013 4:29:43 PM hi Guys, In srst I use the following config... telephony-service srst mode auto-provision all srst dn line-mode octo 1)Do I also need to configure srst dn template srst ephone template ? 2)what are best practices for setting up the cue in srst mode ? If possible include details of.. -mwi -Xfer to VM - button 6 of phone 1 is pressed -MJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] voicemail ip phone service
Experts,
Why does IPExpert seem to make a big deal about running the following
command out of the release notes to restore voicemail button functionality?
Voicemail
run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('ca69f2e4-d088-47f8-acb2-ceea6722272e','Voicemail','Voicemail','Voicemail','Application:Cisco/Voicemail',2,'t',1)
In my experience, you can simply and easily add the voicemail service back
in through the web gui. I just basically manually copy the settings out of
the corporate directory service, but make sure to replace the wording to
say Voicemail and make sure to check the Enterprise Parameters checkbox.
After doing this, I reset the devices and my voicemail buttons work again.
Am I missing something? Is there some advantage to doing this through
CLI? I'm not sure looking up the command in the release notes and then
pasting it into command line is any faster than adding it through the gui.
Thanks for any additional insight you might be able to provide.
Brian
___
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www.ipexpert.com
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Re: [OSL | CCIE_Voice] 3750 IOS version in lab
Experts, Does anyone know what version of software is tested on the 3750? Has Cisco publicly made remarks about what specific IOS version is in the lab exam? All I know is 12.2. Anyone know anything more specific than that? I have a hard time believing the answer is no. Thanks, Brian On Tue, Jun 18, 2013 at 7:30 PM, Brian Valentine bkvalent...@gmail.comwrote: Experts, Does anyone know what version of software is tested on the 3750? Has Cisco publicly made remarks about what specific IOS version is in the lab exam? All I know is 12.2. Anyone know anything more specific than that? Thanks, Brian On Tue, Jun 11, 2013 at 9:03 PM, Brian Valentine bkvalent...@gmail.comwrote: Bump On Mon, Jun 10, 2013 at 7:29 PM, Brian Valentine bkvalent...@gmail.comwrote: Unless it breaks NDA, can someone please tell me what version of IOS is running on the 3750 switch in the lab exam? The blueprint is generic in that it just says 12.2 Mainline. Thanks, Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] need help guys..
Perhaps this will help. http://www.onlinestudylist.com/archives/ccie_voice/2013-January/082684.html Best Wishes, Brian On Wed, Jun 26, 2013 at 11:46 AM, Amit Sharma aryan231...@gmail.com wrote: I am having proctor labs...' when i am working for cucm and uccx integration task.. it is already done by default for us in lab... but when check in cucm not able to see rmcm user that used in uccx ... and when i try to add ipcc extension in cucm end user...it is not having any option to add it... is it any config issue or i missed something that need to do for fix it? -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] 3750 IOS version in lab
Experts, Does anyone know what version of software is tested on the 3750? Has Cisco publicly made remarks about what specific IOS version is in the lab exam? All I know is 12.2. Anyone know anything more specific than that? Thanks, Brian On Tue, Jun 11, 2013 at 9:03 PM, Brian Valentine bkvalent...@gmail.comwrote: Bump On Mon, Jun 10, 2013 at 7:29 PM, Brian Valentine bkvalent...@gmail.comwrote: Unless it breaks NDA, can someone please tell me what version of IOS is running on the 3750 switch in the lab exam? The blueprint is generic in that it just says 12.2 Mainline. Thanks, Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] 3750 IOS version in lab
Bump On Mon, Jun 10, 2013 at 7:29 PM, Brian Valentine bkvalent...@gmail.comwrote: Unless it breaks NDA, can someone please tell me what version of IOS is running on the 3750 switch in the lab exam? The blueprint is generic in that it just says 12.2 Mainline. Thanks, Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] 3750 IOS version in lab
Unless it breaks NDA, can someone please tell me what version of IOS is running on the 3750 switch in the lab exam? The blueprint is generic in that it just says 12.2 Mainline. Thanks, Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration
As have I On Fri, May 31, 2013 at 11:09 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote: I signed as well. Thanks, Ramcharan Arya CCIE # 28926 ( RS) On Fri, May 31, 2013 at 9:09 PM, Ikenna Izugbokwe ikenna.izugbo...@gmail.com wrote: Done. Ikenna Izugbokwe Former - CCIE #36,472 (Voice) On Fri, May 31, 2013 at 8:40 PM, Tian id21...@gmail.com wrote: DONE… ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Edgar Feliz *Sent:* Friday, May 31, 2013 1:36 PM *To:* Suresh Bhandari *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration ** ** DONE...My wife also posted it on FB. ** ** On Fri, May 31, 2013 at 12:40 AM, Suresh Bhandari bring...@gmail.com wrote: *Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration *- Sign the Petition! For the interested candidates... Please join this campaign: http://chn.ge/17A0zXE I already did. Now its your turn. Initially shared by Martin Sloan (martinsloa...@gmail.com) -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- I ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CCM database password
All, I am building out my own lab to mimic the proctorlabs.com setup. I know it wouldn't be much work to replicate what's there, but I would prefer to do a database backup of the PL starting database and restore to my setup... particularly toward the end of one of my sessions so that I can continue studying on my own equipment after the session is over. Is the CUCM database password published somewhere? Thanks, Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] unity connection 8
sounds like you need to give the vm more resources. it needs 4 gigs of ram and 160 gigs of disk space, I believe. On May 21, 2012 9:04 AM, Ray jonha...@yahoo.com wrote: can anyone help!! or a link where i can see how to hack and get UC8 install on VM ESXi 4.1 I have a dell 1950 quadcore with 16g Ram and 1T of HDD, I installed vm esxi 4.1.0 on it when i used the cucm business EDition iso install uc8 ,, it say that product not supported on current hardware: cisco unity connection cisco unified business edition 5000 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] TPKT when CUPC (softphone) makes a call..
That would likely be your remote desktop session. Brian On Mar 25, 2012 11:51 AM, Baktha Muralidharan muralic...@gmail.com wrote: Hello folks In trying to understand the protocols involved. When calling from CUPC softphone to another instance of CUPC (in deskphone control mode), I see some messages/protocols that I could use help with- here is my call flow- CUPC (softphone)--CUPC (deskphone control) on my PC(10.10.0.102)on UCCX (10.10.210.5) 10.10.210.11 is Call manager sub. here are the relevant lines from wireshark- 111:26:01.15 10.10.0.102UDP 10.10.210.1146Source port: 50001 Destination port: sip 211:26:07.44 10.10.0.102SIP/SDP 10.10.210.11 962Request: INVITE sip:3002@10.10.210.11, with session description 311:26:07.52 10.10.210.11 SIP 10.10.0.102414Status: 100 Trying 411:26:07.53 10.10.210.11 SIP 10.10.0.102832Status: 180 Ringing 511:26:07.53 10.10.210.12 TCP 10.10.0.102 1314[TCP segment of a reassembled PDU] 611:26:07.53 10.10.210.12 SIP/XML 10.10.0.102949Request: NOTIFY sip:HQ2@10.10.0.102:50018;transport=TCP 711:26:07.54 10.10.0.102TCP 10.10.210.126650018 51919 [ACK] Seq=1 Ack=2132 Win=68 Len=0 TSval=61597619 TSecr=85867605 811:26:07.65 10.10.210.5TPKT 10.10.0.102104 Continuation 911:26:07.65 10.10.210.5TPKT 10.10.0.102 82 Continuation 10 11:26:07.65 10.10.210.5TPKT 10.10.0.102208 Continuation 11 11:26:07.65 10.10.210.5TPKT 10.10.0.102 97 Continuation 12 11:26:07.65 10.10.0.102TCP 10.10.210.5 5449659 ms-wbt-server [ACK] Seq=1 Ack=276 Win=269 Len=0 13 11:26:07.73 10.10.0.102SIP 10.10.210.12 539Status: 200 OK 14 11:26:07.80 10.10.210.12 TCP 10.10.0.102 6651919 50018 [ACK] Seq=2132 Ack=474 Win=2264 Len=0 TSval=85867875 TSecr=61597638 15 11:26:10.86 10.10.0.102T.12510.10.210.51101003 16 11:26:10.86 10.10.0.102T.12510.10.210.51101003 What are the TPKT packets from UCCX? They source port number for those packets is 3389. the destination port is an ephemeral port (49659). Port number 3389 is Microsoft Terminal Server (RDPhttp://en.wikipedia.org/wiki/Remote_Desktop_Protocol) officially registered as Windows Based Terminal (WBT). Thanks, /Baktha ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] NM-CUE
Check your mask on the voicemail pilot in cucm. On Mar 8, 2012 7:27 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BACD issue new labs #2
In dial-peer 222, you are calling up service aa. You don't have a service called aa. You named your service app-b-acd-aa. HTH On Sun, Feb 19, 2012 at 12:59 PM, Randall Crumm rrcr...@yahoo.com wrote: Hi I am trying to call x4000 and have BACD forward to ephone-hunt list. When i call 4001 it works. When I call 4000 I get a busy signal . Any thoughts are appreciated. application service app-b-acd-aa param voice-mail 4600 paramspace english index 1 param max-time-call-retry 700 param service-name app-b-acd param number-of-hunt-grps 1 param drop-through-option 1 paramspace english language en param handoff-string app-b-acd-aa param max-time-vm-retry 2 paramspace english location flash: param aa-pilot 4000 param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 ! service app-b-acd param queue-len 15 param aa-hunt1 4123 param queue-manager-debugs 1 param number-of-hunt-grps 1 dial-peer voice 222 voip service aa destination-pattern 4000 session target ipv4:10.10.110.3 incoming called-number 4000 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! ! ! telephony-service srst mode auto-provision all em logout 0:0 0:0 0:0 max-ephones 8 max-dn 8 ip source-address 10.10.202.1 port 2000 time-zone 21 time-format 24 max-conferences 8 gain -6 moh music-on-hold.au transfer-system full-consult create cnf-files version-stamp 7960 Feb 19 2012 22:18:09 ! ! ephone-dn 1 number 4001 label 4001 description +442077964001 name +442077964001 ! ! ephone 1 mac-address 0024.142E.76A9 button 1:1 ! ! ephone-hunt 1 longest-idle pilot 4123 list 4001 ! ! SiteC-RTR#show ephone reg ephone-1[0] Mac:0024.142E.76A9 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 IP:10.10.202.122 13500 7962 keepalive 64 max_line 6 button 1: dn 1 number 4001 CM Fallback CH1 IDLE Preferred Codec: g711ulaw Cheers, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] auto qos voip fr atm
Correct me if I'm wrong, but I believe you must enable FRTS. Brian On Feb 18, 2012 4:33 PM, chase mergenthal cm3_...@hotmail.com wrote: I just ran this on my SiteA and SiteB routers; and it broke everything... I removed the config and everything was fine, reapplied auto qos, and all is broke again... Is there anything missing? SiteA-RTR#sho frame-relay pvc 201 PVC Statistics for interface Serial0/0/1:0 (Frame Relay DTE) DLCI = 201, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0/1:0.1 input pkts 19output pkts 13 in bytes 1455 out bytes 1352 dropped pkts 0 in pkts dropped 0 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 3 out bcast bytes 1032 5 minute input rate 0 bits/sec, 1 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec pvc create time 00:00:31, last time pvc status changed 00:00:31 Bound to Virtual-Access2 (down, cloned from Virtual-Template199) cir 384000bc 3840 be 0 byte limit 480interval 10 mincir 384000byte increment 480 Adaptive Shaping none pkts 13bytes 1352 pkts delayed 0 bytes delayed 0 shaping inactive traffic shaping drops 0 Queueing strategy: fifo Output queue 0/40, 0 drop, 0 dequeued SiteA-RTR# class-map match-any AutoQoS-VoIP-Remark match protocol sip match protocol h323 match protocol skinny class-map match-any AutoQoS-VoIP-Control-UnTrust match protocol sip match protocol h323 match protocol skinny class-map match-any AutoQoS-VoIP-RTP-UnTrust match protocol rtp audio ! ! policy-map AutoQoS-Policy-UnTrust class AutoQoS-VoIP-RTP-UnTrust set dscp ef priority 42 compress header ip rtp class AutoQoS-VoIP-Control-UnTrust bandwidth percent 10 set dscp af31 class AutoQoS-VoIP-Remark set dscp default class class-default fair-queue interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template199 class AutoQoS-FR-Se0/0/1:0-201 auto qos voip fr-atm interface Virtual-Template199 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-UnTrust map-class frame-relay AutoQoS-FR-Se0/0/1:0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 -Chase -- If winners never quit and quitters never win, then who coined the phrase, Quit while you’re still ahead.? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] auto qos voip fr atm
I have had to do that as well for auto qos on an mlpofr config. Glad its working. On Feb 18, 2012 5:50 PM, chase mergenthal cm3_...@hotmail.com wrote: Sorry I didn't include that part of the config... After blowing it away a second time, rebooting and reapplying it on both sides, it works... I have no idea why.. interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi -Chase -- If winners never quit and quitters never win, then who coined the phrase, Quit while you’re still ahead.? -- Date: Sat, 18 Feb 2012 17:45:20 -0500 Subject: Re: [OSL | CCIE_Voice] auto qos voip fr atm From: bkvalent...@gmail.com To: cm3_...@hotmail.com CC: ccie_voice@onlinestudylist.com Correct me if I'm wrong, but I believe you must enable FRTS. Brian On Feb 18, 2012 4:33 PM, chase mergenthal cm3_...@hotmail.com wrote: I just ran this on my SiteA and SiteB routers; and it broke everything... I removed the config and everything was fine, reapplied auto qos, and all is broke again... Is there anything missing? SiteA-RTR#sho frame-relay pvc 201 PVC Statistics for interface Serial0/0/1:0 (Frame Relay DTE) DLCI = 201, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0/1:0.1 input pkts 19output pkts 13 in bytes 1455 out bytes 1352 dropped pkts 0 in pkts dropped 0 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 3 out bcast bytes 1032 5 minute input rate 0 bits/sec, 1 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec pvc create time 00:00:31, last time pvc status changed 00:00:31 Bound to Virtual-Access2 (down, cloned from Virtual-Template199) cir 384000bc 3840 be 0 byte limit 480interval 10 mincir 384000byte increment 480 Adaptive Shaping none pkts 13bytes 1352 pkts delayed 0 bytes delayed 0 shaping inactive traffic shaping drops 0 Queueing strategy: fifo Output queue 0/40, 0 drop, 0 dequeued SiteA-RTR# class-map match-any AutoQoS-VoIP-Remark match protocol sip match protocol h323 match protocol skinny class-map match-any AutoQoS-VoIP-Control-UnTrust match protocol sip match protocol h323 match protocol skinny class-map match-any AutoQoS-VoIP-RTP-UnTrust match protocol rtp audio ! ! policy-map AutoQoS-Policy-UnTrust class AutoQoS-VoIP-RTP-UnTrust set dscp ef priority 42 compress header ip rtp class AutoQoS-VoIP-Control-UnTrust bandwidth percent 10 set dscp af31 class AutoQoS-VoIP-Remark set dscp default class class-default fair-queue interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template199 class AutoQoS-FR-Se0/0/1:0-201 auto qos voip fr-atm interface Virtual-Template199 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-UnTrust map-class frame-relay AutoQoS-FR-Se0/0/1:0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 -Chase -- If winners never quit and quitters never win, then who coined the phrase, Quit while you’re still ahead.? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lab---...6 required
Don't feed the trolls. On Feb 6, 2012 9:28 AM, Rrcrumm rrcr...@yahoo.com wrote: So how do you k ow it was test six if you have taken it twice? Please give me a summary of test 1, test 2, test 3, and test 4 And how you would clearly identify each test Also how what is the cost for this lab 6 you are talking about? You have to know because you are asking people to split the cost. After you did the research to k ow that this new study guide recently came out Please provide details not just it is expensive and we know the difference between test one And two Rc R Sent from my iPhone On Feb 5, 2012, at 11:15 PM, Philip Mos mosphi...@yahoo.com wrote: Hi Randall, Bec it is really expensive and i need someone to share the cost with me :) as 3 friends got 6 now so really worried. Thanks -- *From:* Randall Crumm rrcr...@yahoo.com *To:* Wong Misk wongm...@yahoo.com; Methew Ch methe...@yahoo.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Sunday, February 5, 2012 8:21 PM *Subject:* Re: [OSL | CCIE_Voice] lab---...6 required Did you actually read the thread? One guys says he bought the lab, but can't crack it. First, where do you buy one lab? second, if you bought it why would you need to crack it This is BS Cheers, Randall -- *From:* Wong Misk wongm...@yahoo.com *To:* Methew Ch methe...@yahoo.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Sunday, February 5, 2012 3:06 AM *Subject:* Re: [OSL | CCIE_Voice] lab---...6 required Hi, I have got lab 6 in HK and there is lot of changes in it comparing to lab 5, 6 is more hard and it will take more time to clear. It was my 2nd attempt first i got 5 and now 6 i m really depress as also my friends got 6 last week. Now people should be ready for 6 best of luck Regards -- *From:* Methew Ch methe...@yahoo.com *To:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Sunday, February 5, 2012 1:35 PM *Subject:* [OSL | CCIE_Voice] lab---...6 required Hi, I am looking for the partner for purchase If anyone interested please email me the same. I just found the link if anyone is interested we can share the cost. h t t p:// w w w http://%20w%20w%20w/ . c e r t k n o w l e d g e .com/f o r u m/index.php?/topic/24-gb-ccievoicelabscom-real-labs/ Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com 5 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Locations CAC
Several ideas... 1) Check that the MTPs are registered to CUCM. 2) restart CCM service on all CUCM servers. 3) check that you made the RSVP policies mandatory. 4) check that enabled AAR. 5) try debug and show commands. 6) no sccp / sccp. On Sun, Feb 5, 2012 at 2:21 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Ok, after I re-issued the command ip rsvp bandwidth 136 on each subinterface, I was able to make a call go through. Now, I wanna reduce that to 36, so I can try my AAR. I issued the command ip rsvp bandwidth 36, but now the second call goes through. Is there anything I need to do prior to changing the amount of reserved bandwidth? I also went to Locations and Resynched the bandwidth. Still no luck. Emanuel Damasceno CCNP Voice On Sun, Feb 5, 2012 at 5:03 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hello Experts, I am trying to set up an AAR scenario for my studies. I configured 2 Locations, with unlimited bandwidth, but mandatory RSVP from HQ to BR2. I wanna use 5 concurrent calls, and I am also using g729 between sites. I added the MTP-HQ, and MTP-BR2 to CUCM, put them in a MRG, followed by MRGL, and referenced it in its respective device pool. Reset all the phones. So here is my config: HQ dspfarm profile 2 mtp codec g729r8 rsvp maximum sessions software 5 associate application SCCP interface Serial0/0/1:0.2 point-to-point description TO BR2 bandwidth 768 ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 class AutoQoS-FR-Se0/0/1:0-202 auto qos voip trust frame-relay ip rtp header-compression ip rsvp bandwidth 136 BR2 dspfarm profile 1 mtp codec g729r8 rsvp maximum sessions software 5 associate application SCCP interface Serial0/1/0:0.1 point-to-point description to HQ bandwidth 768 ip address 10.10.112.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 102 CISCO class AutoQoS-FR-Se0/1/0:0-102 auto qos voip frame-relay ip rtp header-compression ip rsvp bandwidth 136 The main problem is that on the FIRST call it already says Not Enough Bandwidth, wasn't that supposed to happen if the 6th caller tried to make a call? I already set to TRUE in Service Parameters for Automated Alternate Routing, but it's not showing the Not Enough Bandwidh, Rerouting message. I haven't configured my Partitions and CSSs yet, but what's up with the first call not going through? Am I missing something? Emanuel Damasceno CCNP Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Do Not Feed The Trolls
Remember, please do not feed the trolls. If someone claims to pass and doesn't present their ccie number, ignore them. It's a clear fishing scam. Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] vouchers
I have a few PL vouchers for sale. $25 or best offer. PM me for info. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Phone service button
Is the phone added under the control of the RMCM (rjtapi) user account? Brian On Sat, Dec 11, 2010 at 3:28 PM, Shrini linuxbos...@gmail.com wrote: Hi, I configured IPPA service and couple of phones subscribed to the service. CUCM PUB/SUB service URL are changed to IP address. IPPA serviceURL is using IP address and link is working perfectly ( I and able to open the link in browser) Still on Phone when I select IPPA service it keep on requesting and dies after sometime. Can someone please advice how to fix this , No DNS servers. T I A Shrini ___ 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] Does anyone know what Microsoft BPOS is?
http://lmgtfy.com/?q=Microsoft+BPOSl=1 On Thu, Sep 30, 2010 at 11:56 AM, gwen...@gmail.com gwen...@gmail.com wrote: Sent from my HTC on the Now Network from Sprint! - Reply message - From: CCIE Voice GMAIL givemeccievoice2...@gmail.com Date: Thu, Sep 30, 2010 11:05 am Subject: [OSL | CCIE_Voice] FW: CME and SIP Phone Presence To: apos;osl oslapos; ccie_voice@onlinestudylist.com I wouldn’t be so obvious if it was a real lab question, simply something I’m building on my own. Just a note, I am using entirely SIP, so SCCP wouldn’t matter and to assume that SCCP and SIP will behave the same way would seem troublesome to me. If someone has done CME presence with SIP Phones and comment on this I would appreciate it, otherwise I will assume that this is how SIP Phones will function with presence in CME. Jeff From: voice-gang voice-gang [mailto:mgcptroubleshoot...@gmail.com] Sent: Thursday, September 30, 2010 3:35 AM To: CCIE Voice GMAIL Subject: Re: [OSL | CCIE_Voice] CME and SIP Phone Presence Hi Jeff, No it should not work like that, Bec in SCCP phone when u pickup the heandset then the red light should glow of SCPH2 I just have one concern is this real lab question i am sure this is not ..!! :) Bec i have all the labs let me know if you want to be a study partner. Thks On Thu, Sep 30, 2010 at 4:12 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hey everyone, I am looking for some help on CME presence. I have configured presence and its working with my SIP Phones, but it only works when there is an active call. There is SCPH1 with extension 4001 and SCPH2 with extension 4002. Both are configured with BLFs to point to the other phone. When I pick up SCPH1, the BLF on SCHPH2 does nothing. However, if I make a call with SCHPH1, the BLF on SCHPH2 will turn red. Has anyone experienced this before? Is this the accurate way to function with SIP Phones? Here is the relevant configurations: voice register global mode cme source-address 10.5.202.1 port 5060 max-dn 10 max-pool 2 load 7945 SIP45.9-0-3S load 7942 SIP42.9-0-3S authenticate register authenticate realm cisco.com url directory http://10.5.202.1/localdirectory/ tftp-path flash: create profile sync 0063161211015195 voice register dn 1 number 4001 allow watch name Site C Phone 1 label 4001 ! voice register dn 2 number 4002 allow watch name Site C Phone 2 label 4002 ! voice register pool 1 id mac 0024.9733.6C28 type 7942 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser1 password cisco blf-speed-dial 1 4002 label SCPh2 4002 device ! voice register pool 2 id mac 0024.14B2.F542 type 7945 number 1 dn 2 presence call-list voice-class codec 1 username scuser2 password cisco blf-speed-dial 1 4001 label SCPh1 4001 device presence presence call-list sip-ua sip-server ipv4:10.5.202.1 presence enable Any help is appreciated! Jeff ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SCCP to SIP conversion on CME
Gig0/0.11 is 10.21.200.1? You might want to make that the source address for your tftp server. Looks like the LOAD statement is already there, but you need to serve the files via tftp. you also need an ntp server command under voice register global. Make sure the firmware files are served up using the tftp-server global config commands. I will typically look in the root of the flash drive for the SEPMAC.cnf or .cnf.xml files... delete any you find. Make sure you leave the defaults there.. just delete the ones with specific MAC addresses. Then under voice register global, offer the no create prof and then issue the create prof commands. See if you have any more SEPMAC.cnf or .cnf.xml files. If not, something is wrong in your config. You can debug tftp events while the phone reboots to watch and see what it is downloading for your tftp server. If you aren't getting anything when the phone boots, you might not have it pointed at the right IP address in your dhcp scope. Any time you change anything at the DN, Pool, or global levels, you should go to the voice register global and issue the same commands no create prof and then create prof before you restart your pools. SCCP phones don't require the create cnf-files every time, but sip phones do require the create profile to be issued with every change. Hope some of that helps. As an aside, you should also replace the dtmf-relay sip-notify command with dtmf-relay rtp-nte in the voice register pool. I don't think this is your problem with the phones registering as SCCP, but it will help save your hours more troubleshooting later. Brian On Tue, Sep 14, 2010 at 2:23 PM, groganhockey groganhoc...@gmail.com wrote: You need a LOAD statement under voice register global. mike On Tue, Sep 14, 2010 at 4:43 AM, linuxboss.9 linuxbos...@gmail.com wrote: I used below configuration to register 7961GE as SIP to CME but it is showing as SCCP registered. I have all the SIP firmware in root directory of flash. It should start downloading the SIP firmware but there is no action..there are no debug messages because the phone is already SCCP registered. Did switch port shut/no shut ..no change. Can anyone guide me where I am wrong. voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface GigabitEthernet0/0.11 bind media source-interface GigabitEthernet0/0.11 registrar server expires max 1200 min 300 voice register global mode cme source-address 10.21.200.1 port 5060 max-dn 10 max-pool 5 load 7961GE SIP41.8-5-4S authenticate register tftp-path flash: create profile sync 0005355132715547 voice register dn 1 number name Br2Ph2 label Br2 voice register pool 1 id mac 0AAA.F999.D562 type 7961GE number 1 dn 1 dtmf-relay sip-notify username br2ph2 password cisco codec g711ulaw R3#confi R3#configure t R3#configure terminal Enter configuration commands, one per line. End with CNTL/Z. R3(config)#voic R3(config)#voice re R3(config)#voice register poo R3(config)#voice register pool 1 R3(config-register-pool)#restart No contact info available for pool 1. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] problem with outbound call
Partition was enabled based on time of day was it? On Sun, Aug 22, 2010 at 9:07 PM, CCIE Voice cc...@corb.net wrote: I forgot to mention it but DNA did show that the digit string would route via the intended gateway and route pattern. -- On Aug 22, 2010, at 16:40, Ayman_labib ayman_la...@yahoo.com wrote: Next time try using DNA under Services. I had a similar problem and it was gateway wasn't properly registered. Sent from my iPhone On Aug 22, 2010, at 5:30 PM, Pavan pav.c...@gmail.com wrote: In such cases grabbing detailed SDL SDI traces would immensley help. Without them it is difficult to guess Sent from my phone On Aug 22, 2010, at 3:46 PM, CCIE Voice cc...@corb.net wrote: Tried with 1 sip phone and 1 sccp phone. Route pattern was set to route. Thanks for the ideas though. -- On Aug 22, 2010, at 14:13, bkvalent...@gmail.com bkvalent...@gmail.com wrote: Was the phone using SIP? - Reply message - From: CCIE Voice cc...@corb.net Date: Sun, Aug 22, 2010 3:49 pm Subject: [OSL | CCIE_Voice] problem with outbound call To: ccie_voice@onlinestudylist.com I have run into a strange problem that I can not figure out. Dialing digits on phone at BR2 (with what I can tell are correct CSS/partitions, gateway assignments) disconnect immediately after completing the dialing. e.g. Dialing 912123942123 call disconnects the moment that last digit is dialed. The call never hits the gateway. It is supposed to use MGCP gateway on BR1 router which appeared to be functional. I even converted this to h323 gateway and used a specific route pattern to force to that gateway...same problem. Reset, br1-rtr, reset gateway(s), reset CUCM (pub sub) all to no avail. I have run out of lab time and could not do debugs in rtmt to figure this out but was hoping someone else has experienced it and figured it out. tia...scd -- Steve Dickey ___ 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 ___ 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] Service URLs
Thanks for the help with these, Miron and Daniel. I was digging into the CAD install guide for the IPPA service, but the one-button login link is much faster because I don't need Acrobat to get into it, plus I can copy and paste it. Also, I've been getting the IPPM link from the CUPS deployment guide, which is now in wiki format. Not sure how that's handled in the real lab, so the CUCM SRND is much faster and I know I can rely on it being on my candidate desktop PC during the exam (or at least that's what Ben Ng said during the Ask the Expert). I didn't realize that the CUE command line shows the URL needed for Voice View. FYI, it is using the command show voiceview configuration. se-10-10-202-2# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do Enabled: Yes Idle Timeout (minutes): 5 Very nice. I wish all of these were that easy to look up. Brian On Sun, Aug 15, 2010 at 3:05 AM, Miron Kobelski findko...@gmail.com wrote: Hi Brian, these are the quickest methods to get those URLs that I am aware of. I can't check the locations exactly now, as I'm not in the lab, but you should be able to find them: 1) Extension Mobility CUCM Help search for extension mobility checklist 2) IPMA (IP Manager Assistant) CUCM Help search for ipma checklist 3) IPPA (IP Phone Agent) cisco.com UCCX support page configuration examples IPPA one-button login 4) IPPA - One touch login cisco.com UCCX support page configuration examples IPPA one-button login 5) IPPM (IP Phone Messenger) SRND (search PDF for IPPM) 6) VoiceView Express (CUE) go to CLI and run show voicemail voiceview (or similar) or go to GUI Voiceview configuration page (URLs are listed there) hth kobel On Sun, Aug 15, 2010 at 1:43 AM, Brian Valentine bkvalent...@gmail.com wrote: All, I've been trying to improve my speed in general... but specifically in looking up things that I might need in the lab exam. This evening, I've been working on reviewing where to find all the Service URLs. Most are too cryptic to memorize. So... assuming you don't have these memorized, where would you go to look up the following service URLs during the exam? BTW, I have my answers, but want to see what others say to compare with where I found these. Maybe you know of a quicker way to look up one or more of these. 1) Extension Mobility 2) IPMA (IP Manager Assistant) 3) IPPA (IP Phone Agent) 4) IPPA - One touch login 5) IPPM (IP Phone Messenger 6) VoiceView Express (CUE) Secondary question: Am I missing any? Are there any other IP Phone Services that would be fair game in the lab exam? The only other one I can think of off hand is the VoiceView for CUC, but that requires another server. Does anyone think it could be considered a testable topic? Brian Valentine ___ 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] uccx with cme
Since no one responded, I suppose no one has any information about this. That's a little disappointing. Brian On Thu, Aug 19, 2010 at 1:23 PM, Brian Valentine bkvalent...@gmail.com wrote: Bump On Tue, Aug 17, 2010 at 9:48 PM, Brian Valentine bkvalent...@gmail.com wrote: All, I'm curious if anyone knows if a UCCX to CME integration is a testable topic in the current blueprint. Do any of the Vol2 labs cover this? I went through them all once, and many of them twice. I don't recall a UCCX to CME integration lab. Without breaking NDA, does anyone know if this topic is fair game? I don't see any mention of this on the Ask The Expert forum. Did they discuss this topic at the CCIE Voice session at Cisco Networkers? I've done several UCCX deployments before and am fairly comfortable with it, but must admit that I'm not experienced with a UCCX to CME setup. Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] uccx with cme
Bump On Tue, Aug 17, 2010 at 9:48 PM, Brian Valentine bkvalent...@gmail.com wrote: All, I'm curious if anyone knows if a UCCX to CME integration is a testable topic in the current blueprint. Do any of the Vol2 labs cover this? I went through them all once, and many of them twice. I don't recall a UCCX to CME integration lab. Without breaking NDA, does anyone know if this topic is fair game? I don't see any mention of this on the Ask The Expert forum. Did they discuss this topic at the CCIE Voice session at Cisco Networkers? I've done several UCCX deployments before and am fairly comfortable with it, but must admit that I'm not experienced with a UCCX to CME setup. Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] gatekeeper max calls per endpoint
Working through lab 9. I configured the gatekeeper to allow only 1 call from HQ-GW. here is my config. gatekeeper zone local US ipexpert.com 10.10.110.1 zone local SP ipexpert.com zone prefix US 1... zone prefix US 212* gw-priority 10 HQ-GW zone prefix US 212* gw-priority 9 BR1-GW zone prefix SP 3... zone prefix US 5... gw-type-prefix 2#* default-technology gw-type-prefix 617* hopoff US gw ipaddr 10.10.110.2 1720 bandwidth interzone zone US 32 no shutdown endpoint max-calls h323id HQ-GW 1 Show gatekeeper endpoints shows the max calls as 1, but when it still allows me to place two calls... even shows it in my show gatekeeper endpoint output below... max = 1, current = 2 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.1 1720 10.10.110.1 64771 USH323-GW H323-ID: HQ-GW Voice Capacity Max.= 1 Avail.= 0 Current.= 2 Is this a bug? Seems like it should have blocked the call. Thanks, Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] uccx with cme
All, I'm curious if anyone knows if a UCCX to CME integration is a testable topic in the current blueprint. Do any of the Vol2 labs cover this? I went through them all once, and many of them twice. I don't recall a UCCX to CME integration lab. Without breaking NDA, does anyone know if this topic is fair game? I don't see any mention of this on the Ask The Expert forum. Did they discuss this topic at the CCIE Voice session at Cisco Networkers? I've done several UCCX deployments before and am fairly comfortable with it, but must admit that I'm not experienced with a UCCX to CME setup. Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] proctorlabs down?
Is proctorlabs.com down again? I can't get to it. I also can't do a remote DNS lookup on it. I am able to vpn in, but can't access the config webpage to start my lab or open a ticket for that matter. Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Can anyone access www.proctorlabs.com ???
No... I'm having the same issue. On Sat, Aug 14, 2010 at 9:35 AM, David Lee d16...@gmail.com wrote: Hello, Just wondering if it's just me. I'm trying from 2 different PCs and cannot access the webpage... Thanks, -Dave ___ 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] Can anyone access www.proctorlabs.com ???
Thanks for the heads up. I am able to connect now as well. On Sat, Aug 14, 2010 at 10:49 AM, David Lee d16...@gmail.com wrote: I just got connected now. -Dave On Sat, Aug 14, 2010 at 10:46 AM, Scott Newberry sc...@meganandscott.com wrote: FYI, got an email from Viking. Looking into it. Sent from my mobile phone. Please excuse my brevity and any spelling errors. On Aug 14, 2010 9:42 AM, David Lee d16...@gmail.com wrote: Does anyone remember the access server IP? The EZVPN is working, but the infrastructure is blank, so nothing is accessible... Tyson - not sure if you have some way to get hold of Proctor Labs support... Thanks. On Sat, Aug 14, 2010 at 9:38 AM, Brian Valentine bkvalent...@gmail.comwrote: No... I'm having the same issue. On Sat, Aug 14, 2010 at 9:35 AM, David Lee d16...@gmail.com wrote: Hello, Just wondering if it's just me. I'm trying from 2 different PCs and cannot access the webpage... Thanks, -Dave ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Service URLs
All, I've been trying to improve my speed in general... but specifically in looking up things that I might need in the lab exam. This evening, I've been working on reviewing where to find all the Service URLs. Most are too cryptic to memorize. So... assuming you don't have these memorized, where would you go to look up the following service URLs during the exam? BTW, I have my answers, but want to see what others say to compare with where I found these. Maybe you know of a quicker way to look up one or more of these. 1) Extension Mobility 2) IPMA (IP Manager Assistant) 3) IPPA (IP Phone Agent) 4) IPPA - One touch login 5) IPPM (IP Phone Messenger 6) VoiceView Express (CUE) Secondary question: Am I missing any? Are there any other IP Phone Services that would be fair game in the lab exam? The only other one I can think of off hand is the VoiceView for CUC, but that requires another server. Does anyone think it could be considered a testable topic? Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] mgcp timer receive-rtcp
I'm curious. I've been reading about the command mgcp timer receive-rtcp and that I need to remove it (via no mgcp timer receive-rtcp) when streaming multicast MOH from the flash of an MGCP gateway. I'm doing VOL1 lab1 today and task 6.1 has us do just that. I tried it from BR1, my MGCP gateway, and the command seems to be irrelevant. The call is not getting cleared, despite being at the default (mgcp timer receive-rtcp 5).What am I missing? See below the relevant configuration and show commands. Brian ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ccm-manager music-on-hold ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! mgcp profile default ! telephony-service srst mode auto-provision all max-ephones 10 max-dn 10 no-reg both ip source-address 10.10.201.1 port 2000 max-conferences 8 gain -6 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.10.201.0 10.10.110.2 BR1-RTR#show ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outid Interface === 239.1.1.1 16384 12898/12898 49 g711ulaw Lo0 BR1-RTR#show call active voice brief ... Telephony call-legs: 1 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 1 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 11EE : 50 21497860ms.1 +0 pid:0 Originate connecting dur 00:09:10 tx:83/13280 rx:88/14080 IP 239.1.1.1:16384 SRTP: off rtt:0ms pl:1600/0ms lost:0/1/0 delay:65/65/75ms g711ulaw TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a 11EE : 49 21497860ms.2 +0 pid:0 Originate active dur 00:09:10 tx:88/14784 rx:27524/4403840 Tele 0/0/0:23 (49) [0/0/0.2] tx:550470/550470/0ms g711ulaw noise:-60 acom:45 i/0:-50/-26 dBm ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location of QoS SRND from DocCD Page
During the Ask The Expert forum event several month ago, in response to the question what are exactly the documents we are able to access during the lab? Ben Ng, CCIE Voice content director wrote, We have four SRND documents ready to be opened, also you have the online Cisco document page. 1. UC 7 SRND 2. CUCME 7 SRND 3. UCCX 7 SRND 4. Enterprise QoS SRND 3.3 On Thu, Aug 5, 2010 at 10:06 PM, Amp amccar...@cciequest.com wrote: Ok Matthew, You got me stumped on this one. I have been trying to find it. I have heard from a few sources that the QoS SRND will be on the desktop of the computer that you are on in the lab so trying to find it via the products page may not be necessary; however knowing exactly where to find it is a good thing just in case. Amp Quoting Matthew Berry ciscovoiceg...@gmail.com: I know the QoS SRND is available through cisco.com/go/design, but I am trying to find it from the http://www.cisco.com/cisco/web/psa/default.html web page. Does anyone know the location of the document when accessed from the link above? Thanks, Matthew Berry ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Offering to ProctorLabs Clients
Thought about this today. Would the new topology support multicast across the VPN? If so, that might be well worth it. Brian On Fri, Jul 2, 2010 at 9:54 PM, Tyson Scott tsc...@ipexpert.com wrote: There will be no change to the current setup of what we support. Unfortunately the router must support more than 1 inside interface and you must have a 3550 or better switch to support what I am proposing (I should have mentioned this earlier). So not everyone will be able to support this topology. Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Managing Partner / Sr. Instructor - IPexpert, Inc. Mailto: tsc...@ipexpert.com Telephone: +1.810.326.1444, ext. 208 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Warren Heaviside (wheavisi) Sent: Friday, July 02, 2010 8:46 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Offering to ProctorLabs Clients Great idea Tyson, so long as it doesn't break or over complicate the homes based VPN configuration (871 in my case). Thanks, Warren Warren Heaviside wheav...@cisco.com ENGINEER.CUSTOMER SUPPORT Phone: +1 408 853 7995 Office Hour 9 am - 5 pm Pacific Monday - Friday For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, July 02, 2010 3:27 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 53, Issue 11 Send CCIE_Voice mailing list submissions to ccie_vo...@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Offering to ProctorLabs Clients (Tyson Scott) 2. lab5 vol2 transformation over sip trunk (amr thabt) 3. Re: Offering to ProctorLabs Clients (=?utf-8?B?Ymt2YWxlbnRpbmVAZ21haWwuY29t?=) 4. Re: Offering to ProctorLabs Clients (Tyson Scott) -- Message: 1 Date: Fri, 2 Jul 2010 17:21:47 -0400 From: Tyson Scott tsc...@ipexpert.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Offering to ProctorLabs Clients Message-ID: 005901cb1a2c$95013a80$bf03af...@com Content-Type: text/plain; charset=us-ascii Hey Voice Team, We recently made some design changes to our Voice bootcamp offerings to extend our PL infrastructure directly to the bootcamp phones. Meaning The Voice candidates in our bootcamps are able to test QoS, SRST, and other services offered at each branch site using the phones in the classes as if they were connected to the rack equipment. We can extend this same service to you as voice customers but I first wanted to check into the interest of you all to see if there is interest in the ability to do this. What it would provide is the phones that you have at your remote location, physical phones, would be able to appear as though they are directly connected to the PL devices. Now obviously this is going to add a lot of complexity that I will need to resolve so I want to first find if the interest is there. Please respond if interested. I will test this with a few students once we have the infrastructure in place to support this capability. It would be later next week. Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Managing Partner / Sr. Instructor - IPexpert, Inc. Mailto: tsc...@ipexpert.com Telephone: +1.810.326.1444, ext. 208 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at
[OSL | CCIE_Voice] error in PG
Page 15 of Proctor Guide for VOL 2 Lab 9 has an error on task3.3. The task asks us to block incoming international calls from reaching extension 1002. The PG offers the following answer: voice translation-rule 110 rule 1 reject // type international voice translation-profile BLOCK-INT translate calling 110 dial-peer voice 110 pots translation-profile incoming BLOCK-INT incoming called-number 1002 direct-inward-dial port 0/0/0:23 The problem is that you MUST use the keyword call-block in the dial-peer to block calls, otherwise it fails to block the call. So the dial-peer section in the PG should read: dial-peer voice 110 pots call-block translation-profile incoming BLOCK-INT incoming called-number 1002 direct-inward-dial port 0/0/0:23 Hope that helps someone. IPExpert, please fix the error in the PG. Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Called and Calling numbering type
Just make it do what the lab says. Otherwise, don't change it. On Jul 24, 2010 12:32 PM, cisco voip voip.ccieci...@gmail.com wrote: Hello Experts, i am really confused with what should be the called/calling number type in case of teho calls. My guess if HQ Phone doing teho thru BR1, called numbering type should be subscriber and calling number type should be national.. Need suggestions ___ 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] Called and Calling numbering type
You could. It depends. You earn points in the lab exam by accomplishing the tasks that they have specifically asked you to perform. There are no extra credit points for making things look pretty or consistent. So, don't waste your valuable time doing things that you were not asked to do. If the question says that EVERY time the call goes out BR1 to a particular destination, you should mark the calling number type as X, then, sure, make it X. If it doesn't say, then they don't care - not one bit. You never know - they might even ask you to do something strange like... set the calling party type as International on a local call and set it as Subscriber on the TEHO call. Why would they do that? Because they are not testing to see if you know and can perform best practices. They want to know that you can make it do what they have asked. So, just make it do what they ask. Get the points and then move on to another task. What I'm saying is... should be is whatever the exam asks you to make it. On Sat, Jul 24, 2010 at 12:40 PM, cisco voip voip.ccieci...@gmail.com wrote: I was planning to set calling number type using Calling party transformation pattern at one go... No?? On Sat, Jul 24, 2010 at 10:03 PM, Brian Valentine bkvalent...@gmail.com wrote: Just make it do what the lab says. Otherwise, don't change it. On Jul 24, 2010 12:32 PM, cisco voip voip.ccieci...@gmail.com wrote: Hello Experts, i am really confused with what should be the called/calling number type in case of teho calls. My guess if HQ Phone doing teho thru BR1, called numbering type should be subscriber and calling number type should be national.. Need suggestions ___ 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
[OSL | CCIE_Voice] Called Party display
I'm working on Vol2 Lab7 Task2.4. The task involves the following: HQ phone 2 dials 914158884343. Prefer to use TEHO to route the call out BR1. Local telco expects 7 digits. BR1 is an H323 gateway, so CUCM sends it 98884343. The gateway strips the 9 before sending to telco. Second choice gateway is the HQ gateway, which is MGCP. Local telco will expect 11 digits. CUCM would send the gateway 14158884343. Regardless of which gateway the call goes out the HQ Phone 2 display should say: To 4158884343. Got the call routing and redundancy down fine. That's works well enough. The problem is that no matter what I do, it seems to convert the display on HQ Phone 2 to match whatever digit manipulation was required by the egress gateway. The proctor guide says: The display on the Calling phone will be derived from the Route Pattern manipulation although the actual digits the UCM sends to the gateway is determined by the Route List/Route Group Called # transformations. So, I tried that. I tried doing all my digit manipulation on the RL details level and use the XX as the Called Party transformation on the Route Pattern level. Call goes through, but the HQ Phone 2 still displays To: 98884343. Next I tried setting the RL details to leave it as 415888 and used a Called Party Transformation Pattern at the gateway level to convert the call. I got the same result. Call succeeds. The display on HQ Phone 2 shows To: 98884343. What am I missing? Is this task possible? Thanks in advance, Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Called Party display
no supplementary-service h225-notify cid-update doesn't seem to help. I had an epiphany and figured out a work around to accomplish the task. What the PG suggests seems to work fine, but only on an MGCP gateway. I had to build an additional dial-peer in my BR1 gw with destination-pattern 415888 (forward digits 7). So, from CUCMs perspective, it sends the gateway 4158884343. If I do the manipulation on the H323 gateway, it works. HQ Phone 2 will basically send whatever CUCM sends an H323 gateway. Maybe there is a service param somewhere? Brian On Fri, Jul 23, 2010 at 2:15 PM, Matthew Berry ciscovoiceg...@gmail.com wrote: Voice service voip No supplementary-service h225-notify CID-update Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 23, 2010, at 12:31, Brian Valentine bkvalent...@gmail.com wrote: I'm working on Vol2 Lab7 Task2.4. The task involves the following: HQ phone 2 dials 914158884343. Prefer to use TEHO to route the call out BR1. Local telco expects 7 digits. BR1 is an H323 gateway, so CUCM sends it 98884343. The gateway strips the 9 before sending to telco. Second choice gateway is the HQ gateway, which is MGCP. Local telco will expect 11 digits. CUCM would send the gateway 14158884343. Regardless of which gateway the call goes out the HQ Phone 2 display should say: To 4158884343. Got the call routing and redundancy down fine. That's works well enough. The problem is that no matter what I do, it seems to convert the display on HQ Phone 2 to match whatever digit manipulation was required by the egress gateway. The proctor guide says: The display on the Calling phone will be derived from the Route Pattern manipulation although the actual digits the UCM sends to the gateway is determined by the Route List/Route Group Called # transformations. So, I tried that. I tried doing all my digit manipulation on the RL details level and use the XX as the Called Party transformation on the Route Pattern level. Call goes through, but the HQ Phone 2 still displays To: 98884343. Next I tried setting the RL details to leave it as 415888 and used a Called Party Transformation Pattern at the gateway level to convert the call. I got the same result. Call succeeds. The display on HQ Phone 2 shows To: 98884343. What am I missing? Is this task possible? Thanks in advance, 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
[OSL | CCIE_Voice] CUE CLI UserPrompt
I'm experimenting with the CLI on CUE. I'm trying to figure out how much I can do from the Command Line. So, If I'm asked to create a custom promptand let's say there was a requirement not to record over any existing greetings... Is it possible to figure out the filename of the new custom prompt from the CUE CLI? Thanks folks for any help you can give. Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] unable to upload CUE-CME license file - working on rack and need help ASAP
How about correcting the workbooks? ;-) It kinda stinks to struggle with this stuff while you are renting rack space. This got me this morning. Brian On Thu, Apr 29, 2010 at 11:01 PM, Vik Malhi vma...@ipexpert.com wrote: It was late when I set up the FTP server:-( if in doubt go cisco. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities at www.IPexpert.com/communities On Apr 29, 2010, at 5:01 PM, vccie2010 vccie2...@gmail.com wrote: You are the man, Vik :) I wish I had tried that :( but I was just going by your email link here... http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg13105.html On Thu, Apr 29, 2010 at 4:59 PM, Vik Malhi vma...@ipexpert.com wrote: Can you try cisco/cisco -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: vccie2010 vccie2...@gmail.com Date: Thu, 29 Apr 2010 16:56:46 -0700 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] unable to upload CUE-CME license file - working on rack and need help ASAP cue# $cue-vm-license_12mbx_cme_7.0.1.pkg username ipexpert pass cisco 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]yes Downloading ftp cue-vm-license_12mbx_cme_7.0.1.pkg Error: Download error Can not download cue-vm-license_12mbx_cme_7.0.1.pkg error code 530 : error type 'Access denied: 530' cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# cue# ping 10.10.210.5 PING 10.10.210.5 (10.10.210.5) 56(84) bytes of data. 64 bytes from 10.10.210.5 http://10.10.210.5 : icmp_seq=1 ttl=126 time=6.05 ms 64 bytes from 10.10.210.5 http://10.10.210.5 : icmp_seq=2 ttl=126 time=5.15 ms 64 bytes from 10.10.210.5 http://10.10.210.5 : icmp_seq=3 ttl=126 time=5.27 ms 64 bytes from 10.10.210.5 http://10.10.210.5 : icmp_seq=4 ttl=126 time=5.28 ms 64 bytes from 10.10.210.5 http://10.10.210.5 : icmp_seq=5 ttl=126 time=6.54 ms --- 10.10.210.5 ping statistics --- 5 packets transmitted, 5 received, 0% packet loss, time 24ms rtt min/avg/max/mdev = 5.159/5.663/6.542/0.541 ms, ipg/ewma 6.084/5.879 ms cue# cue# cue# cue# cue# cue# ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VATS in the VOD
Anyone else care to shed some light? Thanks, Brian On Mon, Jul 5, 2010 at 3:28 PM, Brian Valentine bkvalent...@gmail.com wrote: Vik, Well done on the VoD product. It's really very helpful. I was going through the WAN QoS video today. Question for you on VATS - should the fragment size be based on the adaptive rate or the cir? In the VoD you mention that the fragment size in the class should be 960, not 80. I understand that 80 was nonsense, but I was thinking it should be 480. I would appreciate it if you could clarify for me. See attached screenshot. Thanks in advance! Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE integration with CCM problems
What pt is the route point in? Your VM profile has a pilot and css associated to it. Does it contain the route point partition? Brian On Jun 9, 2010 10:02 PM, Pavan pav.c...@gmail.com wrote: No it does not have that partition. Shouldnt the css of the route point be used when the call is redirected to cti port? Sent from my phone On Jun 9, 2010, at 8:58 PM, bkvalent...@gmail.com bkvalent...@gmail.com wrote: Do your phon... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Disable Confrn Key on SIP Phone
have you done the: voice register global create profile and then rebooted the phones? You could also remove the conference softkey. On Tue, Jun 8, 2010 at 12:58 AM, Scott Newberry sc...@meganandscott.com wrote: Just curious as to whether anyone else has seen this... For the life of me, I cannot get the conference softkey to disable on my SIP CME phone. I've only tried this on a 7960, and am wondering if perhaps it's just an issue with the 7960, or maybe just a particular firmware version. Anybody else had this trouble? Seems like this should be easy... Of course, that's usually when stupid mistakes are made. voice register template 1 no conference enable voice register pool 1 template 1 Thanks! Scott http://ccie.meganandscott.com -- Blogging my way to my 8/16/2010 lab date ___ 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] Disable Confrn Key on SIP Phone
Found this previous thread in the archvies. Seems like the same issue. http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg14229.html Brian Valentine On Tue, Jun 8, 2010 at 12:30 PM, Scott Newberry sc...@meganandscott.com wrote: I have. In fact, I also tried removing the softkey, but to no avail. I'm going to try on a 7962 tonight and see if that makes any difference. On Tue, Jun 8, 2010 at 10:49 AM, Brian Valentine bkvalent...@gmail.com wrote: have you done the: voice register global create profile and then rebooted the phones? You could also remove the conference softkey. On Tue, Jun 8, 2010 at 12:58 AM, Scott Newberry sc...@meganandscott.com wrote: Just curious as to whether anyone else has seen this... For the life of me, I cannot get the conference softkey to disable on my SIP CME phone. I've only tried this on a 7960, and am wondering if perhaps it's just an issue with the 7960, or maybe just a particular firmware version. Anybody else had this trouble? Seems like this should be easy... Of course, that's usually when stupid mistakes are made. voice register template 1 no conference enable voice register pool 1 template 1 Thanks! Scott http://ccie.meganandscott.com -- Blogging my way to my 8/16/2010 lab date ___ 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] [SUSPECTED SPAM] RE: Lab and Language settings
Suble differences here, folks... NDA will stop me from answering the question Will languages/locales show up on my actual CCIE exam? I cannot answer that. Having said that, NDA does not stop me from answering the question Should I study languages/locales for the CCIE Voice exam? Here's my answer: You should be comfortable with locales. It's public information that the lab exam questions will be written in English. However, you should know how to change locales and the difference between user and network locales. It's easy. There's not a lot of complexity to it. Take a few minutes (up to 30 maybe) and learn how to do this. Usually that's what seperates experts from the rest - someone who took the time to really learn the technology. Frankly, if you have a locale question show up on your exam, you should consider yourself fortuntate. Those would be some easy points! Make sure you don't miss out because you didn't take 15 - 30 minutes to read about this and play with it once or twice. The NDA does NOT forbid study groups for preparation. It DOES forbid using the study groups to help you during your actual exam. Do NOT ask this group questions about the technology while you are sitting your actual lab exam. The NDA also does NOT forbid talking about the actual exam. It DOES forbit sharing exam content. For instance, if you sit the exam and find that a question confused you, don't ask about it in the forums. Even your score report is subject to NDA. As far as I understand it, you can discuss the actual exam in general terms without disclosing exam content. For that matter, during the Ask The Expert forum on NetPro Ben Ng answered public questions about the exam. He shared information about the exam without giving anything away. Relaying the information that Ben Ng disclosed (including Ben Ng's public comments at Cisco Networkers) or any other Cisco publicly published information, should not violate NDA. I consider all of that fair game. Brian Valentine On Mon, Jun 7, 2010 at 3:15 PM, Matthew Hall 1.matt.h...@gmail.com wrote: It's true this is a dangerous topic, it can all come down to one guys personal judgement at Cisco for all I know and your lab can be revoked and a life ban instated. There is no legal recourse as far as I know, just the end of your CCIE run. If someone point blank asks me for something on the lab, I just avoid answering the question, period. It is fine however to ask tech questions, like why show policy-map interface doesn't work on the 3560. I also answer things that Ben has publicly answered or are knowledge, everyone knows the blueprint on the lab and things like what IOS is used (Ben answered that one). The wording of the NDA is frightening to be honest though. The entire OSL is a violation if you read it strictly. The lines below would forbid the OSL. Disseminating actual exam content via web postings, discussion groups, chat rooms, study guides, etc. Giving or receiving assistance of any kind from anyone for any electronic or lab examination. What if someone sends you a some practice labs or asks you a question that you didn't KNOW was on the lab, you wouldn't know until it was too late. Not to mention that anyone who takes practice labs from vendors will know that there are similarities to the actual stuffheck there have to be, I mean there are only so many ways to setup an mgcp gateway. Where do you draw the line. Best to just be careful, use your best judgement. If it feels like cheating, it maybe. I think cisco is after the cheaters, not people trying to legitimately practice in group settings like we are here. On Jun 5, 2010, at 9:14 AM, wolfsrudel wrote: there's likely enough fundamentalism on the matter to adding other item on the list. let's be less radical and simply ignore such questions and/or comments. for sure, it would best not to forbid than to live and let live. On 6/5/10, Angel Perez gorr...@hotmail.com wrote: Your are right, NDA affects those candidates who have attempted the lab, anyway, please for these people under NDA don't answer any question regarding the lab http://www.cisco.com/web/learning/downloads/guest/learning/c644/ccmigration_09186a00803641d2.pdf http://www.cisco.com/web/learning/le3/ccie/exam/violation_rules.html Thanks Subject: [SUSPECTED SPAM] RE: [OSL | CCIE_Voice] Lab and Language settings From: r.ochi...@mfient.com To: gorr...@hotmail.com; jon1...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 22:33:43 +0300 It isn’t true that I cannot use the word lab….i can ask what the temperature is like in the lab, is the proctor in the lab, what is the lab topology like without necessarily breaking the NDA. You can ask anything, It’s upon me the person restricted by NDA to tell you that I cannot answer that as I’ll be breaking NDA I think NDA would apply to those who’ve attempted or passed the lab
Re: [OSL | CCIE_Voice] SIP TRUNK
You should try debug ccsip messages on the PSTN or CUBE router. It will show you the codec negotiation. On May 29, 2010 1:55 PM, Angel Perez gorr...@hotmail.com wrote: Hi: I have a sip trunk to my pstn router I'm trying to check the codec that the call is using but I can't this info at ucm traces or pstn gw debugs. I have try sip stack traces at ucm and also deb ccsip all at pstn, but I can't this info Any suggestion? -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ 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] Vol1 Lab 4A - X-Lite Issues
Steve, I'm hitting this tonight. Ever figure it out? Thanks, Brian On Thu, Feb 4, 2010 at 4:24 PM, Steve Denney (stdenney) stden...@cisco.com wrote: Hitting an interesting problem and just wondering if anyone else has seen similar symptoms... Working on Vol1, Lab 4A, Task 4.5. This is the task where you set up a SIP Route Pattern and use SIP URI dialing to dial the X-Lite CME SIP Phone (BR2 Ph 4, DN 3006) from the CIPC SIP Phone (HQ Ph2, DN 5002). When dialing from 5002 to 3006 (using the corporate directory on CIPC, as shown in the lab), the X-Lite rings, but hangs up immediately after the call is answered. The output of debug ccsip mess is attached. Looks like the X-Lite is sending a SIP BYE message with the description of Illegal Sdp Negotiation. I tried a call in the other direction as well – direct dial from 3006 to 5002. The CIPC rings, but you cannot actually answer the call. The debug in this case shows a “503 Service Unavailable” message, and the display on the X-Lite says Call failed: Service Unavailable. I’ve double and triple checked all configs (including allow-connections sip to sip), reloaded all routers, Googled for similar issues, and am now officially stumped. :) Debugs attached. Any ideas? cheers, steve ___ 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] UCCX Scripting
Roger, I think the simplest explanation of this is that 1) this exam is not a CCIE in UCCX Scripting. And 2) some scripting could be on the exam. I know that's obscure. That's because no one can say exactly what you will see in the lab. Let me also say that the IPExpert materials are sufficient in preparing you for the lab exam. I would be prepared to edit an existing script according to very specific requirements. I would not expect to have to create a very detailed script from scratch. If you did get something like that, I would think you may consider skipping those 3 or 4 points. Spending a couple of hours for 3 or 4 points wouldn't be worth it in my opinion. I would also be familiar with the built in scripts that come on the UCCX server out of the box. There is an AA script and an ICD script that could be used as a starting point in case you need to build something simple from scratch. This is my opinion, but I can't imagine you would see anything more time intensive than that on the exam. These are the things that IPExpert has you doing in their workbooks. Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Henderson Sent: Thursday, February 18, 2010 6:53 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX Scripting Hey Everyone, What is the best resource(s) to learn the various UCCX scripting methods needed for the lab? Does anyone have any good resources online? How complicated is it likely to get for the lab and how much time should we dedicate to this? Thanks, Roger No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.733 / Virus Database: 271.1.1/2693 - Release Date: 02/17/10 02:35:00 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Finally My Time CCIE #25772
Congrats James! Quite the accomplishment! -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of James Key Sent: Thursday, December 10, 2009 12:29 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Finally My Time CCIE #25772 Received the news yesterday that I passed my lab in San Jose! CCIE #25772. This was my 4th attempt, 3 for V2 and first V3 attempt. A big thanks to all of you guys on this list (and those who have come and gone with their numbers) and to IPexpert! I will try and put together my history preparing for this exam soon. James Key NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ 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] Attention IPexpert Members
Since this deadline has passed, should I assume that my account is the only one that still doesn't work? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Drew LePla Sent: Monday, December 07, 2009 2:54 PM To: ccie...@onlinestudylist.com; ccie_voice@onlinestudylist.com; ccie_secur...@onlinestudylist.com; ccie...@onlinestudylist.com Subject: [OSL | CCIE_Voice] Attention IPexpert Members Attention IPexpert Members, As most of you are aware, we launched a new website on Friday (December 4th). The new website is tightly integrated with our Salesforce account structure, therefore in order to better support our client base, we've changed the login method that needs to be followed when accessing your IPexpert Members Account. We are currently in the process of cutting over all accounts. This new solution will be effective and in place no later than Tuesday, December 8th, at noon EST. The process to login will be as follows: 1. Visit the www.ipexpert.com website, click on Client Login 2. On the left side of the page, you will find a Current Customers area with Email / Username and Password, enter your CURRENT username and password. 3. You will then be walked through an Account Migration process. Your FileOpen login and Members Login will be converted to your email address and your password of choice upon confirming your email address on file. If you have any issues or problems, please contact supp...@ipexpert.com or call at +1.810.326.1444. Regards, Drew LePla - COMP TIA A+, CCNA - IPexpert Lead Technical Support Engineer Mailto: dle...@ipexpert.com Telephone: +1.810.326.1444, ext. 204 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 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/communities and our public website at www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 7 - MoH
Jeff, The MOH server has a device pool, which means it also lives in a specific region.If your phones are in a different region than the MOH server, it will try to stream MOH in g.729 format (or whatever your inter-region codec is set to). But there is a service parameter that, by default, says only allow multicast MOH to stream in G.711 format. That’s why you are getting the beeps. When you place a call between two endpoints and then one party puts the other on hold, the MOH audio is not streamed between those two endpoints… It is streamed between the MOH server and the endpoint who is “holding”. You can eliminate most of the complexity of trying to figure out which codec will be used in various scenarios by doing the following: 1) Create a MOH Region that uses the G.711 codec for all intra-region and inter-region calls (or use whatever codec the lab tells you). 2) Create a MOH Device Pool and assign the MOH region to it. (I typically copy the HQ Device Pool and change the region to MOH.) 3) Assign your MOH servers to the MOH Device Pool. And reset the servers. 4) If necessary, set the service parameter to allow the codec that you want to use. You may want to restart the Cisco IP Voice Media Streaming App service. Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Monday, November 30, 2009 10:32 PM To: darylpsm...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 7 - MoH Daryl - I don't understand what you mean by assigning an MoH server to a reion I did some more testing by manipulating the codecs in the service parameters. Any combination of codecs that includes G729 provides MoH. Any combination that does not only plays beeps I'm going to run DNA to see if it can identify where the 729 is Jeff _ From: Daryl Smith darylpsm...@gmail.com To: Girard, Jeffrey COL MIL USA; OSL Group ccie_voice@onlinestudylist.com Sent: Mon Nov 30 18:08:14 2009 Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 7 - MoH Ok one more thing to check 1. Create another Region called Media and assing the MOH server to that region using G.711 codec 2. Reset the DP 3. Also configure the Voice Streaming Service to use G.711 and G.729 just for kicks DPS There are no secrets to success. It is the result of preparation, hard work, and learning from failure On 11/30/09 6:51 PM, Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil wrote: Daryl - Thanks for the response Yes on both I reset the device pool again with no changes Confirmed that ip multicast routing was enabled and that I had ip pim dense mode on vlan 20 and vlan 30 No change.still get beeps instead of MoH Jeff _ From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Girard, Jeffrey COL MIL USA; OSL Group ccie_voice@onlinestudylist.com Sent: Mon Nov 30 16:41:31 2009 Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 7 - MoH Have you reset the device pool once you completed the MRG and MRL setup? Did you enable the interface and the router for multicast Ip multicast-routing Int fa0/0.210 Ip pim dense-mode the server interface Int fa0/0.200 Ip pim dense-mode DPS There are no secrets to success. It is the result of preparation, hard work, and learning from failure On 11/30/09 4:39 PM, Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil wrote: Maybe I have spent too much time labbing and need to take a break - this should be simple but I can't get it to work I was not able to get the lab requirements to work, so I went back to basics...trying to get MoH working between the 2 HQ phones. I have my own lab equipment - not PL This is what I have done: Verified service is running and restarted both servers Set audilo source 1 as multicast Set pub and sub as servers, multicast enabling sub with increment on IP Created two new MRGs - one for pub and one for sub Added them both the MRGL_HQ Both HQ P1 and P2 have locations set to HQ, both use MRGL_HQ, both use HQ device pool. BW in HQ is unlimited. One phone is SIP other is SCCP When I place a call and press the hold on either phone, I get beeps instead of MoH. Norma$ly, this would indicate to me a codec mismatch - but both phones are in the same region/location/device pool. So how can I be getting a codec mismatch? Jeff _ ___ 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] CME Ephone-dn registration with GK
I disagree with this.. I believe the no-reg will keep your primary
extension or number from registering. which is all he has on this octo-line.
For instance, he could have a secondary (E164) number on that ephone-dn.
No-reg primary would keep the primary extension from registering, but the
secondary could register. In this case he doesn't have a secondary number
on his dn.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder
Sent: Thursday, November 12, 2009 6:00 AM
To: Ehab Salem; ccie_voice@onlinestudylist.com
Cc: Hussam Ahmad
Subject: Re: [OSL | CCIE_Voice] CME Ephone-dn registration with GK
hit ? after number 3001 no-reg, you will see more options.
When you are saying number 3001 no-reg that means do not register the
primary line. But in ur config you have octo-line.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ehab Salem
Sent: Thursday, 12 November 2009 9:40 PM
To: ccie_voice@onlinestudylist.com
Cc: Hussam Ahmad
Subject: [OSL | CCIE_Voice] CME Ephone-dn registration with GK
Hi All,
I've configured my CME BR2-RTR to register with the Gatekeeper, I need the
BR2-RTR not to register its ephone-dns, so this is the configuration on the
BR2-RTR:
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 PL ipaddr 10.10.110.1 1719
h323-gateway voip h323-id BR2-RTR
h323-gateway voip tech-prefix 3
h323-gateway voip bind srcaddr 10.10.110.3
!
telephony-service
no auto-reg-ephone
max-ephones 4
max-dn 5 no-reg
ip source-address 10.10.110.3 port 2000
auto assign 1 to 2
network-locale ES
network-locale 1 ES
network-locale 2 ES
network-locale 3 ES
network-locale 4 ES
time-zone 28
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Nov 12 2009 10:31:52
!
!
ephone-dn 1 octo-line
number 3001 no-reg
description 32143001
name BR2-Phone 1
!
!
ephone 1
no phone-ui speeddial-fastdial
no phone-ui snr
no multicast-moh
device-security-mode none
mac-address 001C.58F0.7548
max-calls-per-button 5
busy-trigger-per-button 3
type 7970
button 1:1
and this is the Gatekeeper Configuration:
gatekeeper
zone local PL cisco.com 10.10.110.1
zone prefix PL 1... gw-priority 10 gk-trunk_1
zone prefix PL 1... gw-priority 9 gk-trunk_2
zone prefix PL 1... gw-priority 0 BR2-RTR
zone prefix PL 5... gw-priority 10 gk-trunk_1
zone prefix PL 5... gw-priority 9 gk-trunk_2
zone prefix PL 5... gw-priority 0 BR2-RTR
no shutdown
but it still registering the ephone-dn with the gatekeeper:
HQ-RTR#sh gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags
--- - --- - - -
10.10.110.3 1820 10.10.110.3 62279 PLH323-GW
H323-ID: BR2-RTR
E164-ID: 3001
Voice Capacity Max.= Avail.= Current.= 0
HQ-RTR#debug h225 asn1
value RasMessage ::= registrationRequest :
{
requestSeqNum 145
protocolIdentifier { 0 0 8 2250 0 4 }
discoveryComplete TRUE
callSignalAddress
{
ipAddress :
{
ip '0A0A6E03'H
port 1820
}
}
rasAddress
{
ipAddress :
{
ip '0A0A6E03'H
port 62279
}
}
terminalType
{
vendor
{
vendor
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
productId '436973636F47617465776179'H
versionId '32'H
}
gateway
{
protocol
{
voice :
{
supportedPrefixes
{
{
prefix dialedDigits : 3
}
}
},h323 :
{
supportedPrefixes
{
}
}
}
}
mc FALSE
undefinedNode FALSE
}
terminalAlias
{
h323-ID : {BR2-RTR},
dialedDigits : 3001
}
gatekeeperIdentifier {PL}
endpointVendor
{
vendor
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
productId '436973636F47617465776179'H
versionId '32'H
}
timeToLive 60
keepAlive FALSE
willSupplyUUIEs FALSE
maintainConnection TRUE
[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
Re: [OSL | CCIE_Voice] terminal emulator used in lab.
Accidentally forgot to include the OSL in my reply -- sorry for the PM, Michael. When I sat the RS lab last year, I was glad to have had some experience with SecureCRT prior to sitting the actual exam. I was able to take advantage of some of the built in features to speed up my configuration. Simple things like copy and paste are different between various emulators. Its nice to know how to work the tools so that its second nature when I get in there. Brian On Thu, Nov 12, 2009 at 11:19 AM, Michael Ciarfello mciarfe...@iplogic.com wrote: 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] terminal emulator used in lab. (Brian Valentine)
You are correct. I never heard anyone complain that they failed the lab because of not being familiar with SecureCRT. I have heard that people failed because they ran out of time. I'm just looking for every advantage possible to increase my speed. 10 extra minutes at the end of the lab could mean the difference between passing or failing or at least between sanity and insanity for the next 24 hours while I wait to find out if I passed. To each his own I guess. When I took the RS lab, it was obviously a bigger deal because I was in SecureCRT all day long. There are a number of tweaks that SecureCRT will allow you to make that might make the lab a little easier. For example. You can: 1) adjust the color. Maybe making the text green and the background black will help you see things more clearly. 2) adjust the scroll buffer. I would like to be able to scroll back in the buffer further than the default. This could save time. How many lines will SecureCRT remember by default? What is the buffer limit of the application? I believe I can increase this. Anyone know how far? 3) make your own keystrokes that are most familiar to you. I prefer shift-insert for paste over right clicking. I don't like accidentally pasting configs into routers. Etc. etc. Getting hands on with this tool and knowing how to use it to my advantage in the lab seems like a good idea. That's all I'm trying to say. Brian -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jefferson Wilson Sent: Thursday, November 12, 2009 9:06 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] terminal emulator used in lab. (Brian Valentine) I concur with Michael on this. I thought my demo secure CRT for the home lab was different then the lab though. Are there 2 versions? The BLS video's mention to be practice with it though. Putty and CRT have different key stokes to cut and paste. I have not broken down and paid $100 for the CRT lic yet. I will eventually just to make my home studies a little more real. Not having Secure CRT experience though didn't cause me to fail on my first attempt. It was easy to use after the first couple of cut and pastes that day. Jefferson -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, November 12, 2009 12:00 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 45, Issue 84 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. terminal emulator used in lab. (Brian Valentine) 2. Re: terminal emulator used in lab. (Michael Ciarfello) 3. Re: terminal emulator used in lab. (Jonathan Charles) 4. Re: terminal emulator used in lab. (Daniel Rodriguez) 5. Re: terminal emulator used in lab. (Brian Valentine) -- Message: 1 Date: Thu, 12 Nov 2009 11:13:36 -0500 From: Brian Valentine bkvalent...@gmail.com Subject: [OSL | CCIE_Voice] terminal emulator used in lab. To: OSL Group ccie_voice@onlinestudylist.com Message-ID: 7f8374fe0911120813o153fd7fcr1997230a35941...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 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 -- Message: 2 Date: Thu, 12 Nov 2009 11:19:55 -0500 From: Michael Ciarfello mciarfe...@iplogic.com Subject: Re: [OSL | CCIE_Voice] terminal emulator used in lab. To: Brian Valentine bkvalent...@gmail.com, OSL Group ccie_voice@onlinestudylist.com Message-ID: 46458cd1692cd0448d1f56269c6a7e90110d4b7...@albs-exch01.iplogic.com Content-Type: text/plain; charset=us-ascii Didn't look at ANY version stuff. Didn't really even notice
[OSL | CCIE_Voice] vol2 lab4 task2.3
Is it just me or is this task impossible. The stipulation is that you are only allowed to use a single route pattern. You also have to keep the Class of Service requirements of task 1.1 and you also have a mandate that you must ensure that users do not experience interdigit timeout. Given all of these stipulations, I don't see how this task is acheivable. Can someone shed light? Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vol2 lab4 task2.3
Thanks Omar. I get the idea and you are absolutely correct - this is acheivable.Thanks for the insight. I was reluctant to check the PG, not wanting to spoil the rest of my lab. No worries about the 911/999 calls... they were under a different task. On Sat, Nov 7, 2009 at 11:57 AM, Omar Dahmani omar.dahm...@gmx.de wrote: This task should be achievable. Here are some ideas: - Single route pattern pointing to a route list / local route group for the globalized called numbers: \+! with urgent priority checked. - For Class of Service Requirements use the line device approach and translation patterns for blocking - Localized called numbers with the leading 9 or 0 should be globalized using translation patterns with a CSS including the single globalized route pattern. - To avoid interdigit timeout the translation patterns for local and long distance numbers should have urgent priority checked. - Translation Pattern for localized international numbers should NOT have urgent priority checked. To avoid interdigit timeout here use the \+!#. - ANI and DNIS transformation with Transformation Pattern on the gateways - For the Emergency numbers we still have to configure separate route patterns. However, those numbers can use a single route list. HTH -Omar *Von:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *Brian Valentine *Gesendet:* Samstag, 7. November 2009 15:34 *An:* OSL Group *Betreff:* [OSL | CCIE_Voice] vol2 lab4 task2.3 Is it just me or is this task impossible. The stipulation is that you are only allowed to use a single route pattern. You also have to keep the Class of Service requirements of task 1.1 and you also have a mandate that you must ensure that users do not experience interdigit timeout. Given all of these stipulations, I don't see how this task is acheivable. Can someone shed light? Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Registration problem with CUCME as SIP Phone
Try shutting down the switchports connecting the phones. Then go into voice register global and do no create profile. Then check flash to make sure you don't have any rogue SIP.cnf files. If you find any, delete them. While you are at it, go into telephony-service and do no create cnf-files. Then check flash and make sure there are no rogue SEP.cnf.xml files. If you find any delete them. Then go back to voice register global and do create profile. Also go back to telephony-service and do create cnf-files. Then do a no shut on the switchports connecting the phones. Keep in mind that some firmware versions are a one-way track. Once you upgrade to a certain point, you will not be permitted to downgrade. I know that the 1.3(3) version of the 7921/7925s are like that. You cannot go backwards once you are on that specific version. You might want to read the release notes of the firmware version your phones are currently at to see if this applies. HTH, Brian On Sat, Nov 7, 2009 at 1:22 PM, Gobind Singh Gill gob...@me.com wrote: Hi Omar, Thanks for the input. But I have already configured these commands but no luck :-S. --Gobind On Sat, Nov 7, 2009 at 7:11 PM, Omar Dahmani omar.dahm...@gmx.de wrote: Hi Gobind, something I often forgot to configure on a CME with SIP phones is the registrar server: voice service voip sip registrar server -Omar -Ursprüngliche Nachricht- Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Gobind Singh Gill Gesendet: Samstag, 7. November 2009 17:17 An: ccie_voice@onlinestudylist.com Betreff: Re: [OSL | CCIE_Voice] Registration problem with CUCME as SIP Phone Any idea guys? I am still struggling to get this phone registered as SIP. That didn't work..its still not registering and getting same output from debug tftp events. On Sat, Nov 7, 2009 at 5:24 AM, vccie2010 vccie2...@gmail.com wrote: First of all make sure you have? entry for followintg which is missingrestart the tftp and then see what happens... tftp-server flash:PHONE/7940-7960/P0S3-08-9-00.loads alias P0S3-08-6-00.sbn On Fri, Nov 6, 2009 at 7:45 PM, Gobind Singh Gill gob...@me.com wrote: Here's the output:- BR2-RTR# sh voice register tftp-bind tftp-server syncinfo.xml url flash:/syncinfo.xml tftp-server SIPDefault.cnf url flash:/SIPDefault.cnf tftp-server softkeyDefault_kpml.xml url flash:/softkeyDefault_kpml.xml tftp-server softkeyDefault.xml url flash:/softkeyDefault.xml tftp-server softkey1_kpml.xml url flash:/softkey1_kpml.xml tftp-server softkey1.xml url flash:/softkey1.xml tftp-server cme_dialplan_1.xml url flash:/cme_dialplan_1.xml tftp-server SIP001BD4C6C85E.cnf url flash:/SIP001BD4C6C85E.cnf tftp-server SEP001BD4C6C85E.cnf.xml url flash:/SEP001BD4C6C85E.cnf.xml BR2-RTR#sh flash | i -08 6 ? ? ? ? ?459 Jan 19 2009 18:24:20 P0S3-08-6-00.loads 7 ? ? ? 753560 Jan 19 2009 18:24:34 P0S3-08-6-00.sb2 8 ? ? ? ? ?444 Jan 14 2008 00:53:26 P0S3-08-10-00.loads 9 ? ? ? 756388 Jan 14 2008 00:54:18 P0S3-08-10-00.sb2 10 ? ? ?130952 Jan 14 2008 01:02:56 P0S3-08-10-00.sbn 13 ? ? ?129824 Jan 19 2009 18:25:16 P003-08-6-00.bin 14 ? ? ?130228 Jan 19 2009 18:25:30 P003-08-6-00.sbn 15 ? ? ?130548 Jan 14 2008 01:01:14 P003-08-10-00.bin 16 ? ? ?130952 Jan 19 2009 18:11:32 P003-08-10-00.sbn 137 ? ? 129824 Dec 18 2008 14:39:42 PHONE/7940-7960/P003-08-6-00.bin 138 ? ? 130228 Dec 18 2008 14:39:44 PHONE/7940-7960/P003-08-6-00.sbn 143 ? ? ? ?459 Dec 18 2008 14:39:56 PHONE/7940-7960/P0S3-08-6-00.loads 144 ? ? 753560 Dec 18 2008 14:40:06 PHONE/7940-7960/P0S3-08-6-00.sb2 On Sat, Nov 7, 2009 at 4:35 AM, vccie2010 vccie2...@gmail.com wrote: and also debug ccsip message and sh voice reg tftp-bind outputs On Fri, Nov 6, 2009 at 7:29 PM, vccie2010 vccie2...@gmail.com wrote: can you post sh flashand sh run On Fri, Nov 6, 2009 at 5:55 PM, Gobind Singh Gill gob...@me.com wrote: Hi Guys I am getting problem registering Proctor Lab's 7960 at BR2 as SIP Phone. It has SCCP firmware right now. This is what I have checked:- 1). It has the DHCP address :- *** BR2-RTR#sh cdp neigh f0/3/1 det - Device ID: SEP001BD4C6C85E Entry address(es): ?IP address: 10.10.202.53 Platform: Cisco IP Phone 7960, ?Capabilities: Host Interface: FastEthernet0/3/1, ?Port ID (outgoing port): Port 1 Holdtime : 172 sec Version : P00308000500 advertisement version: 2 Duplex: full Power drawn: 6.300 Watts *** 2). It has proper configuration for Voice Register Pool,DN and Global (Correct me if I am wrong):- *** voice register global ?mode cme ?source-address 10.10.202.1 port 5060 ?max-dn 2 ?max-pool 2 ?load 7960-7940 P0S3-08-6-00 ?authenticate register
Re: [OSL | CCIE_Voice] Registration problem with CUCME as SIP Phone
As I recall, I had a hard time getting the 8-6-00 files to work as well...Try the whole thing over again with the 08-10-00 files. I believe they will work better for you. Try this: no tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.loads alias P0S3-08-6-00.loads no tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.sb2 alias P0S3-08-6-00.sb2 no tftp-server flash:PHONE/7940-7960/P003-08-6-00.bin alias P003-08-6-00.bin no tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.sbn alias P0S3-08-6-00.sbn ! tftp-server flash:P0S3-08-10-00.loads tftp-server flash:P0S3-08-10-00.sb2 tftp-server flash:P0S3-08-10-00.sbn tftp-server flash:P003-08-10-00.bin tftp-server flash:P003-08-10-00.sbn ! voice register pool 1 username 3005 password cisco ! voice register global no create profile load 7960 P0S3-08-10-00 create profile ! inter fas0/3/0 shut inter fas0/3/1 shut ! inter fas0/3/0 no shut inter fas0/3/1 no shut end On Sat, Nov 7, 2009 at 3:39 PM, Gobind Singh Gill gob...@me.com wrote: Sorry Omar, guess I uploaded the old config file. I have attached the new latest running configuration with this email and I also have them in my flash:- BR2-RTR#sh flash | i -08 6 459 Jan 19 2009 18:24:20 P0S3-08-6-00.loads 7 753560 Jan 19 2009 18:24:34 P0S3-08-6-00.sb2 8 444 Jan 14 2008 00:53:26 P0S3-08-10-00.loads 9 756388 Jan 14 2008 00:54:18 P0S3-08-10-00.sb2 10 130952 Jan 14 2008 01:02:56 P0S3-08-10-00.sbn 13 129824 Jan 19 2009 18:25:16 P003-08-6-00.bin 14 130228 Jan 19 2009 18:25:30 P003-08-6-00.sbn 15 130548 Jan 14 2008 01:01:14 P003-08-10-00.bin 16 130952 Jan 19 2009 18:11:32 P003-08-10-00.sbn 137 129824 Dec 18 2008 14:39:42 PHONE/7940-7960/P003-08-6-00.bin 138 130228 Dec 18 2008 14:39:44 PHONE/7940-7960/P003-08-6-00.sbn 143 459 Dec 18 2008 14:39:56 PHONE/7940-7960/P0S3-08-6-00.loads 144 753560 Dec 18 2008 14:40:06 PHONE/7940-7960/P0S3-08-6-00.sb2 That's what I am trying to figure out that why I am getting that message if the files are there in the flash and I have proper bindings. I had these bindings since the very first email when I started this thread. You can see that I sent the output to vccie2010 in this very thread before, this is strange or I am missing something due to which the phone is not able to find the file, any idea? --Gobind On Sat, Nov 7, 2009 at 9:29 PM, Omar Dahmani omar.dahm...@gmx.de wrote: Hi Gobind, Looking at the debug output: Nov 7 20:31:36.300: TFTP: Looking for P0S3-08-6-00.loads Nov 7 20:31:37.152: TFTP: Looking for P0S3-08-6-00.sbn But there is no Opend flash:. this means that the CME cant't find the Files! In the running config, tftp-server commands are missing. Something like: tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.loads alias P0S3-08-6-00.Loads tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.sbn alias P0S3-08-6-00.sbn ..and so on HTH -Omar -Ursprüngliche Nachricht- Von: g...@gobind.net [mailto:g...@gobind.net] Im Auftrag von Gobind Singh Gill Gesendet: Samstag, 7. November 2009 20:45 An: Brian Valentine Cc: Omar Dahmani; ccie_voice@onlinestudylist.com Betreff: Re: [OSL | CCIE_Voice] Registration problem with CUCME as SIP Phone Hi Brian Thanks for the input. I have tried your steps but it didn't work. I have also attached sh run output in text file attached with this email. I'm still getting the same output:- Nov 7 20:31:16.916: TFTP: Looking for CTLSEP001BD4C6C85E.tlv Nov 7 20:31:16.984: TFTP: Looking for SEP001BD4C6C85E.cnf.xml Nov 7 20:31:16.988: TFTP: Opened flash:/SEP001BD4C6C85E.cnf.xml, fd 7, size 2670 for process 265 Nov 7 20:31:17.008: TFTP: Finished flash:/SEP001BD4C6C85E.cnf.xml, time 00:00:00 for process 265 Nov 7 20:31:36.208: TFTP: Looking for CTLSEP001BD4C6C85E.tlv Nov 7 20:31:36.232: TFTP: Looking for SEP001BD4C6C85E.cnf.xml Nov 7 20:31:36.232: TFTP: Opened flash:/SEP001BD4C6C85E.cnf.xml, fd 7, size 2670 for process 265 Nov 7 20:31:36.256: TFTP: Finished flash:/SEP001BD4C6C85E.cnf.xml, time 00:00:00 for process 265 Nov 7 20:31:36.300: TFTP: Looking for P0S3-08-6-00.loads Nov 7 20:31:37.152: TFTP: Looking for P0S3-08-6-00.sbn Nov 7 20:31:45.440: TFTP: Looking for CTLSEP001BD4C6C85E.tlv Nov 7 20:31:45.508: TFTP: Looking for SEP001BD4C6C85E.cnf.xml Nov 7 20:31:45.512: TFTP: Opened flash:/SEP001BD4C6C85E.cnf.xml, fd 7, size 2670 for process 265 Nov 7 20:31:45.528: TFTP: Finished flash:/SEP001BD4C6C85E.cnf.xml, time 00:00:00 for process 265 Nov 7 20:31:45.760: TFTP: Looking for English_United_States/7960-font.xml Nov 7 20:31:45.764: TFTP: Opened system:/its/united_states/7960-font.xml, fd 7, size 8777 for process 265 Nov 7 20:31:45.820: TFTP: Finished system:/its/united_states/7960-font.xml, time 00:00:00 for process 265 Nov 7 20:31:46.528: TFTP: Looking for English_United_States/SCCP-dictionary-ext.xml Nov 7 20:31:46.624: TFTP: Looking for English_United_States/SCCP
Re: [OSL | CCIE_Voice] FRF.12 help
Found it in the SRND. Thanks for pointing me in the right direction, Tal. In case others are interested, here's the formula: Fragment Size in Bytes = (PVC Speed in kbps * Maximum Allowed Jitter in ms) / 8 So, in our task, we have a 384kbps cir. Fragement size = (384 * 10) / 8 = 480 bytes. Brian On Sat, Nov 7, 2009 at 8:45 PM, Tal IPexpert t...@ipexpert.com wrote: Reference qos srnd, there is a chart for those values based on link On 11/7/09, Brian Valentine bkvalent...@gmail.com wrote: On VOL2 Lab 4 Task 5.1, the task has us configure FRF.12 LFI. The PG gives this as part of the solution: map-class frame-relay BR1 frame-relay fragment 480 service-policy output SHAPE-BR1 Question: How do we arrive at 480? Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Sent from my mobile device Regards, Talmadge Almand CCIE #20901 (voice) Sr, Support Engineer - IPexpert, Inc. URL:http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] V3 second attempt
Phil, Congrats! Quite the accomplishment! Brian -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Phil G Sent: Sunday, November 01, 2009 10:16 AM To: ccie OSL Subject: Re: [OSL | CCIE_Voice] V3 second attempt Here are my experience and strategy of journey to my CCIE: After passing the written exam last year in May, i wanted to schedule a lab and was very suprised that the soonest date was in January 2009. So i took my first lab on 14th of January. I started to learn in October 2008 with my lab at my office, which was: 8 7960, 3 2801, 1 2651XM, a 3750 as HQ-Switch, a 3550 as Branch-Switch, a VG248 and an ATA. Server Software was running on a DL380G4 with VMWare Workstation. Here the first advise: NEVER schedule a lab short time after Xmas. Your family may accept that they don't see you a lot of time during your training, but Xmas = no chance. :-) During my training for attempt #1 Cisco announced that the new V3-lab will start in July 2009. So with 8 months waiting time for a lab-seat, my first attempt in v2 would be my last attempt in v2. So all or nothing. I failed in January. I decided not to look every day if there my be a free seat, because i just wanted to keep my level of training that high with nearly no chance of a seat. So my first thought was, my project CCIE will go on for me in 2010. But then in May 2009 i purchased the BLS solution of IPX and decided to take a lab after the big change, so i scheduled a seat in August. I started to rebuild my office lab, i got some 7965s and 7962s and the NFR package and started to install the application-server on 2 DL380 G4 running VMWare ESXi. I kicked the 2651XM out and got a 2821 instead. So my lab was ready for V3. Then i took the first V3 attempt. I failed again, but this time it was OK, because now i don't had to wait 8 months again, it was easy to get a seat whenever i wanted. So i analysed my attempt, and took my second and successful v3-attempt on 22nd of October. The OEQ-part this time was harder for me than the first v3-attempt, but then the lab was pretty cool. The next day i got my number! Time strategy is one of the most crucial part of the lab. You need your OWN strategy. I have read a lot of postings and websites about time strategy. But my own time strategy was very unusual: I am not very fast on the keyboard and on IOS my best friend is ? and the TAB-key, even in the real lab. I practiced in my office lab the configurations, so that i was able to do 95% of all configuration without checking any documentation. I configured nothing in notepad, everything was configured directly on router-prompt, with help of ? and TAB. I configured the several sections as the appear in the workbook. Yes, you read correct. Nothing with: Touch every equipment only once, or sort by technology. I started with section 1 and ended with the last section. And believe it or not: time was never an issue in my lab attempts. At my last attempt i had 1,5 hours for troubleshooting and 1,5 hours left for testing/checking the sections. Training-sources: The common SRNDs, and for V3 the IPX-BLS, especially the VOL 2 Labs, and my first V3-attempt. Yes, folks, it is OK to fail, if you analyse your attempt inside out and learn what your weak topics are and what topics are OK. You must know everything what has not worked in your failed attempt and and you must learn in your lab how it is working. Don't make the same mistakes twice. HTH, Phil G Phil G wrote: Hi! I took my second V3-lab attempt on Thursday in Brussels (my third attempt in sum) and finally i got my number! As you can imagine i am very happy that i finally nailed it down. I want to thank everybody on this list, which has been a very use- and helpful resource during intensive lab-preparation. ___ 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
[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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Say it isn't so!
I got an email from “the other guys” yesterday welcoming Mark Snow to their team. http://ieoc.com/forums/p/8799/57626.aspx#57626 Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem in Corporate Directory
If you really wanted to emulate the way your PC locks out, you could use Extension Mobility to do this. Set the phone to have a base CSS (internal/911), if desired, and then require the user to be logged into the phone through EM -- giving their UDP greater calling privileges. Then set it to auto logout EM after say an hour of idle time. You can also set EM to remember the username of the person who logged in last to speed up the unlock process. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmed Khairy Sent: Saturday, October 24, 2009 2:13 PM To: asif raza Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem in Corporate Directory Importance: High I think it is a service can be added to the CUCM to lock the IP phones and I heard but I'm not sure that this option exist in CUCM 7 and also the option of powering off the IP phone. but I think that it can be secured from external access by CSS , FAC. Best Regards, Eng.Ahmed Mohamed Khairy Abd El Bary CCVP MCSE(Messaging). VOIP Telephony Administrator Children's Cancer Hospital (57357), Egypt Phone: +2 (02) 25351500 Ext:1144 Direct: +2 (02) 25351771 Mobile: +2 (011) 4987232 --- P Please do not print this email unless it is absolutely necessary. Spread environmental awareness. NOTICE: This email contains confidential or proprietary information that may be legally privileged. It is intended only for the named recipient(s). If an addressing or transmission error has misdirected the email, please notify the author by replying to this message. If you are not the named recipient, you are not authorized to use, disclose, distribute, copy, print, or rely on this email, and should immediately delete it from your computer system. _ From: asif raza [asifraz...@hotmail.com] Sent: Saturday, October 24, 2009 8:23 AM To: Ahmed Khairy Subject: RE: [OSL | CCIE_Voice] Problem in Corporate Directory Thanks a loot brother. It solved my problem. Since you are an IP telephony administrator, my job role is almost the same, that's why I want to ask one more thing that how do u secure your IP telephony setup from misusing, Or when somebody left his seat, other employees misuse his phone, how can we avoid this? Is their is any option in CUCM to lock the phone, (like we do in windows by pressing CTLR+ALT+DEL). Do you have any solution in your mind regarding this? Best Regards Asif Raza Network Engineer F u t u r e T e c h n o l o g y 311, Park Avenue Shahrah-e-Faisal, Karachi 75400 Ph: +92-21 4311908-9, Fax:4536571 Cell: +92 321 2916566 Email:asifraz...@hotmail.com _ From: ahmed.kha...@57357.com To: me_rashid...@yahoo.com; ccie_voice@onlinestudylist.com Date: Fri, 23 Oct 2009 14:40:56 +0200 Subject: Re: [OSL | CCIE_Voice] Problem in Corporate Directory You must put the Directory number in the user properties of each user from CUCM Administration Best Regards, Eng.Ahmed Mohamed Khairy Abd El Bary CCVP MCSE(Messaging). VOIP Telephony Administrator Children's Cancer Hospital (57357), Egypt Phone: +2 (02) 25351500 Ext:1144 Direct: +2 (02) 25351771 Mobile: +2 (011) 4987232 --- P Please do not print this email unless it is absolutely necessary. Spread environmental awareness. NOTICE: This email contains confidential or proprietary information that may be legally privileged. It is intended only for the named recipient(s). If an addressing or transmission error has misdirected the email, please notify the author by replying to this message. If you are not the named recipient, you are not authorized to use, disclose, distribute, copy, print, or rely on this email, and should immediately delete it from your computer system. _ From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rashid Khan [me_rashid...@yahoo.com] Sent: Friday, October 23, 2009 7:30 AM To: ccie voice Subject: [OSL | CCIE_Voice] Problem in Corporate Directory Hi Friends, I am having problem in Corporate Directory, that when I search for someone it does shows me the results. But with some names it doesn't show their extension number. It only shows their names, Creating lots of problem for me. Can anybody me help me in this regard Thanks in Anticipation n Best Regards Rashid Khan _ Windows 7: It helps you do more. Explore Windows 7. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WL MTAGL:ON:WL:en-US:WWL_WIN_evergreen3:102009 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] EZVPN Proctor labs
CCing the list for their benefit. If you use an 871 router, you would need to keep your DSL modem. I believe the 851 router can terminate DSL directly. You'd have to check with your ISP if they will support it as your modem. I am using three 7960s and two 7961s. I have a 3524-PWR-XL-EN switch between the 871 and my phone providing connectivity and Power over Ethernet to them. I have an 871 Router. The config was attached to the post I made to the study list yesterday. It has all of the configuration details you are looking for. If you need further support, you'll want to contact the support group at proctorlabs.com to help you get connected. Brian From: Muhammad Asif [mailto:asif.reh...@live.com] Sent: Tuesday, October 06, 2009 5:38 AM To: bkvalent...@gmail.com Subject: EZVPN Proctor labs Hi Brian, I was just reading your post on onlinestudy list and i am interested to know how did you setup this ezvpn ? What exactly do i need e.g what series/model of router and switch is required at least ? and how many IP Phones and their models ? I have a DSL connection at my home how can i terminate that on router do i need to remove the DSL Model and terminate it directly on router ? please let me know in detail. Regards, Asif _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Finally -- 871 EZVPN to Proctorlabs - WORKING!
All, I thought some might be interested. I finally got my 871 to work with Proctorlabs.com via EZVPN. This is what I did to finally get it working: 1) I used NEM. Whenever I tried using mode client, I could connect, but I had serious problems with phones not registering or staying registered. Also I couldn't get two way audio. My 871 would also crash every couple of hours and reboot. 2) I used only one subnet for my ip phones. Whenver I tried adding another subnet, the vpn connection would take errors and the connection would drop. So, keep it to one subnet. 3) I set the alternate TFTP server on all of my IP phones. 4) I set the ip mtu 1300 on my WAN interface. With the attached configuration, I had no problems with phones registering or staying registered. Also I had no problems with the VPN dropping. And finally, I had no problems with audio. Everything works well and sounds great. I hope this helps others. Brian Valentine IPexpertHome871.log Description: Binary data ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VOL1 Lab 8A - CCM 7960 SIP Softkeys
All, Is it possible to specify softkeys on a 7960 SIP phone in CUCM? If so, how? Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vpim
I got to thinking about it. There is a DNS server in the current lab blueprint. This means VPIM could be testable without them announcing a new piece of hardware like a Windows AD/DNS server. http://www.cisco.com/en/US/docs/ios/ipaddr/configuration/guide/iad_config_dn s_ps6350_TSD_Products_Configuration_Guide_Chapter.html#wp1063353 Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Wednesday, September 30, 2009 12:30 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com; Brett Hillman Subject: Re: [OSL | CCIE_Voice] vpim The only trend I've noticed is that Cisco's made the platforms accessible, not sure if VPIM's inclusion in CUWL licensing is a factor. For now, looks like VPIM will not be an issue - I can live with that. I just hope that if they change their mind that they'll accomodate. I'd like to not go back to the old days of tricking the CM, Unity, and IPCCx installs into thinking that Vmware was actually MCS. No thank you, I like this new process better. On Tue, Sep 29, 2009 at 9:23 PM, Michael Ciarfello mciarfe...@iplogic.com wrote: Agreed. But I think Cisco will continue to test it becasue CUWL licensing and most CUC licenses now come with VPIM. Same concept with IPCC. IPCC has been coming with CCM4.x as a 5-user license for free for years. (it's evolved a little. Don't want to bore with the complete history.) There is a reason Cisco did that and put it on the exam. In hopes that that 5-user (probably co-resident CCM/IPCC) becomes a larger CallCenter with more licenses. Not the same reasoning for VPIM though. Maybe TAC is seeing more and more voicemail system interoperability. Be thankful PIMGs are not on the exam. Yea, I am glad the 6500 is gone and the vg248 is gone. Although the vg248 was no big deal. Also yea, I've had to go through a lot of nonsense to get a lot of stuff for my lab. Borrowed a 61G-GE phone that needs AF power. Now I have to go find the proper power brick. Don't want to order to ebay. never know what you are going to get. Cube3 turns into cube2, etc. _ From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.com] Sent: Wednesday, September 30, 2009 12:14 AM To: Brett Hillman Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] vpim Frankly, the only thing I worry about is having to go through a lot of nonsense to gain a feature in my lab set up. It takes time and energy away from training. The smartest move Cisco made in their training cirriculum, by far, is opening the platforms up to be easily set up in a lab for training and testing. I'd hate for them to start testing on features that aren't included with the demo license. I suspect VPIM was included because it was part of the first set of mock labs that came out before the v3 testing started in July. We still have 5 more mock labs to go to round out the 10 total that are going to be available. The last 5 may be a more accurate reflection of the material needing specific attention. I'm going to use the last 3 months before my lab date focusing on mock labs 6 7 8 9 and 10. On Tue, Sep 29, 2009 at 8:21 PM, Brett Hillman bghill...@ventech.com wrote: I would not spend any time on this. The voice mail I saw was very basic. I would work hard on plus dialing and qos and srst. Would know mobility and snr basics. - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Tue Sep 29 22:14:25 2009 Subject: CCIE_Voice Digest, Vol 43, Issue 194 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Volume 2 Lab 2 Question 8.3 - VPIM and demo license? (Nara Shikamaru) 2. Re: vol 2 lab 3 : RSVP just won't work (Mike Thompson) 3. Re: vol 2 lab 3 : RSVP just won't work (Aamir Panjwani) -- Message: 1 Date: Tue, 29 Sep 2009 20:05:01 -0700 From: Nara Shikamaru shikam...@kagadis.com Subject: Re: [OSL | CCIE_Voice] Volume 2 Lab 2 Question 8.3 - VPIM and demolicense? To: Michael Ciarfello mciarfe...@iplogic.com, OSL Group ccie_voice@onlinestudylist.com Message-ID: a3c822920909292005i642b3b2epec4595d36b85a...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Frustrating.
Re: [OSL | CCIE_Voice] vpim
Actually, the router can be either a DNS client or a DNS server or both. That link points at the specific section that shows how to configure an IOS router to act as a DNS server. Don't know why anyone would want it to do this in the real world, but it is possible. VPIM is possible without them introducing any new hardware, announced or otherwise. Configuring the Router as a DNS Server Perform this task to configure the router as a DNS server. A Cisco IOS router can provide service to DNS clients, acting as both a caching name server and as an authoritative name server for its own local host table. When configured as a caching name server, the router relays DNS requests to other name servers that that resolve network names into network addresses. The caching name server caches information learned from other name servers so that it can answer requests quickly, without having to query other servers for each transaction. When configured as an authoritative name server for its own local host table, the router listens on port 53 for DNS queries and then answers DNS queries using the permanent and cached entries in its own host table. Brian From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Wednesday, September 30, 2009 11:17 AM To: Brian Valentine; 'Nara Shikamaru' Cc: ccie_voice@onlinestudylist.com; 'Brett Hillman' Subject: RE: [OSL | CCIE_Voice] vpim What's the below link for? The router can only be a DNS client. Correct. They can add an AD server or Linux DNS server at about anytime without announcing it. We just have the networkers statements which are an approximation. From: Brian Valentine [mailto:bkvalent...@gmail.com] Sent: Wednesday, September 30, 2009 7:13 AM To: 'Nara Shikamaru'; Michael Ciarfello Cc: ccie_voice@onlinestudylist.com; 'Brett Hillman' Subject: RE: [OSL | CCIE_Voice] vpim I got to thinking about it. There is a DNS server in the current lab blueprint. This means VPIM could be testable without them announcing a new piece of hardware like a Windows AD/DNS server. http://www.cisco.com/en/US/docs/ios/ipaddr/configuration/guide/iad_config_dn s_ps6350_TSD_Products_Configuration_Guide_Chapter.html#wp1063353 Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Wednesday, September 30, 2009 12:30 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com; Brett Hillman Subject: Re: [OSL | CCIE_Voice] vpim The only trend I've noticed is that Cisco's made the platforms accessible, not sure if VPIM's inclusion in CUWL licensing is a factor. For now, looks like VPIM will not be an issue - I can live with that. I just hope that if they change their mind that they'll accomodate. I'd like to not go back to the old days of tricking the CM, Unity, and IPCCx installs into thinking that Vmware was actually MCS. No thank you, I like this new process better. On Tue, Sep 29, 2009 at 9:23 PM, Michael Ciarfello mciarfe...@iplogic.com wrote: Agreed. But I think Cisco will continue to test it becasue CUWL licensing and most CUC licenses now come with VPIM. Same concept with IPCC. IPCC has been coming with CCM4.x as a 5-user license for free for years. (it's evolved a little. Don't want to bore with the complete history.) There is a reason Cisco did that and put it on the exam. In hopes that that 5-user (probably co-resident CCM/IPCC) becomes a larger CallCenter with more licenses. Not the same reasoning for VPIM though. Maybe TAC is seeing more and more voicemail system interoperability. Be thankful PIMGs are not on the exam. Yea, I am glad the 6500 is gone and the vg248 is gone. Although the vg248 was no big deal. Also yea, I've had to go through a lot of nonsense to get a lot of stuff for my lab. Borrowed a 61G-GE phone that needs AF power. Now I have to go find the proper power brick. Don't want to order to ebay. never know what you are going to get. Cube3 turns into cube2, etc. _ From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.com] Sent: Wednesday, September 30, 2009 12:14 AM To: Brett Hillman Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] vpim Frankly, the only thing I worry about is having to go through a lot of nonsense to gain a feature in my lab set up. It takes time and energy away from training. The smartest move Cisco made in their training cirriculum, by far, is opening the platforms up to be easily set up in a lab for training and testing. I'd hate for them to start testing on features that aren't included with the demo license. I suspect VPIM was included because it was part of the first set of mock labs that came out before the v3 testing started in July. We still have 5 more mock labs to go to round out the 10 total that are going to be available. The last 5 may be a more
Re: [OSL | CCIE_Voice] T1 CAS
Stupid question. Does Fas0/0 have an IP address on it (maybe it is router on a stick). Is fas 0/0 up and up? Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Sunday, September 27, 2009 10:40 AM To: James Key Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] T1 CAS I would recommend that, too, except that I've noticed that it tends to work when CUCM says it's registered but the isdn status say tei_assigned instead of multiple_frame_established. I ran into this yesterday in my lab again, gets me every time. I think that it can be avoided by doing the CUCM confguration before the gateway's. On Sun, Sep 27, 2009 at 6:56 AM, James Key j...@jackhenry.com wrote: Remove both MGCP bind commands, see if it comes up, then add back in. _ From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan [j.jho...@gmail.com] Sent: Sunday, September 27, 2009 8:41 AM To: Nara Shikamaru Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] T1 CAS the domanin name i removed. and yes the router host name matches too? On Sat, Sep 26, 2009 at 8:32 PM, Nara Shikamaru shikam...@kagadis.com wrote: Verify that the gateway's hostname.domain-name matches the Domain Name field on the router. On Sat, Sep 26, 2009 at 1:11 PM, J Hogan j.jho...@gmail.com wrote: Hello All I have a T1 configured and I keep getting registration rejected. I beleive I am doing everything right 1.) I restarted MGCPno mgcp, mgcp 2.) I only have 16 DSPs 2.) here is what i did thughts voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw card type t1 0 3 controller T1 0/3/0 controller T1 0/3/0 framing esf linecode b8zs cablelength short 110ft ds0-group 1 timeslots 1-10 type em-wink-start ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server 192.168.60.6 ccm-manager config ! mgcp mgcp call-agent 192.168.60.6 service-type mgcp version 0.1 mgcp rtp unreachable timeout 1000 action notify mgcp modem passthrough voip mode nse mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package no mgcp package-capability res-package no mgcp timer receive-rtcp mgcp sdp simple mgcp fax t38 inhibit mgcp rtp payload-type g726r16 static mgcp bind control source-interface FastEthernet0/0 mgcp bind media source-interface FastEthernet0/0 -- J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI Yahoo ID: jhogan552000 AIM ID: jhogan55 MSN ID: jhogan55 ICQ ID: 257599283 Live Life And Do Not Kill Time. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ -- -Shikamaru -- J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI Yahoo ID: jhogan552000 AIM ID: jhogan55 MSN ID: jhogan55 ICQ ID: 257599283 Live Life And Do Not Kill Time. NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab score report
All, I believe that for the RS labs, they are graded by proctors at another location, and therefore, another time-zone. So, sometime around midnight after I took my exam, I got the results from my RS lab attempt last year. I have not attempted my voice lab yet. It is scheduled for a few months out. But I have a question. Are the voice labs graded the same way? For those of you who have taken the voice lab before, how long was it until you got your score? The next business day? I suspect remote proctors are not able to grade voice labs since the person grading would probably need to pick up a handset and make test calls to grade the dialplan. Thanks for your feedback. Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VOD missing UCCX
Looking through the new VOD table of contents, I don't see a UCCX module. Curious. why is it missing? And is IPexpert planning to release one at some point in the future? Thanks, Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 7961 phones pre-provisioned BAT file?
Does anyone else see this or am I missing something? Brian On Sat, Sep 12, 2009 at 12:32 PM, Brian Valentine bkvalent...@gmail.com wrote: Vol 1 Lab 10A says: NOTE: If you are using your own Hardware Cisco 7961 Phones instead of 7962 phones, please perform the following: First delete the four 7962 phones that already exist in the DB. Next run the BAT tool for Phone Install. We have already pre-provisioned a file that you simply need to import (and change MAC address) containing the 7961 phone types. Where is that file? I don't see it loaded on the CUCM. It would save me a lot of time. Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] 7961 phones pre-provisioned BAT file?
Vol 1 Lab 10A says: NOTE: If you are using your own Hardware Cisco 7961 Phones instead of 7962 phones, please perform the following: First delete the four 7962 phones that already exist in the DB. Next run the BAT tool for Phone Install. We have already pre-provisioned a file that you simply need to import (and change MAC address) containing the 7961 phone types. Where is that file? I don't see it loaded on the CUCM. It would save me a lot of time. Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] New VoD's?
Wayne, Did these ship yet? Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson Sent: Saturday, September 05, 2009 7:34 PM To: Jason Granat Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] New VoD's? Jason - please (in addition to Rick's response) - these are being shipped out in the order they were received... Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert, Inc. Mailto: wlaw...@ipexpert.com Mobile: +1.810.334.1564 :: Message sent from iPhone. On Sep 5, 2009, at 7:09 PM, Jason Granat j...@slash128.com wrote: For the $25 shipping fee for a DVD I was hoping for USPS Overnight :-) Sent while mobile On Sep 5, 2009, at 15:34, Tanner Ezell tanner.ez...@gmail.com wrote: Well unless it was sent USPS, it is Saturday after all :) On Sat, Sep 5, 2009 at 3:20 PM, Jason Granat j...@slash128.com wrote: Anyone recieve them yet? I was told they would ship by 9/4 at the latest and we'd hopefully be recieving them on 9/5 Nothing showing up here yet... Sent while mobile http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ http://slash128.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 www.ipexpert.com
Re: [OSL | CCIE_Voice] Required quantity of Gen 2 phones
7961 model phones do support G722. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7961g_7961g-ge_7941g_79 41g-ge/6_0/english/administration/guide/7961cus.html#wp1032224 They do not support ILBC. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Tuesday, September 08, 2009 6:45 PM To: Mark Holloway Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Required quantity of Gen 2 phones I wouldn't worry about the G722 codec. I would just focus on 7961 phones for Globalization support among other things. -- 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 Sep 8, 2009, at 5:17 PM, Mark Holloway wrote: I think Vik mentioned once before you can provision/assign the HD codec to 7941/7961 phones but you can't answer the call unless you have the newer handset. I wonder if answering on speaker phone would work? I've seen the 7962 as low as $209, 7942 for $162, 7961 for $140, 7941 for $119 all refurbished. On Sep 8, 2009, at 2:06 PM, Thomas Koch wrote: Jeffery, This was a thread about 3-4 weeks ago from Mark Snow from IPExpert.. I'm in the same boat. I have (1) 7961 phone. I have 3 7960's... I'll try to send the original e-mail to you Unfortunately no. In fact the XLite SIP softphone can't do hardly anything that a hardware SIP phone can do (softkeys, etc). BTW - eBay typically has 7961's available for roughly 150USD. All you need is 3 of those and 2 7960's for our labs. Thomas J Kochb Owner/Consultant Digitones, LLC Cell: 630-808-4910 E-mail: digito...@comcast.net -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Tuesday, September 08, 2009 3:55 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Required quantity of Gen 2 phones I am prepping to begin studying with the Version 3 materials. I have been following the thread about Gen 1 vs Gen 2 phones. The lab that I built to study Version 2 has only 7960s. What is the min qty of Gen 2 phones that I need to get - 6? 2 for HQ, BR1, and BR2? --- 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Thomas J Koch (digitones @comcast.net).vcf___ 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 ___ 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] Required quantity of Gen 2 phones
Who says that you can't have a 2 way conversation? We are putting 7961s all over the place with default handsets and without disabling G.722. For an intra-region call between two 7961 phones, CallManager will indeed upgrade the G711 codec to G722 by default. Try making a G711 call between two 7961 phones and hitting ?? on the phone. You certainly can have two way audio and use G722 with a standard handset. You just won't get the benefit of hearing wideband audio because the speaker in that handset can't produce the sounds in the expanded parts of the audio spectrum. The phones will stream G722 between them and with the standard handset it will sound just like a G711 call. That's been my experience anyway. Brian From: Mark Holloway [mailto:m...@markholloway.com] Sent: Tuesday, September 08, 2009 7:51 PM To: Brian Valentine Cc: 'Mark Snow'; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Required quantity of Gen 2 phones The handset that comes with it doesn't support wideband and therefore you cannot have a 2 way conversation unless you upgrade the handset and/or headset. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps853 7/prod_white_paper0900aecd806fa57a.html Additionally, other newer Cisco Unified IP Phones 7900 Series phones (Cisco Unified IP Phone 7906G, 7911G, 7921, 7931G, 7941G-GE, 7961G, 7961G-GE, 7970G, and 7971G-GE models) support G.722 with an optional wideband handset or headset. On Sep 8, 2009, at 4:43 PM, Brian Valentine wrote: 7961 model phones do support G722. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7961g_7961g-ge_7941g_79 41g-ge/6_0/english/administration/guide/7961cus.html#wp1032224 They do not support ILBC. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Tuesday, September 08, 2009 6:45 PM To: Mark Holloway Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Required quantity of Gen 2 phones I wouldn't worry about the G722 codec. I would just focus on 7961 phones for Globalization support among other things. -- 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 Sep 8, 2009, at 5:17 PM, Mark Holloway wrote: I think Vik mentioned once before you can provision/assign the HD codec to 7941/7961 phones but you can't answer the call unless you have the newer handset. I wonder if answering on speaker phone would work? I've seen the 7962 as low as $209, 7942 for $162, 7961 for $140, 7941 for $119 all refurbished. On Sep 8, 2009, at 2:06 PM, Thomas Koch wrote: Jeffery, This was a thread about 3-4 weeks ago from Mark Snow from IPExpert.. I'm in the same boat. I have (1) 7961 phone. I have 3 7960's... I'll try to send the original e-mail to you Unfortunately no. In fact the XLite SIP softphone can't do hardly anything that a hardware SIP phone can do (softkeys, etc). BTW - eBay typically has 7961's available for roughly 150USD. All you need is 3 of those and 2 7960's for our labs. Thomas J Kochb Owner/Consultant Digitones, LLC Cell: 630-808-4910 E-mail: digito...@comcast.net -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Tuesday, September 08, 2009 3:55 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Required quantity of Gen 2 phones I am prepping to begin studying with the Version 3 materials. I have been following the thread about Gen 1 vs Gen 2 phones. The lab that I built to study Version 2 has only 7960s. What is the min qty of Gen 2 phones that I need to get - 6? 2 for HQ, BR1, and BR2? --- 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Thomas J Koch (digitones @comcast.net).vcf___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab
Re: [OSL | CCIE_Voice] SIP phone registration when using PLabs hardware only
IPExpert has a guide for how to do this. Look in your My ebooks section under: CCIE Voice Workbook v6.0 Volume 1 (CCIE v3 Blueprint) (by section) CCIE Voice: BLS Volume 1, Proctor Guide Appendix A-B - Updated Apr 27, 2009 This Appendix A-B guides you nicely through the process. As far as firmware, if a lab you are working on expects a certain firmware, it should be on the flash of that router. If not, use the web-based support once you are signed into proctorlabs. Someone should be able to assist you. Brian On Mon, Sep 7, 2009 at 8:52 AM, Sanjay Psp1...@yahoo.co.uk wrote: Hi Chaps, I'm using the PL labs setup exclusively to do the IPExpert voice v3 books via a hardware VPN connection . I've noticed in the workbooks that in the prerequisites at the beginning of the chapters e.g 3A that a 7961 is preferred for SIP. In a session a few days ago [pod 21], I noticed that there is no SIP firmware loaded/provided on the BR2 UCME router except for the 7940/60 and 7942/62. Question: how do i register my 7961 [already upgraded with the correct v8.x loader SIP by registering the phone to the callmanager] when there is no SIP firmware provided for the 7961 on the UCME. Options a) upload the 7961 SIP files on the router myself -[ does PL really want users loading firmware on to the flash (also any SIP phone software licence issue ?) ] b) must use a 7962 instead for the PLabs only route [ which i don't have ] ? c) I have 7960's but they don't support all the SIP functions. d) I tried registering the 7961 SIP phone from callmanager to the UCME direct but it failed to register in the end ... presumably I still had the load SIP41.8-5-2S under voice register global which points to other sip files needed for the phone loaded on the flash which currently are not there .also i had no tftp-server files pointing to the 7961 SIP files I've seen the Cisco/IPexpert documentation for SIP phone setup which seems straight forward if the all the firmware files are loaded onto the flash - thats no problem. Any pointers in the right direction would help - I'm sure i have missed something really obvious. warmest regards Sanjay ___ 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
[OSL | CCIE_Voice] VOL 1 - Task 5.20 - MVA no outbound option
After following the PG for task 5.20 in volume 1 workbook, when I call into the MVA access number (2123945999), I am greeted by the MVA recording. It asks me immediately for my PIN. Upon entering it, the system gives me options, starting with 2. There is no option 1 to make an outbound call. Am I missing something? Some service parameter somewhere? Thanks, Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Passed, thanks!
I could be wrong, but I believe that the CCIE has to work for you for at least one full year before the credential hits your company's profile officially. Brian -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Sunday, September 06, 2009 2:09 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Passed, thanks! Yes, but I can hire a CCIE, dump him after six months, he can bounce to another company and I have a year to replace him... technically, one CCIE could be credited for 3 companies... J On Sat, Sep 5, 2009 at 9:50 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: You are correct and that is probably still going on. But take this alternate perspective: Discount is the bigest reason, but the gap between the certification levels is not as great as it used to be. OIP program is the great protector of lesser certified companies from always getting beaten out by higher level partners. There are also VIP rebates, etc that cloud the profitability issue for higher certified partners even more. So those two programs (and others) help the lesser certified partner compete and be profitable. There is also the recently to be more stringent requirement of the CCIE MUST work for the company. No more long distance CCIEs or buying numbers, etc. The partner will lose their certification and the CCIE will lose their certification also. Heard of an example where a company with office on one side of the world had a CCIE associated with them that lived on the other side of the world. Heck of a commute. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Saturday, September 05, 2009 3:02 AM To: Nara Shikamaru Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Passed, thanks! One tiny problem. Cisco has placed requirements on their partners that without CCIEs, you can't attain any real status. So, companies will hire the useless CCIEs. And a lot of people believe it to be a meal ticket, get your CCIE, never want for work again Jonathan On Fri, Sep 4, 2009 at 6:16 PM, Nara Shikamarushikam...@kagadis.com wrote: No, he's right, it doesn't make someone an expert. No method of training can prepare people for the real world like . . . the real world! Experience is king. I work with an Engineer at Cisco who is, in fact, a CCIE . . . in RS. His specialty today is in IPCC Enterprise, he's not CCIE Voice and I can't think of anyone I would rather speak with when it comes to call centers. It's clear when we talk that he not only has a strong command of voice applications and call centers but he also understands the appropriate application of the technology when it comes to organizations. He gets it because he's done it. No education or training in the world can beat experience, Ivy League schools can't teach a person to be an effective professional, high schools and colleges can't prepare people for everything. Education and training is the best start to any worthwhile endeavor. But that's all it is; a start. On Thu, Sep 3, 2009 at 8:03 PM, Wayne Lawson groupst...@ipexpert.com wrote: Erwan - you don't think the CCIE is the expert of networking.are you nuts?Do you not understand the IT industry? Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert, Inc. Mailto: wlaw...@ipexpert.com Mobile: +1.810.334.1564 :: Message sent from iPhone. On Sep 3, 2009, at 10:45 PM, Erwan Erwan e_er...@yahoo.com wrote: Hi Jon, Congrats, definitely I understand your feeling , when u have to passed with hard work compare to those that cheat the lab. Just my opinion looking at the situation in my company on what we see about CCIE I do not really agree if CCIE cert is the expert/doctorate in networking, cause it more to config and troubleshoot for the cisco equipments. And i think that is the reason Cisco create it beside the marketing behind it :) And I meet lots out there with 20 years exp , even without CCNA , got the skills and knowledge beyond CCIEs, like understanding the protocol and work on multiplaform for voice. Sometimes those guy can solve the issue better than TAC cause they hv more comprehensive knowledge. Just opinion :) Thks, --- On Thu, 9/3/09, Jonathan Charles jonv...@gmail.com wrote: From: Jonathan Charles jonv...@gmail.com Subject: Re: [OSL | CCIE_Voice] Passed, thanks! To: jeremy co jeremy.coo...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Thursday, September 3, 2009, 2:29 PM The problem is that there are some integrators that actually do a technical interview... Some companies understand the CCIE is meaningless and ignore the certification. I do not. If you have a CCIE, then I need to CCIE
Re: [OSL | CCIE_Voice] Lab 11A Auto Attendant Issue
In the call handler, under greetings, you probably have the system prompt radio button selected. Try setting it to personal recording. Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Dave Wong Sent: Thursday, September 03, 2009 10:11 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 11A Auto Attendant Issue Hi I have a problem with question 11.3 under Unity Connections Call handling lab 11A of the v3 voice lab work volume 1. I've configured everything according to the proctor guide, but when i dial 5000 from any CUCM phone i hear the greeting Sorry AA is not available, recording your message at the tone.. . The call handler name I've given to the auto attendant in CUC7 is AA. Does anyone have an idea why this is the case? Summary of my configuration CUCM - CTI route point configured for the auto attendant service with DN 5000 configured to call forward all to Voicemail CUC7 - system call handler created with name of AA and extension 5000. The caller input keys of 1,2,3 are also configured according to question requirements. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] V3 Attempt Two
CONGRATS! 25309 is a round enough number, huh?! I bet you can live with that! That's quite the achievement! And to think, you left 30 minutes and a few points on the table and still passed. You are my hero! Best wishes in all your future endeavors. On Wed, Sep 2, 2009 at 10:28 AM, Jonathan Charlesjonv...@gmail.com wrote: OK, I got 25309 Weird, I passed who knew? Jonathan On Wed, Sep 2, 2009 at 2:13 AM, kapil atrishnice_cha...@yahoo.com wrote: I cracked mine on 3rd go in V2. Be consistent...good luck --- On Wed, 9/2/09, Ravindra Lakpriya lakpr...@gmail.com wrote: From: Ravindra Lakpriya lakpr...@gmail.com Subject: Re: [OSL | CCIE_Voice] V3 Attempt Two To: Tanner Ezell tanner.ez...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Wednesday, September 2, 2009, 11:00 AM Let's hope for the best man. U ll nail it. All the best dude. On Wednesday, September 2, 2009, Tanner Ezell tanner.ez...@gmail.com wrote: Good luck Jonathan, look forward to hearing the results! On Tue, Sep 1, 2009 at 9:53 PM, Jonathan Charles jonv...@gmail.com wrote: OK, took v3 again in RTP today... finished 30 minutes early... Well, not really... what happened was that I was doing some last minute tweaking (just retesting stuff, cleaning up some config) and some key huge point items stopped working... I undid what I did to break stuff, got up and walked away... yes, there was 30 minutes on the table, but it could have been the death of me... Anyway, waiting on results I would like to claim optimism, as I studied the crap out of my shortcomings last time, but I have done this before where I walked out of a lab pretty confident to see zero on sections I thought I aced... to be honest, I am like 85% sure I failed again. As they all say, the test is fair, nothing out of left field, some surprises on what was on there and what wasn't... there are some sleazy traps, but if you have a clue, you will work around em pretty quick... Took the first one in SJ, took this one in RTP... so, I can compare... In SJ, the phones are nailed to the walls in the cubicle... in RTP, they are on the desk (so you can flip em over and look at em...)... not sure which I prefer... I kinda like throwing them at the wall... But then again, in SJ, Ben Ng is sitting 4 feet from you, so, no intimidation there... I saw the remnants of the old v2 labs sitting in RTP, still had phones and fax machines... looked abandoned... Everything else I could say would be NDA... so, guys, do what you always do, look for the flurry of questions on 'how do I in this group or as veiled customer issues on Puck As a joke, here are the four questions I would ask: Why on this day are we limited on how we can dial? When on all other days we can dial however we want? Why on this day must we use frame-relay, when on all other days, all of our customers have MPLS? Why on this day are we running unpatched, basically beta-versions of CUCM, CUPS, CUCCX, when on other days we can install patches to get around bugs? Why on this day do I have to fly all the way to Raleigh and start the test at 7:15AM, when the guys who go to San Jose get to sleep in and take their test at 9:00? Jonathan If you are Jewish, those are funny. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Ravindra Lakpriya +94 773 532 094 ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VODs
IPexperts, For those of us who ordered the new and improved VODs, when will they ship? Thanks, Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VODs
Can someone from IPexpert please confirm? Are these shipping out on the 4th or 5th? Brian From: Dave Genton [mailto:dave.gen...@insight.com] Sent: Tuesday, September 01, 2009 9:07 AM To: 'cpar...@cparker.us'; 'bkvalent...@gmail.com' Cc: 'ccie_voice@onlinestudylist.com' Subject: Re: [OSL | CCIE_Voice] VODs They start on 4th, I talked to them, I should have on 5th they are saying Thanks, Dave Genton Architect - CCIE #6746 Insight Networking _ From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Brian Valentine bkvalent...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Tue Sep 01 06:03:54 2009 Subject: Re: [OSL | CCIE_Voice] VODs I hear they will ship after the 5th Original Message Subject: [OSL | CCIE_Voice] VODs From: Brian Valentine bkvalent...@gmail.com Date: Tue, September 01, 2009 4:49 am To: ccie_voice@onlinestudylist.com IPexperts, For those of us who ordered the new and improved VODs, when will they ship? Thanks, Brian _ ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] meet-me confrencing
Have you tried adjusting the regions/codecs? Try putting the phone and the CFB in the same region (or least set interregion calls to g711ulaw. See if meetme works then. If not, look more at the dialplan/partitions/calling search spaces. If meetme works with the phones in the same region, you would then know that you are dealing with a codec issue. Brian On Mon, Aug 31, 2009 at 5:19 PM, J Hoganj.jho...@gmail.com wrote: I have now added a transcoder and still the same issue. fast busy. is there a guide that walks through everything you need to get meet-me going? I can only assume I am leaving something out here? thanks On Sat, Aug 29, 2009 at 11:19 AM, Brian Valentine bkvalent...@gmail.com wrote: If the regions for your phone and your CFB are set up so that the phone would be calling the CFB using G729, the call will fail. Software CFB (the one that runs on the CUCM) does not support G729 calls. The only codec it supports is G711ulaw. You would either need to adjust the regions or invoke a transcoder. Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ilya Rubinchik Sent: Saturday, August 29, 2009 12:07 PM To: 'J Hogan' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] meet-me confrencing Hi I should press MeetMe software button instead of NewCall when creating confrnce -- Best regards, Ilya Rubinchik Chief UC Engineer Mars Solutions Ltd. 22, Munis str., Mirabad District Tashkent, 100080, Uzbekistan Tel UZ: +998 71 2907364 Fax UZ: +998 71 2907356 Mob UZ: +998 97 7128456 ICQ # 15508236 MSN: ilya.rubinc...@gmail.com Skype: im_citius E-mail: ilya.rubinc...@followmars.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan Sent: Saturday, August 29, 2009 7:51 PM To: Kumar, Narinder Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] meet-me confrencing Hello Everyone again :-) I have the CFB_2 registered. the IPVMS is activated, the CSS for the Meet-me is set to none. the media group on the phone has the group assigned to it that the CFB_2 belongs to. I hit new call -- meetme then I dial 2000. the pattern I have is 20[0-5]0 and I also have one as 1333 and everytime I get a busy signal thanks On Sat, Aug 29, 2009 at 9:26 AM, J Hogan j.jho...@gmail.com wrote: Team Thanks my software cfb is registered. But still no dice Sent from my iPhone On Aug 29, 2009, at 5:26 AM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: Hogan, As Aamir said make sure your software conf bridge is registered, also you need to initiate the meetme conference by pressing the meetme softkey and dialling the number, once the meetme conference is up than you will be able to join the conf by dialing the meetme number from other phones. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani Sent: Saturday, 29 August 2009 6:31 PM To: J Hogan; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] meet-me confrencing Make sure your software conference bridge is registered…if not go to IPVMS service parameter and turn it on From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan Sent: Saturday, 29 August 2009 3:51 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] meet-me confrencing Forgive me if this is a stupid question. But with meet-me confrencing for all internal to the CM conferences we should not need and conference bridge (router DSPs)? when I configure my meet me number all i get is a busy I configured the meet me pattern 1212 then I left the partition to none for full access. But even tried making the partition the same as the lines I was calling into it from. and Non secure but all I get is busy thanks -- J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI Yahoo ID: jhogan552000 AIM ID: jhogan55 MSN ID: jhogan55 ICQ ID: 257599283 Live Life And Do Not Kill Time. __ 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 __ 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
Re: [OSL | CCIE_Voice] Route Group
You probably already created route patterns and pointed them directly at the gateways/trunks. Once you do that, you can't add them to a route group. Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Tony Tong Sent: Sunday, August 30, 2009 4:52 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Route Group Hi, Perhaps someone can help me with this. I tried to create a route group and found none of my created gateways/trunks listed. not sure why? Rgds, Tony ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 1 Lab 2A -- SIP 7960 multiple calls per line
List, BR1-Phone1 is a 7960 SIP phone. When I add a line to it, the line seems only to support 2 calls. Multiple Call/Call Waiting Settings on Device SEP... Note:The range to select the Max Number of calls is: 1-2 Is there a way to change this so that the 7960 SIP phone can support more than 2 calls? Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 2A -- SIP 7960 multiple calls per line
OK Thanks. I have some gen 2 phones to play with as well. I just wanted to make sure there wasn't a setting in the CUCM that I was missing. Thanks all for the feedback. -Original Message- From: Daryl Smith [mailto:darylpsm...@gmail.com] Sent: Sunday, August 30, 2009 9:42 PM To: Jonathan Charles; Brian Valentine Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 2A -- SIP 7960 multiple calls per line I agree I have gen 1 phones and I have to schedule lab time to make sure my configs are valid. On 8/30/09 8:28 PM, Jonathan Charles jonv...@gmail.com wrote: Correction, I extended my sentence that should have read SIP sucks... Seriously, if you are studying for the lab, get some Gen 2s minimum (7941/61)... Jonathan On Sun, Aug 30, 2009 at 8:27 PM, Jonathan Charlesjonv...@gmail.com wrote: You are not going to be happy with Gen 1 phones and SIP... they suck. J On Sun, Aug 30, 2009 at 8:19 PM, Brian Valentinebkvalent...@gmail.com wrote: List, BR1-Phone1 is a 7960 SIP phone. When I add a line to it, the line seems only to support 2 calls. Multiple Call/Call Waiting Settings on Device SEPŠ.. Note:The range to select the Max Number of calls is: 1-2 Is there a way to change this so that the 7960 SIP phone can support more than 2 calls? 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 DPS There are no secrets to success. It is the result of preparation, hard work, and learning from failure ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] meet-me confrencing
If the regions for your phone and your CFB are set up so that the phone would be calling the CFB using G729, the call will fail. Software CFB (the one that runs on the CUCM) does not support G729 calls. The only codec it supports is G711ulaw. You would either need to adjust the regions or invoke a transcoder. Brian From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ilya Rubinchik Sent: Saturday, August 29, 2009 12:07 PM To: 'J Hogan' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] meet-me confrencing Hi I should press MeetMe software button instead of NewCall when creating confrnce -- Best regards, Ilya Rubinchik Chief UC Engineer Mars Solutions Ltd. 22, Munis str., Mirabad District Tashkent, 100080, Uzbekistan Tel UZ: +998 71 2907364 Fax UZ: +998 71 2907356 Mob UZ: +998 97 7128456 ICQ # 15508236 MSN: mailto:ilya.rubinc...@gmail.com ilya.rubinc...@gmail.com Skype: im_citius E-mail: mailto:ilya.rubinc...@followmars.com ilya.rubinc...@followmars.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan Sent: Saturday, August 29, 2009 7:51 PM To: Kumar, Narinder Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] meet-me confrencing Hello Everyone again :-) I have the CFB_2 registered. the IPVMS is activated, the CSS for the Meet-me is set to none. the media group on the phone has the group assigned to it that the CFB_2 belongs to. I hit new call -- meetme then I dial 2000. the pattern I have is 20[0-5]0 and I also have one as 1333 and everytime I get a busy signal thanks On Sat, Aug 29, 2009 at 9:26 AM, J Hogan j.jho...@gmail.com wrote: Team Thanks my software cfb is registered. But still no dice Sent from my iPhone On Aug 29, 2009, at 5:26 AM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: Hogan, As Aamir said make sure your software conf bridge is registered, also you need to initiate the meetme conference by pressing the meetme softkey and dialling the number, once the meetme conference is up than you will be able to join the conf by dialing the meetme number from other phones. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir Panjwani Sent: Saturday, 29 August 2009 6:31 PM To: J Hogan; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] meet-me confrencing Make sure your software conference bridge is registered…if not go to IPVMS service parameter and turn it on From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan Sent: Saturday, 29 August 2009 3:51 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] meet-me confrencing Forgive me if this is a stupid question. But with meet-me confrencing for all internal to the CM conferences we should not need and conference bridge (router DSPs)? when I configure my meet me number all i get is a busy I configured the meet me pattern 1212 then I left the partition to none for full access. But even tried making the partition the same as the lines I was calling into it from. and Non secure but all I get is busy thanks -- J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI Yahoo ID: jhogan552000 AIM ID: jhogan55 MSN ID: jhogan55 ICQ ID: 257599283 Live Life And Do Not Kill Time. __ 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 __ _ 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. -- J. Hogan
Re: [OSL | CCIE_Voice] transfers from CUC to Mobile Destination
Thanks! From: basant yadav [mailto:basant.ya...@gmail.com] Sent: Saturday, August 15, 2009 8:39 AM To: Brian Valentine Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] transfers from CUC to Mobile Destination Hi Brian Set the Service Parameter Display Original Calling Number on Transfer from Cisco Unity on Call Manager to True. HTH - Basant On Sat, Aug 15, 2009 at 1:37 PM, Brian Valentine bkvalent...@gmail.com wrote: Anyone know . When a call transfers out of a Unity Connection Call Handler out to an extension which is enabled for Mobile Connect, the remote destination shows the CLID as the voicemail port. Is it possible to pass the caller's CLID through? 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] Trouble with CRS JTAPI client versions error
Yes. This is a known issue when you install CRS on top of normal Windows server 2003. If you used the Install CD that comes from Cisco, you avoid this issue. There is a simple work-around. 1.create a folder named WINNT in c: 2.copy c:\WINDOWS\Java to WINNT Hope that helps. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Friday, August 14, 2009 9:40 AM To: OSL Group Subject: [OSL | CCIE_Voice] Trouble with CRS JTAPI client versions error After CRS 7 installation, has anyone run into the error, The Cisco JTAPI client versions are inconsistent. Please go to Cisco JTAPI Resync in the Unified Telephony Subsystem to install the JTAPI client.? Whenever I do this, the attempt fails and it says to try it again. It never fixes the issue. -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 11 digit local and LD issue
Well, I'll admit that local route groups is not one of my strengths. I haven't used them yet, and I'm not yet convinced that they will be very useful in a production environment. Having said that, I'll tell you have that I have configured this exact setup (but without local route groups) for a client. It was actually quite easy. Put the seven digit patterns in the Local partition (9.[2-9]XX). Put the 11 digit pattern in the Long Distance partition(9.1[2-9]XX[2-9]XX). Same as normal. The only difference is that when we send the local call, we strip predot and then prefix 1XXX (where XXX is the local area code). So, if the local area code were 408, and someone who has the CSS with the local partion in it dials the local number 95551212, we would send it out to the telco as 14085551212. If they dial 914085551212, but they don't have the LD partition in their CSS, the call would fail. Now, couldn't you do that using local route groups? I would imagine it would work if the gateways were H.323. Just make the translation in the dialpeer instead of in the callmanager. Am I way off? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, August 14, 2009 10:06 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 11 digit local and LD issue This isn't from the workbooks, but hopefully good discussion. If you are designing a dial plan where the PSTN requires 11 digits for both local and LD, what have some of you done to apply calling restrictions so that certain phones (lobby, breakroom, etc.) can only dial local calls. We don't want to use any long distance access codes such as FAC or PSTN codes. We are looking at using the Line/Device approach with local route groups. The only way I see it is to know of every area code/prefix that is considered local to that site, then create route patterns based off of those. This would be tedious work as new prefixes could get added and you may not know about them. 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
Re: [OSL | CCIE_Voice] Exciting Updates!!!!!
https://www.ipexpert.com/index.cfm/product/sku/CCIE_Voice_Lab_Video_on_Deman d_Series Website was updated with apparently no announcement in the OSL. Unless I missed something. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Larry Hadrava Sent: Friday, August 07, 2009 2:04 PM To: ccie_voice@onlinestudylist.com; ccie...@onlinestudylist.com; Cisco certification Subject: [OSL | CCIE_Voice] Exciting Updates! Hello All: This was just posted on the IPexpert blog: http://ipexpert.ccieblog.com/ CCIE Security 3.0 CCIE Voice 3.0 Clients, I know that most of you are anticipating the new products I'm about to announce - so here you go. Monday (actually, probably this weekend) - I will be updating our website to show the new CCIE Security 3.0 Video on Demand Course and new CCIE Voice 3.0 Video on Demand Course. All information for existing customers will be on those respected pages (we will also make a post on the http://www.onlinestudylist.com/ OSL support list, http://www.twitter.com/ipexpert Twitter and our http://www.facebook.com/pages/IPexpert/24586557119?ref=ts Facebook Group to notify you as soon as the website has been updated). Also, this weekend 4 CCIE Security 3.0 Volume 2 Workbook Labs (4 full 8-hour mock labs) will be added to your https://www.ipexpert.com/index.cfm/member? members area. If you do not see them Monday, contact supp...@ipexpert.com to request them. *As an FYI - our new mock labs will include Open Ended Questions as well as Troubleshooting. As many of you know - we're staying extremely busy cranking out new CCIE Voice 3.0 products, CCIE Security 3.0 products and CCIE RS 4.0 products - so thanks for your patience! Also, As always - if there's anything I can assist you with - please contact me directly at wlaw...@ipexpert.com - or feel free to contact your designated Training Advisor at sa...@ipexpert.com, via chat at http://www.ipexpert.com/chat www.ipexpert.com/chat or via telephone at +1.810.326.1444. Thanks Wayne -- Thanks Larry Hadrava CCIE #12203 CCNP Sr. Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Exciting Updates!!!!!
Of course I get . UPGRADE ERROR Our system does not recognize that you are eligible for this upgrade. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru Sent: Monday, August 10, 2009 7:41 PM To: Larry Hadrava Cc: ccie_voice@onlinestudylist.com; ccie...@onlinestudylist.com; Cisco certification Subject: Re: [OSL | CCIE_Voice] Exciting Updates! Am I reading the blog correctly? $25 to SHIP A DVD?! $50 TO MAKE ONE?! On Fri, Aug 7, 2009 at 11:03 AM, Larry Hadrava lar...@ipexpert.com wrote: Hello All: This was just posted on the IPexpert blog: http://ipexpert.ccieblog.com/ CCIE Security 3.0 CCIE Voice 3.0 Clients, I know that most of you are anticipating the new products I'm about to announce - so here you go. Monday (actually, probably this weekend) - I will be updating our website to show the new CCIE Security 3.0 Video on Demand Course and new CCIE Voice 3.0 Video on Demand Course. All information for existing customers will be on those respected pages (we will also make a post on the http://www.onlinestudylist.com/ OSL support list, http://www.twitter.com/ipexpert Twitter and our http://www.facebook.com/pages/IPexpert/24586557119?ref=ts Facebook Group to notify you as soon as the website has been updated). Also, this weekend 4 CCIE Security 3.0 Volume 2 Workbook Labs (4 full 8-hour mock labs) will be added to your https://www.ipexpert.com/index.cfm/member? members area. If you do not see them Monday, contact supp...@ipexpert.com to request them. *As an FYI - our new mock labs will include Open Ended Questions as well as Troubleshooting. As many of you know - we're staying extremely busy cranking out new CCIE Voice 3.0 products, CCIE Security 3.0 products and CCIE RS 4.0 products - so thanks for your patience! Also, As always - if there's anything I can assist you with - please contact me directly at wlaw...@ipexpert.com - or feel free to contact your designated Training Advisor at sa...@ipexpert.com, via chat at http://www.ipexpert.com/chat www.ipexpert.com/chat or via telephone at +1.810.326.1444. Thanks Wayne -- Thanks Larry Hadrava CCIE #12203 CCNP Sr. Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ 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
[OSL | CCIE_Voice] NM-CUE and 2801
Am I missing something? The CUE Compatibility matrix says that the NM-CUE is supported on 2801. Since the 2801 doesn't have network modules, should I assume this is a mistake? http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/compatibility/cuecom p.htm Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NM-CUE and 2801
Table 6 is also bad. Thanks, all, for the feedback. That's what I thought. Been a long week already. Just wanted to make sure I wasn't dreaming. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Thursday, August 06, 2009 9:19 PM To: Jason Granat Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] NM-CUE and 2801 Feels good to help. Yea, Table 7 is inaccurate. oops. _ From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat [...@slash128.com] Sent: Thursday, August 06, 2009 8:59 PM To: Jason Granat Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] NM-CUE and 2801 Hope that didn't sound too short. I was just excited to post an answer to a question here that I finally knew the answer to :-) Sent while mobile On Aug 6, 2009, at 17:53, Jason Granat j...@slash128.com wrote: If you scroll down to table 10 it shows that the only CUE module supported on the 2801 is the AIM-CUE. Sent while mobile On Aug 6, 2009, at 17:26, Brian Valentine bkvalent...@gmail.com wrote: Am I missing something? The CUE Compatibility matrix says that the NM-CUE is supported on 2801. Since the 2801 doesn't have network modules, should I assume this is a mistake? http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/compatibility/cuecom p.htm Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCME 7.0 GUI....
Hmm. I've seen this before. Trying to remember when/why. Do you have ... telephony-service web admin system name admin password cisco Brian -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Sunday, August 02, 2009 12:54 PM To: chikki venu Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCME 7.0 GUI I have those commands... And since these are HTTP requests, not TFTP requests, there should be no need for the TFTP-server commands. Here is a debug ip http all: Aug 2 16:50:14.794: its_urlhook url: /telephony_service.html, method 1 Aug 2 16:50:14.798: Sun, 02 Aug 2009 16:50:14 GMT 10.50.5.115 /telephony_service.html auth_required Protocol = HTTP/1.1 Method = GET Aug 2 16:50:14.798: BR2(config)# Aug 2 16:50:18.282: its_urlhook url: /telephony_service.html, method 1 Aug 2 16:50:18.290: validate_username_password (admin, csico)validate_admin_user (admin: csico) validate_admin_user: validate admin [0] locally validate_admin_user: password from acct failed! validate_admin_user (admin: csico) validate_username_password: check password for phone[-1] Aug 2 16:50:18.290: HTTP: Authentication failed for realm its_access Aug 2 16:50:18.290: HTTP: Authentication failed for level 15 BR2(config)# Aug 2 16:50:20.290: Sun, 02 Aug 2009 16:50:20 GMT 10.50.5.115 /telephony_service.html auth_failed Protocol = HTTP/1.1 Method = GET Aug 2 16:50:20.290: BR2(config)# Aug 2 16:50:22.198: its_urlhook url: /telephony_service.html, method 1 Aug 2 16:50:22.202: HTTP: Priv level granted 15 Aug 2 16:50:22.206: Sun, 02 Aug 2009 16:50:22 GMT 10.50.5.115 /telephony_service.html ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:22.206: Aug 2 16:50:22.206: telephony_service_server_get_action url:/telephony_service.html Aug 2 16:50:22.262: its_urlhook url: /ITSMain, method 1 Aug 2 16:50:22.266: HTTP: Priv level granted 15 Aug 2 16:50:22.266: Sun, 02 Aug 2009 16:50:22 GMT 10.50.5.115 /ITSMain ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:22.266: Aug 2 16:50:22.270: telephony_service_server_get_action url:/ITSMain BR2(config)# Aug 2 16:50:22.270: ipkeyswitch_generate_html_header: start 1, 0 Aug 2 16:50:22.270: ipkeyswitch_generate_html_header: admin java variable Aug 2 16:50:22.286: outputPhoneLoad: phoneLoad_num=0 Aug 2 16:50:22.418: ipkeyswitch_generate_html_header: admin variable done Aug 2 16:50:22.574: its_urlhook url: /admin_user.js, method 1 Aug 2 16:50:22.578: HTTP: Priv level granted 15 Aug 2 16:50:22.578: Sun, 02 Aug 2009 16:50:22 GMT 10.50.5.115 /admin_user.js ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:22.578: Aug 2 16:50:22.582: telephony_service_server_get_action url:/admin_user.js BR2(config)# Aug 2 16:50:26.374: its_urlhook url: /dom.js, method 1 Aug 2 16:50:26.382: HTTP: Priv level granted 15 Aug 2 16:50:26.382: Sun, 02 Aug 2009 16:50:26 GMT 10.50.5.115 /dom.js ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:26.382: Aug 2 16:50:26.386: its_urlhook url: /logohome.gif, method 1 Aug 2 16:50:26.394: HTTP: Priv level granted 15 Aug 2 16:50:26.394: Sun, 02 Aug 2009 16:50:26 GMT 10.50.5.115 /logohome.gif ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:26.394: Aug 2 16:50:26.394: its_urlhook url: /sxiconad.gif, method 1 Aug 2 16:50:26.402: HTTP: Priv level granted 15 Aug 2 16:50:26.402: Sun, 02 Aug 2009 16:50:26 GMT 10.50.5.115 /sxiconad.gif ok BR2(config)#Protocol = HTTP/1.1 Method = GET Aug 2 16:50:26.402: Aug 2 16:50:26.406: its_urlhook url: /Tab.gif, method 1 Aug 2 16:50:26.414: HTTP: Priv level granted 15 Aug 2 16:50:26.414: Sun, 02 Aug 2009 16:50:26 GMT 10.50.5.115 /Tab.gif ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:26.414: Aug 2 16:50:26.414: telephony_service_server_get_action url:/dom.js Aug 2 16:50:26.418: telephony_service_server_get_action url:/logohome.gif Aug 2 16:50:26.422: telephony_service_server_get_action url:/sxiconad.gif Aug 2 16:50:26.426: telephony_service_server_get_action url:/Tab.gif Aug 2 16:50:26.442: its_urlhook url: /CiscoLogo.gif, method 1 BR2(config)# Aug 2 16:50:26.454: HTTP: Priv level granted 15 Aug 2 16:50:26.454: Sun, 02 Aug 2009 16:50:26 GMT 10.50.5.115 /CiscoLogo.gif ok Protocol = HTTP/1.1 Method = GET Aug 2 16:50:26.454: Aug 2 16:50:26.458: its_urlhook url: /favicon.ico, method 1 Aug 2 16:50:26.458: lds_urlhook, url=/favicon.ico Aug 2 16:50:26.458: telephony_service_server_get_action url:/CiscoLogo.gif BR2(config)# And attached is a screenshot of what I get when I go to the web page... It looks like it is a java error... Any ideas? Jonathan On Sun, Aug 2, 2009 at 10:38 AM, chikki venuchik...@yahoo.com wrote: hi j you need match the loaction of the GUI fiels on flash and http path command on the cinfigurtaion to see the listing of the files on the flash you have following commands show flash: or dir flash:
