Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post back in January
A follow up on the directed call park scenario with the BLF monitor of the slot. Been awhile since I played with this so I want to ensure my understanding is correct. Essentially the BLF monitor of directed call park really allows nothing beyond having an appearance on the phone that will show if a call is currently parked in that slot - correct? The BLF is completely useless for actually being able to pickup a parked call - correct? Seemed odd to me when I was playing around with it - why program a feature like BLF Directed Call Park but not allow the feature to be used to actually capture the call. Just want to ensure my memory of the feature is correct. Thanks Scott Date: Thu, 1 Apr 2010 20:22:11 -0430 From: o...@ipexpert.com To: ciscovoiceg...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post back in January Matthew, Right, the statement has to be corrected in the PG, About the sip dial rule, make sure that for each pattern description, there's only one dial parameter pattern and timeout pair, hth, On Thu, Apr 1, 2010 at 1:13 PM, Matthew Berry ciscovoiceg...@gmail.com wrote: Otto - I am working through Vol 1 Lab 8 Question 8.2. In the verifications section of the PG (p. 478) I am told: Retrieve the call by pressing the BLF Speed Dial from one of the phones... However, whenever I try to retrieve the call this way I get a reorder tone and Park Slot Unavailable. I was reading one of your responses on the OSL archive to this issue and you said: When you hit the directed call park BLF SD, the ucm thinks that you want to park a call in that slot, not that are going to retrieve it, and since there can be only one call in the park slot, you see the unavailable message when a call is already there. If that is true, it seems that there is an error in the PG. Does that sound right? I am able to retrieve the call by dialing 80-8555, but not by going off-hook and pressing the Call Park BLF Speed Dial. Also, my 8... SIP dial rule with Timeout = 0 has not taken effect after several restarts. Any ideas? Please let me know if my assumption is correct. -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written Gmail: ciscovoiceguru Skype: ciscovoiceguru Twitter: ciscovoiceguru 1st Lab Attempt: Aug 16, 2010 -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Bandwidth Per Call
I have found two possible solutions: 1) The intra-region specification previously mentioned - which seems to make no sense as the GK calls are most likely INTER region - I.e. between a HQ region and a GK region that is G.729 across the board - but it is a bug so not surprising it seems off. 2) Enable BRQ in service parameters - even if GK CAC is not in use - this for some reason forces the initial G.729 negotiation. I have found that without these techniques the call will eventually utilize G.729 - as evidence by phone stats. But a show gatekeeper call will show the call at 128K. With either of these techniques enabled show gatekeeper call will show 16K. Scott From: gorr...@hotmail.com To: mn...@netelligent.com; martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Date: Thu, 1 Apr 2010 07:57:44 + Subject: Re: [OSL | CCIE_Voice] Bandwidth Per Call Hello: If you think that your config is ok (dial-peer with codec g729 and trunk in g729 region) you are probably under this bug: CSCsl74701 The workaround is setting intra-region coded to g279 under service parameter. If you do a deb h225 asn1 you must see a ARQ with 128 and then a BRQ message updating bw to 16 (please post this debug), finally you will see the call as 16k under sho gatek calls Also check this link: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg12537.html hth From: mn...@netelligent.com To: martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Date: Wed, 31 Mar 2010 22:33:38 -0500 Subject: Re: [OSL | CCIE_Voice] Bandwidth Per Call Since the call is using a voip dial-peer I assumed it was a g.729 call. Also, when I press the? Button twice, I do see that the call is at g.729. Mark Nigh Systems Engineer mn...@netelligent.com (p) 314.392.6926 From: Marty Beutler [mailto:martybeut...@hotmail.com] Sent: Wednesday, March 31, 2010 7:46 PM To: Mark Nigh; CCIE OSL Subject: RE: [OSL | CCIE_Voice] Bandwidth Per Call Hey Mark, I think I'd start by adding a codec statement to the dialpeers, specifying g729. If the GK says 128, its a g711 call. Good luck, -Marty From: mn...@netelligent.com To: ccie_voice@onlinestudylist.com Date: Wed, 31 Mar 2010 19:35:47 -0500 Subject: [OSL | CCIE_Voice] Bandwidth Per Call I am working on Lab 1, question 4.3 specially the bandwidth under the “show gatekeeper call”. I am unable to get the output to show 16kbps. The calls completes and if I do a call status on the phone I see that the codec is g.729, but the gatekeeper see is as 128kbps. Here is the GK configuration: gatekeeper zone local CUCM cisco.com zone local CUCME cisco.com zone prefix CUCM 2... gw-priority 10 gk-trunk_2 zone prefix CUCM 2... gw-priority 9 gk-trunk_1 gw-type-prefix 1#* gw ipaddr 10.5.24.101 1720 gw ipaddr 10.5.24.100 1720 gw-type-prefix 2#* gw ipaddr 10.6.255.3 1720 no shutdown Dial-peer configuration: dial-peer voice 100 voip destination-pattern 2... session target ras tech-prefix 1# Output of show gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 9-9197 244 128(Kbps) Endpt(s): Alias E.164Addr src EP: CUCME 4002 CallSignalAddr Port RASSignalAddr Port 10.6.255.3 1720 10.6.255.3 56566 Endpt(s): Alias E.164Addr dst EP: gk-trunk_21#2002 CallSignalAddr Port RASSignalAddr Port 10.5.24.101 43384 10.5.24.101 32785 Your thoughts are appreciated. Mark Nigh Systems Engineer mn...@netelligent.com (p) 314.392.6926 This transmission and any attached files are privileged, confidential or otherwise the exclusive property of the intended recipient or Netelligent Corporation. If you are not the intended recipient, any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is strictly prohibited. If you have received this transmission in error, please contact us immediately by responding to this message or by telephone (314-392-6900) and promptly destroy the original transmission and its attachments. This transmission and any attached files are privileged, confidential or otherwise the exclusive property of the intended recipient or Netelligent Corporation. If you are not the intended recipient, any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is strictly prohibited. If you have received this transmission in error, please contact us immediately by responding to this message or by telephone (314-392-6900) and promptly destroy the original transmission and its attachments. Hotmail: Powerful Free email with security
Re: [OSL | CCIE_Voice] Bandwidth Per Call
Yes - I meant that the solution works simply to ensure show gatekeeper calls at 16K when CAC is not actually in use. Either way - probably better to use intra region G.729 specification - just making the point that either will work with no CAC. From: gorr...@hotmail.com To: scarruthe...@hotmail.com; mn...@netelligent.com; martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Bandwidth Per Call Date: Thu, 1 Apr 2010 11:41:43 + Hi Scott: With the second aproach, even if you see the call as 16k you call can be rejected by gatekeeper in the following situation gatekeeper badwith zone ucm 32 ! 2 g729 calls for example or any number lower than 128 brq enable at service param A call try to cross de gk at the inital ARQ message the call will ask for 128k so the call is rejected, and BRQ will never be processed, if you dont have the bandwith command at gk after the ARQ command the BRQ will update the bandwith to 16k.. hope this make sense The first aproach looks to works in all situations Regards From: scarruthe...@hotmail.com To: gorr...@hotmail.com; mn...@netelligent.com; martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Bandwidth Per Call Date: Thu, 1 Apr 2010 04:14:38 -0700 I have found two possible solutions: 1) The intra-region specification previously mentioned - which seems to make no sense as the GK calls are most likely INTER region - I.e. between a HQ region and a GK region that is G.729 across the board - but it is a bug so not surprising it seems off. 2) Enable BRQ in service parameters - even if GK CAC is not in use - this for some reason forces the initial G.729 negotiation. I have found that without these techniques the call will eventually utilize G.729 - as evidence by phone stats. But a show gatekeeper call will show the call at 128K. With either of these techniques enabled show gatekeeper call will show 16K. Scott From: gorr...@hotmail.com To: mn...@netelligent.com; martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Date: Thu, 1 Apr 2010 07:57:44 + Subject: Re: [OSL | CCIE_Voice] Bandwidth Per Call Hello: If you think that your config is ok (dial-peer with codec g729 and trunk in g729 region) you are probably under this bug: CSCsl74701 The workaround is setting intra-region coded to g279 under service parameter. If you do a deb h225 asn1 you must see a ARQ with 128 and then a BRQ message updating bw to 16 (please post this debug), finally you will see the call as 16k under sho gatek calls Also check this link: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg12537.html hth From: mn...@netelligent.com To: martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Date: Wed, 31 Mar 2010 22:33:38 -0500 Subject: Re: [OSL | CCIE_Voice] Bandwidth Per Call Since the call is using a voip dial-peer I assumed it was a g.729 call. Also, when I press the? Button twice, I do see that the call is at g.729. Mark Nigh Systems Engineer mn...@netelligent.com (p) 314.392.6926 From: Marty Beutler [mailto:martybeut...@hotmail.com] Sent: Wednesday, March 31, 2010 7:46 PM To: Mark Nigh; CCIE OSL Subject: RE: [OSL | CCIE_Voice] Bandwidth Per Call Hey Mark, I think I'd start by adding a codec statement to the dialpeers, specifying g729. If the GK says 128, its a g711 call. Good luck, -Marty From: mn...@netelligent.com To: ccie_voice@onlinestudylist.com Date: Wed, 31 Mar 2010 19:35:47 -0500 Subject: [OSL | CCIE_Voice] Bandwidth Per Call I am working on Lab 1, question 4.3 specially the bandwidth under the “show gatekeeper call”. I am unable to get the output to show 16kbps. The calls completes and if I do a call status on the phone I see that the codec is g.729, but the gatekeeper see is as 128kbps. Here is the GK configuration: gatekeeper zone local CUCM cisco.com zone local CUCME cisco.com zone prefix CUCM 2... gw-priority 10 gk-trunk_2 zone prefix CUCM 2... gw-priority 9 gk-trunk_1 gw-type-prefix 1#* gw ipaddr 10.5.24.101 1720 gw ipaddr 10.5.24.100 1720 gw-type-prefix 2#* gw ipaddr 10.6.255.3 1720 no shutdown Dial-peer configuration: dial-peer voice 100 voip destination-pattern 2... session target ras tech-prefix 1# Output of show gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 9-9197 244 128(Kbps) Endpt(s): Alias E.164Addr src EP: CUCME 4002 CallSignalAddr Port RASSignalAddr Port 10.6.255.3 1720 10.6.255.3 56566 Endpt(s): Alias E.164Addr dst EP: gk-trunk_21#2002 CallSignalAddr Port RASSignalAddr Port 10.5.24.101 43384 10.5.24.101 32785
[OSL | CCIE_Voice] PL - Inability To Change CUE License
Anyone ever encounter the following error when attempting to change the CUE license on proctorlab's modules? Attempting to change from the CME to CM license. Would seem to be an obvious flash space issue but I cleared some crash files, etc but no actions seem to allow the license install. Additionally - below the file install attempt you can see the flash is really not all that low on space. I've tried resetting the module several times but nothing helps. se-10-10-202-250# $p://10.10.210.5/cue-vm-license_12mbx_ccm_7.0.1.pkg Online install/download is not allowed due to insufficient FLASH capacity 256503808 bytes total (123088896 bytes free) Thanks Scott _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/210850553/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper registration questions
Paul, I haven't played around with forcing the output Type to either VOIP-GW or H323-GW much - but are you sure that your attempts is not backwards? In other words - I had the site C GW configured as a CUBE/with allow-connections configured and my gatekeeper output looked like: GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 60462 GKH323-GW E164-ID: 3001 E164-ID: 3002 E164-ID: 3999 H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 10.10.210.101720 10.10.210.1032885 GKVOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 10.10.210.111720 10.10.210.1132786 GKVOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 I removed the allow connection statements and my output changed to - this would seem to indicate the device will register as a VOIP-GW when it is not acting as a ip to ip GW: HQ-RTR#show gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 51687 GKVOIP-GW E164-ID: 3001 E164-ID: 3002 E164-ID: 3999 H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 10.10.210.101720 10.10.210.1032885 GKVOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 10.10.210.111720 10.10.210.1132786 GKVOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 Date: Thu, 25 Mar 2010 12:57:37 -0400 From: pa...@marshallcomm.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper registration questions All, Can anyone explain the core of how a gateway registration is recorded with the “sh gatek endp” command. Normally, as I understand it, a full H323 gw will register to a gk as type “H323-GW”, whereas the UCM will register as “VOIP-GW”. However, I’m running into a situation where no matter what I do my BR2 endpoint is registering with the HQ gk as VOIP-GW. I found reference elsewhere to a 323gw will register as voip-gw if it is configured as a CUBE. In this case though it isn’t configured as a CUBE. I’ve put the pertinent VERY basic commands below. What I’m really looking for is some clarification on the different types and why they register differently if possible. Thanks for any help, Paul (#16842 RS/Sec) HQ-RTR#sh run | s gatekeeper gatekeeper zone local HQ ipexpert.com 10.10.110.1 no shutdown HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 65184 HQ VOIP-GW E164-ID: 3102 H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.1.200.20 48779 10.1.200.20 32824 HQ VOIP-GW H323-ID: HQgk_1 Voice Capacity Max.= Avail.= Current.= 0 10.1.200.21 37078 10.1.200.21 32849 HQ VOIP-GW H323-ID: HQgk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registratio BR2-RTRen BR2-RTR#sh run int Loop0 Building configuration... Current configuration : 163 bytes ! 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 h323-id BR2-RTR end BR2-RTR#sh gateway H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1 H.323 service is up Gateway BR2-RTR is registered to Gatekeeper HQ Alias list (CLI configured) E164-ID 3102 H323-ID BR2-RTR Alias list (last RCF) E164-ID 3102 H323-ID BR2-RTR H323 resource thresholding is Disabled BR2-RTR# BR2-RTR#sh run | s voice service voice service voip _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME busy-trigger-per-button
Otto, Could hunt stop channel be used in combination with max-calls-per-button? In other words - if I had a shared line at site C of 3500 - I want one phone with this appearance to be able to accept 2 calls on the shared line - I want the second phone to be able to accept 3 calls on the shared line - with an additional stipulation that only 4 simultaneous calls to the shared line should be possible - could I do it as follows: ephone-dn 10 number 3500 huntstop channel 4 ephone 1 max-calls-per-button 2 button 1:10 ephone 2 max-calls-per-button 3 button 1:10 I thought this would work but when I have tried it does not. Thanks Scott Date: Tue, 16 Mar 2010 09:02:34 -0430 From: o...@ipexpert.com To: aar...@packet360.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME busy-trigger-per-button Hi, Have you tried to use huntstop channel 2 on the 1001 dn?, this will limit the number of incoming calls the 1001 dn will receive, busy-trigger-per-button command takes into account incoming and outgoing calls, On Sat, Mar 13, 2010 at 12:43 AM, Aman Arora aar...@packet360.com wrote: Hey Folks Is it possible to set different busy-trigger-per-button for each button (line) on a phone on CME. For example : If I have line 1 : 1000 And line 2 : 1001 I need to limit 4 incoming calls on line 1 and limit 2 incoming calls on line 2. How can I achieve this. I guess busy-trigger-per-button sets limits for all the buttons on the particular ephone. Thanks Aman -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, March 12, 2010 8:05 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 49, Issue 85 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: UC and cme sip integration (Omotayo) 2. Re: SIP Hardware Transcoder (Jeff Cotter) 3. Re: SIP Hardware Transcoder (Omotayo) -- Message: 1 Date: Fri, 12 Mar 2010 23:01:11 +0100 From: Omotayo adefilabi...@gmail.com Subject: Re: [OSL | CCIE_Voice] UC and cme sip integration To: Flemming Ortvald f...@netdesign.dk Cc: OSL Group ccie_voice@onlinestudylist.com Message-ID: 3082f9d41003121401o3d85ff29id1e503233e21d...@mail.gmail.com Content-Type: text/plain; charset=windows-1252 Hello, it work ok now I was using the wrong ip address on the unity connection all the while Thanks On Fri, Mar 12, 2010 at 8:30 AM, Flemming Ortvald f...@netdesign.dk wrote: Unity connection can do both g729 and g711, you can use ?voice class codec? on ?voice register dn? to expand codec support for sip. Med venlig hilsen Flemming Ortvald Network System Eng. NetDesign A/S +45 4435 8346 T?nk p? milj?et inden udskrivning af denne e-post og tilknyttede vedh?ftninger *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* 11 March, 2010 20:58 *To:* Flemming Ortvald *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration Hello, I have to configure a transcoder on the br2 router? Unity connection support g729 only? Rgd On Thu, Mar 11, 2010 at 8:24 PM, Flemming Ortvald f...@netdesign.dk wrote: You will need a transcoder or chnage the sip endpoints to support g.711, natively it only supports g.729 Best regards Flemming Ortvald Network System Eng. NetDesign A/S +45 4435 8346 T?nk p? milj?et inden udskrivning af denne e-post og tilknyttede vedh?ftninger *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* 11 March, 2010 20:07 *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration Hello all, As anyone been able to get the SIP integration between Unity Connection and Cme to work? I followed the Proctorlabs Guide I posted this sometime lat week and revised as advised but keep getting a reorder tone( Number Unknown) when the message button is pressed Below is the relevant configuration voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 voice
Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK
Is the lab you are referring to - lab 5C - in CCIE Voice Workbook V6.0 Volume 1? When I download the section labeled Workbook 5A, 5b, and 5C - the PDF actually only contains 5A and 5B - I do not find a lab 5C - and thus I find no PSTN GK scenarios. Am I going to the correct section? Thanks Scott Date: Sun, 14 Mar 2010 11:17:56 -0500 Subject: Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK From: cciet...@gmail.com To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com Yes it does esp lab 5c On 3/13/10, scott carruthers scarruthe...@hotmail.com wrote: In some CCIE V2 labs Proctorlabs PSTN router was configured as GK so that we could practice remote zone scenarios. I have not reviewed all of the new lab IP Expert labs - do any call for sending calls thru our own HQ GK to a remote zone outside of our direct control? Do any of the Proctorlabs pre-configs configure the PSTN router as a GK? Thanks Scott _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_1 -- Sent from my mobile device www.ccietalk.com _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/210850553/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK
In some CCIE V2 labs Proctorlabs PSTN router was configured as GK so that we could practice remote zone scenarios. I have not reviewed all of the new lab IP Expert labs - do any call for sending calls thru our own HQ GK to a remote zone outside of our direct control? Do any of the Proctorlabs pre-configs configure the PSTN router as a GK? Thanks Scott _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_1___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists
Thanks Naoufal, I actually already had the presence call-list, presence-enable, and allow watch commands in place - the latter two had to be there as the BLF SD was working for the same lines I was attempting to get presence for in the call list. That's why it was odd - everything seemed to be in place. Anyone else have trouble with this? Thanks Scott Date: Tue, 9 Mar 2010 08:55:23 + From: naoufal.kerbo...@cbi.ma To: scarruthe...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists Hi scott, try to add this lines on your config: presence presence call-list ! sip-ua presence enable ! also allow watch under ephone-dn for all directory numbers. Regards Naoufal scott carruthers wrote: I have review the configuration a dozen times - I have BLF speed dials between two phones properly showing presence status on the line appearance/SD but I cannot get the call list presence functionality to work. I have presence call-list specified under presence config mode - obviously the allow watch, etc is configured on the phones properly as the BLFs are working. Any ideas? Thanks Scott Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. http://clk.atdmt.com/GBL/go/201469229/direct/01/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Issues With CME BLF Call Lists
I have review the configuration a dozen times - I have BLF speed dials between two phones properly showing presence status on the line appearance/SD but I cannot get the call list presence functionality to work. I have presence call-list specified under presence config mode - obviously the allow watch, etc is configured on the phones properly as the BLFs are working. Any ideas? Thanks Scott _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/201469229/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists
Otto, This was actually for a CME BLF call list. On a CME phone I am able to get the BLF SDs to work but the presence status in the missed calls directory does not work - despite having presence call-list parameter set - and the watch commands, etc proper. Any ideas? Thanks Scott Subject: Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists To: scarruthe...@hotmail.com; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com From: o...@ipexpert.com Date: Tue, 9 Mar 2010 01:35:30 + Hi Scott, I don't know if I understood well, but did you make sure the blf for call list enterprise parameter is set to enable?, also are the phones within the same presence group? Thanks, -Original Message- From: scott carruthers scarruthe...@hotmail.com Date: Mon, 8 Mar 2010 14:33:18 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Issues With CME BLF Call Lists ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/201469229/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Endpoint Max-Calls Command
Otto, Many thanks for spending the time to detail this. This makes complete sense - configure another CM trunk that will exist in a secondary route group that the CM will attempt to re-try the call via. I will try this in a lab today but seems simple enough. I thought that when I used this config in the voice IE V2/CM 4.X days there was also a service parameter that was needed to allow the CM to re-attempt the call - possibly that requirement/parameter is no longer in CM 7.X. Thanks again, Scott Date: Sun, 7 Mar 2010 13:30:35 -0430 Subject: Re: [OSL | CCIE_Voice] Gatekeeper Endpoint Max-Calls Command From: o...@ipexpert.com To: scarruthe...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi Scott, Your gk config is good, So, you have to work a little from your ucm cluster. A workaround to your requirement is to configure 2 separate trunks, each with only one server within its ucm group ie ccm-trunk-pub, ucm group consisting on the ucm pub only ccm-trunk-sub, ucm group consisting on the ucm sub only The output from the sh gatek endpoint may look like this: * HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 59738 CME VOIP-GW H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.210.1036392 10.10.210.1032797 HQVOIP-GW H323-ID: gk-trunkpub_1 Voice Capacity Max.= 1000 Avail.= 1000 Current.= 0 10.10.210.1134175 10.10.210.1132790 HQVOIP-GW H323-ID: gk-trunksub_2 Voice Capacity Max.= 1000 Avail.= 1000 Current.= 0 Total number of active registrations = 3 * Then, configure a route group with the two new gw with gk-trunksub_2 in the first place and the distribution algorithm top down, then the rl and rp, in that way your calls will always be sent from the sub node and if they fail the pub node will take care, Your gk config may have to change a bit to accomodate this changes, At the end you will see an output like this, *** HQ-RTR#sh gatek call Total number of active calls = 2. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 42-32984 15 0(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunksub_25002 CallSignalAddr Port RASSignalAddr Port 10.10.210.1134175 10.10.210.1132790 Endpt(s): Alias E.164Addr dst EP: BR2-RTR 1#3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1720 10.10.110.3 59738 LocalCallIDAge(secs) BW 43-32986 10 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunkpub_1 5003 CallSignalAddr Port RASSignalAddr Port 10.10.210.1036392 10.10.210.1032797 Endpt(s): Alias E.164Addr dst EP: BR2-RTR 1#3002 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1720 10.10.110.3 59738 *** My gk config was the following *** gatekeeper zone local HQ ipexpert.com 10.10.110.1 zone local CME ipexpert.com zone prefix CME 3... zone prefix HQ 5... gw-priority 10 gk-trunksub_2 zone prefix HQ 5... gw-priority 9 gk-trunkpub_1 no shutdown endpoint resource-threshold endpoint max-calls h323id gk-trunksub_2 1 endpoint max-calls h323id gk-trunkpub_1 1000 !*** There's also a good reading on how the outbound calls are handled by the h.225 trunks in ucm, http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1044813 HTH, ,On Sat, Mar 6, 2010 at 9:04 PM, scott carruthers scarruthe...@hotmail.com wrote: With the gatekeeper max-calls command - I have the following config to only allow one active call to the CM sub and all subsequent calls will route to the pub: no shutdown endpoint resource-threshold endpoint max-calls h323id CM-Trunk_2 1 endpoint max-calls h323id CM-Trunk_1 1000 This works fine for call from CME to CM thru the GK. But in the CM to CME direction one call succeeds and any subsequent calls fail. It seems CM will not re-initiate the call attempt from the pub. I thought there was a service parameter needed in conjunction with the max call command on the GK but I am not able to find. Am I missing a simple service parameter or is something else needed
[OSL | CCIE_Voice] Hardcoding GK RAS Port
Been awhile since I try to force a GK to use specific RAS ports. As you can see below - in CM service parameters I have made the specifications necessary to ensure the GK controlled trunk will utilize port 1720. This is accurately updated on the GK. But I am unable to force the GK to use 1719. The trunk 1720 portion is the important portion from a functionality standpoint - I can now statically configure alias/gw-type-prefix statements without the possibility of the trunk port altering upon reset. But I am concerned if I am shown a screen shot on the GK port listed for the CM trunk is 1719. In the past I have never had an issue - but hadn't touched this in awhile - and in two recent labs I have now had this issue. Any thoughts? Device Name of GK-controlled Trunk That Will Use Port 1720 None Host Name/IP Address of GK That Will Use RAS UDP Port 1719 None GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 62348 CME H323-GW H323-ID: CME Voice Capacity Max.= Avail.= Current.= 0 10.10.210.101720 10.10.210.1033136 CCM VOIP-GW H323-ID: CM-Trunk_1 Voice Capacity Max.= 1000 Avail.= 1000 Current.= 0 10.10.210.111720 10.10.210.1132785 CCM VOIP-GW H323-ID: CM-Trunk_2 Voice Capacity Max.= 1 Avail.= 1 Current.= 0 Total number of active registrations = 4 _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/201469230/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Manipulating Displayed Number On Phone For Egress Call
I posted a very similar topic a few weeks ago - thought I would have no further problems - but now having issue with MGCP Question posed - when a HQ phone dials an international ensure the IP phone is updated once the call is connected to show the called number without the access code or international prefix. In other words - if HQ phone 1 dials 90113432141891 - once the call is connected the HQ phone should display: To 3432141891. In my last post on this topic I was sending the call to a H323 GW and I was previously unaware of the voice service voip command - no supplementary-service h225-notify cid-update. When I use this I can now create a route pattern of 9011.! strip predot in the route pattern strip predot and prefix 011 in the route list the phone will update only with the manipulation in the route pattern and I can meet the requirement with no problem. But now I encounter an issue meeting the requirement with MGCP GWs - which I had previously taken for granted thinking this would be no problem. I would again create a route pattern of 9011.! strip predot in the route pattern strip predot and prefix 011 in the route list and since with MGCP the phone will only update with the manipulations in the route pattern I would be done. But this does not seem to be the case. When the call is connect/in the ringing state the display shows 0113432141891. It appears to be updating with the manipulations made in the route list which I did not believe should happen. Any ideas? I have search thru possible config alterations in the CM MGCP GW instance but find nothing relevant. Thanks Scott _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/201469230/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Manipulating Displayed Number On Phone For Egress Call
Thanks for the idea Ash - I will try it with that method in my next lab. Update of this matter - I played around with this more with another guy and we were able to narrow this down. So the summary is: -On a route pattern of 9011.! - the route pattern manipulation will properly update on the connected call -On a route pattern of 9011.!# - the route pattern manipulation will not properly update on the connected call So it seems to be the trailing # that breaks it - not sure why. In the lab I doubt it would be of concern - we could either use the 9011.! pattern if this was called for - or if the proctor said interdigit timeout should not experienced we could create specific patterns for the PSTN international line - 90113432141891 (ie) and make it urgent pri and it should be fine. Maybe someone out there has a reason why the trailing # would have an impact. Date: Sat, 6 Mar 2010 21:58:20 + From: siddas...@gmail.com To: scarruthe...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Manipulating Displayed Number On Phone For Egress Call Try the following: Pattern: 9011! Partition: US-International (or whatever) Called party transformation: Pattern: 9011.! Partition: HQ-Called PreDot At Device pool -Called party Transformation CSS-- add HQ-Called. At the RL level, prefix digit 011. I think this should work. Ash scott carruthers wrote: I posted a very similar topic a few weeks ago - thought I would have no further problems - but now having issue with MGCP Question posed - when a HQ phone dials an international ensure the IP phone is updated once the call is connected to show the called number without the access code or international prefix. In other words - if HQ phone 1 dials 90113432141891 - once the call is connected the HQ phone should display: To 3432141891. In my last post on this topic I was sending the call to a H323 GW and I was previously unaware of the voice service voip command - no supplementary-service h225-notify cid-update. When I use this I can now create a route pattern of 9011.! strip predot in the route pattern strip predot and prefix 011 in the route list the phone will update only with the manipulation in the route pattern and I can meet the requirement with no problem. But now I encounter an issue meeting the requirement with MGCP GWs - which I had previously taken for granted thinking this would be no problem. I would again create a route pattern of 9011.! strip predot in the route pattern strip predot and prefix 011 in the route list and since with MGCP the phone will only update with the manipulations in the route pattern I would be done. But this does not seem to be the case. When the call is connect/in the ringing state the display shows 0113432141891. It appears to be updating with the manipulations made in the route list which I did not believe should happen. Any ideas? I have search thru possible config alterations in the CM MGCP GW instance but find nothing relevant. Thanks Scott Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper Endpoint Max-Calls Command
With the gatekeeper max-calls command - I have the following config to only allow one active call to the CM sub and all subsequent calls will route to the pub: no shutdown endpoint resource-threshold endpoint max-calls h323id CM-Trunk_2 1 endpoint max-calls h323id CM-Trunk_1 1000 This works fine for call from CME to CM thru the GK. But in the CM to CME direction one call succeeds and any subsequent calls fail. It seems CM will not re-initiate the call attempt from the pub. I thought there was a service parameter needed in conjunction with the max call command on the GK but I am not able to find. Am I missing a simple service parameter or is something else needed? _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/201469229/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUC Not Licensed For VPIM
When I attempt to add a VPIM location is Unity Connection I receive the following license error. Are the proctorlabs servers not licensed for VPIM? Anyone attempt VPIM in these labs yet? Status The requested operation would result in a license violation. Unable to create VPIM Location Save New VPIM Location Display Name* Dtmf Access ID* Partition cuc7-pub Partition Domain Name* IP Address* Remote phone prefix Save Fields marked with an asterisk (*) are required. The Demo license info show nothing for VPIM: SERVER this_host ANY VENDOR cisco INCREMENT LicVoicePortsMax cisco 7.0 permanent 2 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID0/LicLineID \ PAKdummyPak/PAK SIGN=A3DF5BBED8B0 INCREMENT LicSubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID1/LicLineID \ PAKdummyPak/PAK SIGN=FA226A483396 INCREMENT LicVMISubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID2/LicLineID \ PAKdummyPak/PAK SIGN=22D6A4F63854 INCREMENT LicAdvancedUserMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID3/LicLineID \ PAKdummyPak/PAK SIGN=85B5BD2CDF32 INCREMENT LicRealspeakSessionsMax cisco 7.0 permanent 2 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID4/LicLineID \ PAKdummyPak/PAK SIGN=24848F662AEC INCREMENT LicServerBackend cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID5/LicLineID \ PAKdummyPak/PAK SIGN=6750CF4C26B4 INCREMENT LicIMAPSubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID6/LicLineID \ PAKdummyPak/PAK SIGN=0A5E3C90C67A INCREMENT LicUnityVoiceRecSessionsMax cisco 7.0 permanent 2 \ HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID7/LicLineID \ PAKdummyPak/PAK SIGN=12E962E6B592 INCREMENT LicServerVoiceRec cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID8/LicLineID \ PAKdummyPak/PAK SIGN=5C6FF1C641AE INCREMENT LicMaxMsgRecLenIsLicensed cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID9/LicLineID \ PAKdummyPak/PAK SIGN=573BA6B413B6 INCREMENT LicRegionIsUnrestricted cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID10/LicLineID \ PAKdummyPak/PAK SIGN=40EBACAE87D8 _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/201469229/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Adjust 3750 Egress Priority Queue Bandwidth
Bump - anyone have thoughts on this one? From: scarruthe...@hotmail.com To: ccie_voice@onlinestudylist.com Date: Sun, 14 Feb 2010 10:40:58 -0800 Subject: [OSL | CCIE_Voice] Adjust 3750 Egress Priority Queue Bandwidth I wanted some thoughts on how others would handle a request to tweak the amount of bandwidth availble to an egress priority queue on a 3750. So for example a request to allocate 25% of available bandwidth for switchports connected to IP phones on the 3750. I have heard suggestions to handle this in the following manner - this is assuming auto qos voip trust cisco-phone has been run on the port already: interface fa 1/0/2 no priority-queue out srr-queue bandwidth shape 4 0 0 0 srr-queue bandwidth share 0 33 33 33 But I'm struggling to see that this meets the requirement. In this configuration we would be enabling shaping of queue 1 and assigning it 25% of available bandwidth. Then assigning remaining bandwidth equally to the remaining three queues. But this does not appear to be meeting the requirement of assigning the priority queue 25% of the bandwidth. We would be assigning the queue that RTP traffic is placed in by default 25% of total bandwidth but the initial no priority queue out command technically disables a strict priority queue and thus it does not seem to fit the requirement. Thoughts? While I struggle to see the disablement of the priority queue as strictly meeting the requirement - I also find no explicit means to allocate the priority queue a strict amount of bandwidth (I.e. the equal if the ingress queue command - mls qos srr-queue input bandwidth 75 25 that could be used to meet this requirement for default priority ingress queue 1. How about skipping the initial no priority queue-out command but only issuing the shape and share commands as specified above? Wouldn't leaving the priority queue enabled and assigning it a shape value of 25% (1/4) satsify the requirement better? Thanks Scott Hotmail: Trusted email with powerful SPAM protection. Sign up now. _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469227/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Adjust 3750 Egress Priority Queue Bandwidth
I wanted some thoughts on how others would handle a request to tweak the amount of bandwidth availble to an egress priority queue on a 3750. So for example a request to allocate 25% of available bandwidth for switchports connected to IP phones on the 3750. I have heard suggestions to handle this in the following manner - this is assuming auto qos voip trust cisco-phone has been run on the port already: interface fa 1/0/2 no priority-queue out srr-queue bandwidth shape 4 0 0 0 srr-queue bandwidth share 0 33 33 33 But I'm struggling to see that this meets the requirement. In this configuration we would be enabling shaping of queue 1 and assigning it 25% of available bandwidth. Then assigning remaining bandwidth equally to the remaining three queues. But this does not appear to be meeting the requirement of assigning the priority queue 25% of the bandwidth. We would be assigning the queue that RTP traffic is placed in by default 25% of total bandwidth but the initial no priority queue out command technically disables a strict priority queue and thus it does not seem to fit the requirement. Thoughts? While I struggle to see the disablement of the priority queue as strictly meeting the requirement - I also find no explicit means to allocate the priority queue a strict amount of bandwidth (I.e. the equal if the ingress queue command - mls qos srr-queue input bandwidth 75 25 that could be used to meet this requirement for default priority ingress queue 1. How about skipping the initial no priority queue-out command but only issuing the shape and share commands as specified above? Wouldn't leaving the priority queue enabled and assigning it a shape value of 25% (1/4) satsify the requirement better? Thanks Scott _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469227/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues
Thanks Mike. I will email proctorlabs support and let you know what I get for a response. Has anyone else encountered similiar issues and been able to get resolution? As I said - I thought that proctorlabs/IP Expert were aware of Easy VPN issues but thought the problems were now rectified. From Mike's and my own experience it seems like problems that appears to be phone type centric remain. Thanks Scott Date: Sun, 7 Feb 2010 18:05:47 -0500 Subject: Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues From: 2xcci...@gmail.com To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com CC: 2xcci...@gmail.com Hey Scott, Yes, I have been having the same type of problems. The 7960s usually appear to work fine. If I use a 7961 or 7971 for whatever reason they have issues. For instance in some cases I can't call from hq to br1. The phones will ring, but when I pick, there is no audio. Even though there is no audio if you leave the phone off the hook eventually the call drops. In some cases when calling the PSTN the phones will reset when I pickup or try to put on hold the 7961/71s will state that CM is down. Or like you said, you will pick up the PSTN phone and the 7961 phone does not acknowledge that the phone has been picked up. It is a pain. I thought it was just me. When studying on the voice version 2 pods I did not have any issues, but I was not using 7971s or 7961s. Not sure if that makes a difference. Thanks, Mike On Sun, Feb 7, 2010 at 4:32 PM, scott carruthers scarruthe...@hotmail.com wrote: I haven't used a Proctorlabs session in a few weeks. In the past I had issues with the V3 easy VPN tunnels when using hardware phones. I believe there were updates sent out stating that many people were experiencing problems with the easy VPN solution but that the problems were not rectified. Are others having continued problems? Issues include: -Several phones in my lab register without issue, show line appearances, will go hook and receive dial tone, can dial PSTN numbers, but the call will not cut thru to two way audio - the phone will continue ringing after I attempt to pick it up on the PSTN phone. -This is not a GW, etc issue - some phones are fine - others are not - indentical configs and calling from same sites. -Not sure if this related to phone model by my lone 7960 is fine but 7970s experience issues. Thanks Scott Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469227/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Manipulating IP Phone Called Number Display
This is the goal: -From an HQ phone a call is placed to an international number -The DNIS is dialed as 9011343214 -The request is to allow the call to complete and after call completion the calling party's IP phone should be updated so that the access and international codes are stripped -In other words - HQ places a call to 9011343214 - once the call completes the HQ phone display should be updated to show the called party number as 343214 -Using called party transforms on the GW this is simple to do - any manipulation done via the transform will be updated on the originating IP phone -But I want a clean method of accomplishing this goal using traditional call routing. So for the call placed - in my topology - I am routing the call thru a route pattern of 9.011! - this points to a route list in which I am stripping predot - no transforms in play - traditional/old school call routing only. -What I expected to happen was the DNIS manipulation only thru the route pattern would affect the calling party's IP phone display. So if left as I have stated thus far - the display would remain as 9011343214. Thus I was of the belief if I wanted to accomplish the task I could tweak the route pattern to be 9011.! - strip it predot at the route pattern - solely to affect the display on the IP phone - and then change the route list to strip predot at the route overriding what was done at the route pattern and allowing the correct digits to be sent to the GW. I expected my goal would be accomplished with this methodology - the display would update correctly based on the maniplulation up to and thru the route pattern and would override those manipulations at the route list level for actual call routing. -What I seemed to experience - however - was that the DNIS manipulation at the route list level affected the display on the calling party's phone. So in the above example the IP phone would display 011343214 once connected and not 343214 as requested. Any thoughts? Have others played around with manipulating the calling party's called number display to reflect a specific requested number of digits without using dialed number transforms? The HQ GW in my example was H323 - during my next lab I want to see if this equation changes if it is instead MGCP and possibly in this scenario the changes only made thru the route pattern level and not thru the route list level would impact the display on the calling party's phone - but I wouldn't have expected there to be a difference in this functionality based on the signaling protocol. I accomplished the task by - in this case - manipulating the DNIS to 343214 at the route list level and then dealt with the digits that would be sent to the telco on the H.323 GW but I am trying to ascertain if this is expected behavior and the cleanest method of accomplishing the task. _ Hotmail: Free, trusted and rich email service. http://clk.atdmt.com/GBL/go/201469228/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues
I did not - I may have missed a note in these regards if one was sent out. Unfortunately I have been limping along (never had issues in V2) with less than recommended IOS due to hardware constraints. If I enter interface virtual-template1 I do not have a type tunnel option in the current IOS. What version is needed for this? May need to break down and finally get to recommended hardware/software versions if this has been known to cure such issues. Thanks Scott From: tsc...@ipexpert.com To: scarruthe...@hotmail.com; 2xcci...@gmail.com; 2xcci...@gmail.com; ccie_voice@onlinestudylist.com CC: wlaw...@ipexpert.com; dle...@ipexpert.com; vma...@ipexpert.com Subject: RE: [OSL | CCIE_Voice] Proctorlabs VPN Issues Date: Mon, 8 Feb 2010 11:00:38 -0500 Did you guys add the following to your configuration? interface Loopback0 ip address 10.10.100.15 255.255.255.255 ! interface Virtual-Template1 type tunnel ip unnumbered Loopback0 ! crypto ipsec client ezvpn IPx-Voice-vRack mode network-plus virtual-interface 1 Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Technical 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 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 From: Wayne Lawson [mailto:wlaw...@ipexpert.com] Sent: Monday, February 08, 2010 8:37 AM To: Drew LePla; Tyson Scott; Vik Malhi Cc: Wayne A. Lawson II Subject: Fwd: [OSL | CCIE_Voice] Proctorlabs VPN Issues Guys - I'd like to get together with all 3 of you. It appears that PLI problems are continuing and we need to get this figured out. Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. Begin forwarded message: From: scott carruthers scarruthe...@hotmail.com Date: February 8, 2010 6:52:23 AM EST To: 2xcci...@gmail.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues Thanks Mike. I will email proctorlabs support and let you know what I get for a response. Has anyone else encountered similiar issues and been able to get resolution? As I said - I thought that proctorlabs/IP Expert were aware of Easy VPN issues but thought the problems were now rectified. From Mike's and my own experience it seems like problems that appears to be phone type centric remain. Thanks Scott Date: Sun, 7 Feb 2010 18:05:47 -0500 Subject: Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues From: 2xcci...@gmail.com To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com CC: 2xcci...@gmail.com Hey Scott, Yes, I have been having the same type of problems. The 7960s usually appear to work fine. If I use a 7961 or 7971 for whatever reason they have issues. For instance in some cases I can't call from hq to br1. The phones will ring, but when I pick, there is no audio. Even though there is no audio if you leave the phone off the hook eventually the call drops. In some cases when calling the PSTN the phones will reset when I pickup or try to put on hold the 7961/71s will state that CM is down. Or like you said, you will pick up the PSTN phone and the 7961 phone does not acknowledge that the phone has been picked up. It is a pain. I thought it was just me. When studying on the voice version 2 pods I did not have any issues, but I was not using 7971s or 7961s. Not sure if that makes a difference. Thanks, Mike On Sun, Feb 7, 2010 at 4:32 PM, scott carruthers scarruthe...@hotmail.com wrote: I haven't used a Proctorlabs session in a few weeks. In the past I had issues with the V3 easy VPN tunnels when using hardware phones. I believe there were updates sent out stating that many people were experiencing problems with the easy VPN solution but that the problems were not rectified. Are others having continued problems? Issues include: -Several phones in my lab register without issue, show line appearances, will go hook and receive dial tone, can dial PSTN numbers, but the call will not cut thru to two way audio - the phone
Re: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display
Mark, No - there are no called party transforms. This is 100% traditional call routing. I do not have a single called party transform configured and this is intentional. So not a matter of the called party transform trumping. Thanks Scott From: mn...@netelligent.com To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 8 Feb 2010 13:38:12 -0600 Subject: RE: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display Scott, Called Xforms will trump all and match the called number at the point of it matching the route pattern. Do you still have your called xform configured? If so, you are matching on that thus changing the display of the phone. You mention “no transform in play”, but wasn’t sure if that meant configured? Mark Nigh From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of scott carruthers Sent: Monday, February 08, 2010 6:39 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display This is the goal: -From an HQ phone a call is placed to an international number -The DNIS is dialed as 9011343214 -The request is to allow the call to complete and after call completion the calling party's IP phone should be updated so that the access and international codes are stripped -In other words - HQ places a call to 9011343214 - once the call completes the HQ phone display should be updated to show the called party number as 343214 -Using called party transforms on the GW this is simple to do - any manipulation done via the transform will be updated on the originating IP phone -But I want a clean method of accomplishing this goal using traditional call routing. So for the call placed - in my topology - I am routing the call thru a route pattern of 9.011! - this points to a route list in which I am stripping predot - no transforms in play - traditional/old school call routing only. -What I expected to happen was the DNIS manipulation only thru the route pattern would affect the calling party's IP phone display. So if left as I have stated thus far - the display would remain as 9011343214. Thus I was of the belief if I wanted to accomplish the task I could tweak the route pattern to be 9011.! - strip it predot at the route pattern - solely to affect the display on the IP phone - and then change the route list to strip predot at the route overriding what was done at the route pattern and allowing the correct digits to be sent to the GW. I expected my goal would be accomplished with this methodology - the display would update correctly based on the maniplulation up to and thru the route pattern and would override those manipulations at the route list level for actual call routing. -What I seemed to experience - however - was that the DNIS manipulation at the route list level affected the display on the calling party's phone. So in the above example the IP phone would display 011343214 once connected and not 343214 as requested. Any thoughts? Have others played around with manipulating the calling party's called number display to reflect a specific requested number of digits without using dialed number transforms? The HQ GW in my example was H323 - during my next lab I want to see if this equation changes if it is instead MGCP and possibly in this scenario the changes only made thru the route pattern level and not thru the route list level would impact the display on the calling party's phone - but I wouldn't have expected there to be a difference in this functionality based on the signaling protocol. I accomplished the task by - in this case - manipulating the DNIS to 343214 at the route list level and then dealt with the digits that would be sent to the telco on the H.323 GW but I am trying to ascertain if this is expected behavior and the cleanest method of accomplishing the task. Hotmail: Free, trusted and rich email service. Get it now. This transmission and any attached files are privileged, confidential or otherwise the exclusive property of the intended recipient or Netelligent Corporation. If you are not the intended recipient, any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is strictly prohibited. If you have received this transmission in error, please contact us immediately by responding to this message or by telephone (314-392-6900) and promptly destroy the original transmission and its attachments. _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/201469229/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display
Mark, The configuration was on proctorlabs gear and is now gone but I can easily summarize. Recap of goal - after user makes an international call the calling party's IP phone display should be updated and show 343214. HQ Phone dials 9011343214 hits route pattern 9011.! within the route pattern I am stripping predot solely for the purpose of affecting the final display on the calling party phone call is sent to a route list that contain a single route group that contains the HQ H.323 GW the route list-route group DNIS manipulation strips predot and prefixes a 9011 - I want to send this as 9011! to the H323 GW so that I only need a single dial peer for CM/SRST functionality call is sent to the H323 GW. In this flow I would expect the originating IP phone to show the called number of 343214 or only the manipulation thru the route pattern. If this worked as I would have expected I would have met the goal. But instead the originating IP phone shows 9011343214. I can make this work without a problem - I went back and took the DNIS manipulation off of the route pattern - had the route list manipulate it to 343214 - and then have a dial peer on the GW prefix 011. But obviously not ideal - need another dial peer for SRST - and this just isn't how I expected it to function. Again I thought the rule was the phone will only reflect the manipulation up to and thru the route pattern but not what is done in the route list. Thanks Scott From: mn...@netelligent.com To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 8 Feb 2010 14:24:16 -0600 Subject: RE: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display Can you give us your configuration with regards to the Route Pattern and Route List Details? The behavior you are experiencing isn’t what I expect but rather what you expect is what I think I am thinking is correct, also. Mark Nigh From: scott carruthers [mailto:scarruthe...@hotmail.com] Sent: Monday, February 08, 2010 2:06 PM To: Mark Nigh; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display Mark, No - there are no called party transforms. This is 100% traditional call routing. I do not have a single called party transform configured and this is intentional. So not a matter of the called party transform trumping. Thanks Scott From: mn...@netelligent.com To: scarruthe...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 8 Feb 2010 13:38:12 -0600 Subject: RE: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display Scott, Called Xforms will trump all and match the called number at the point of it matching the route pattern. Do you still have your called xform configured? If so, you are matching on that thus changing the display of the phone. You mention “no transform in play”, but wasn’t sure if that meant configured? Mark Nigh From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of scott carruthers Sent: Monday, February 08, 2010 6:39 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Manipulating IP Phone Called Number Display This is the goal: -From an HQ phone a call is placed to an international number -The DNIS is dialed as 9011343214 -The request is to allow the call to complete and after call completion the calling party's IP phone should be updated so that the access and international codes are stripped -In other words - HQ places a call to 9011343214 - once the call completes the HQ phone display should be updated to show the called party number as 343214 -Using called party transforms on the GW this is simple to do - any manipulation done via the transform will be updated on the originating IP phone -But I want a clean method of accomplishing this goal using traditional call routing. So for the call placed - in my topology - I am routing the call thru a route pattern of 9.011! - this points to a route list in which I am stripping predot - no transforms in play - traditional/old school call routing only. -What I expected to happen was the DNIS manipulation only thru the route pattern would affect the calling party's IP phone display. So if left as I have stated thus far - the display would remain as 9011343214. Thus I was of the belief if I wanted to accomplish the task I could tweak the route pattern to be 9011.! - strip it predot at the route pattern - solely to affect the display on the IP phone - and then change the route list to strip predot at the route overriding what was done at the route pattern and allowing the correct digits to be sent to the GW. I expected my goal would be accomplished with this methodology - the display would update correctly based on the maniplulation up to and thru the route pattern and would override those manipulations at the route list level
Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues
Thanks Tyson - I will replace my 2621 which I cannot get 12.4.15 on and with a 831. I will let you know if I have continued problems after this is completed. Scott From: tsc...@ipexpert.com To: scarruthe...@hotmail.com; 2xcci...@gmail.com; ccie_voice@onlinestudylist.com CC: wlaw...@ipexpert.com; dle...@ipexpert.com; vma...@ipexpert.com Subject: RE: [OSL | CCIE_Voice] Proctorlabs VPN Issues Date: Mon, 8 Feb 2010 11:33:20 -0500 Scott, You have to run IOS 12.4T. So we implemented some fVRF configuration at proctorlabs to make pod to pod transition for students easier. (Which in the future will allow you guys to save and load configurations if we finish working out the logistics of that). With this change though our EZVPN implementation has moved to phase 8 and it seems thru more testing that the workaround below is required on the client end to match with the features we have implemented. If you have a router that can run at least 12.4(15)T then you don't have to change it out. I am using an 831 at home and it works great so you don't have to have the latest and greatest by any means. I would be happy to work with you on this if you would like to setup a time to do testing. Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Technical 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 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 From: scott carruthers [mailto:scarruthe...@hotmail.com] Sent: Monday, February 08, 2010 11:16 AM To: tsc...@ipexpert.com; 2xcci...@gmail.com; ccie_voice@onlinestudylist.com Cc: wlaw...@ipexpert.com; dle...@ipexpert.com; vik mahli Subject: RE: [OSL | CCIE_Voice] Proctorlabs VPN Issues I did not - I may have missed a note in these regards if one was sent out. Unfortunately I have been limping along (never had issues in V2) with less than recommended IOS due to hardware constraints. If I enter interface virtual-template1 I do not have a type tunnel option in the current IOS. What version is needed for this? May need to break down and finally get to recommended hardware/software versions if this has been known to cure such issues. Thanks Scott From: tsc...@ipexpert.com To: scarruthe...@hotmail.com; 2xcci...@gmail.com; 2xcci...@gmail.com; ccie_voice@onlinestudylist.com CC: wlaw...@ipexpert.com; dle...@ipexpert.com; vma...@ipexpert.com Subject: RE: [OSL | CCIE_Voice] Proctorlabs VPN Issues Date: Mon, 8 Feb 2010 11:00:38 -0500 Did you guys add the following to your configuration? interface Loopback0 ip address 10.10.100.15 255.255.255.255 ! interface Virtual-Template1 type tunnel ip unnumbered Loopback0 ! crypto ipsec client ezvpn IPx-Voice-vRack mode network-plus virtual-interface 1 Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Technical 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 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 From: Wayne Lawson [mailto:wlaw...@ipexpert.com] Sent: Monday, February 08, 2010 8:37 AM To: Drew LePla; Tyson Scott; Vik Malhi Cc: Wayne A. Lawson II Subject: Fwd: [OSL | CCIE_Voice] Proctorlabs VPN Issues Guys - I'd like to get together with all 3 of you. It appears that PLI problems are continuing and we need to get this figured out. Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. Begin forwarded message: From: scott carruthers scarruthe...@hotmail.com Date: February 8, 2010 6:52:23 AM EST To: 2xcci...@gmail.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Proctorlabs VPN Issues
[OSL | CCIE_Voice] Proctorlabs VPN Issues
I haven't used a Proctorlabs session in a few weeks. In the past I had issues with the V3 easy VPN tunnels when using hardware phones. I believe there were updates sent out stating that many people were experiencing problems with the easy VPN solution but that the problems were not rectified. Are others having continued problems? Issues include: -Several phones in my lab register without issue, show line appearances, will go hook and receive dial tone, can dial PSTN numbers, but the call will not cut thru to two way audio - the phone will continue ringing after I attempt to pick it up on the PSTN phone. -This is not a GW, etc issue - some phones are fine - others are not - indentical configs and calling from same sites. -Not sure if this related to phone model by my lone 7960 is fine but 7970s experience issues. Thanks Scott _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/201469230/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCP Localization - Missed Calls Directory
CUPC client plus dial question - number globalized coming in - localized when it hits CUPC - but the odd thing was the CUCP received/missed calls directory was then showing the localized ANI. so call from PSTN number 212 394 2123 - globalized to +12123942123 on HQ phone i localized it to 3942123 ANI but missed calls was also showing the localized number of 3942123 instead of the globalized full e.164 plus dial ANI i would have expected. have you ever noticed this? is it possible CUPC handles this differently than a 7962? and puts localized number in call directories? _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/196390709/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCP Localization - Missed Calls Directory
Device pool. The calling party transformation is working as intended - the number is globalized at the GW - and localized in the DP. If I remove the calling party transformation the ANI will show in the LCD display as globalized number and it will show in missed calls as globalized number (I.e. +12123942123). So everything is as I would expect expect that when the localization is applied it seems to be affecting the ANI that is populated in the missed calls dir (I.e. 3942123). Is this expected for some reason on a CUPC but a 7962 and all other hardware phones will display globalized number in missed/received calls? Not sure why that would make sense for the CUPC to be different. Thanks Scott Date: Fri, 15 Jan 2010 09:21:02 -0600 Subject: Re: [OSL | CCIE_Voice] CUCP Localization - Missed Calls Directory From: darylpsm...@gmail.com To: scarruthe...@hotmail.com Did you apply the transformation to the phone or the device pool? DPS There are no secrets to success. It is the result of preparation, hard work, and learning from failure “PASSING VOICE CCIE EXAM” Week of Jan 25th-29th!!! On 1/15/10 9:07 AM, scott carruthers scarruthe...@hotmail.com wrote: CUPC client plus dial question - number globalized coming in - localized when it hits CUPC - but the odd thing was the CUCP received/missed calls directory was then showing the localized ANI. so call from PSTN number 212 394 2123 - globalized to +12123942123 on HQ phone i localized it to 3942123 ANI but missed calls was also showing the localized number of 3942123 instead of the globalized full e.164 plus dial ANI i would have expected. have you ever noticed this? is it possible CUPC handles this differently than a 7962? and puts localized number in call directories? Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. http://clk.atdmt.com/GBL/go/196390709/direct/01/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/196390710/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] IP Communicator And Plus Dialing
Has anyone attempted to use IP communicator while configuring E.164 (plus dial) labs? My configuration appears to be correct to mask ANI for internal IP phone to IP phone calls – sending calls thru a xlation pattern in which I have use external phone number mask selected – but the call arrives at a registered IP communicator in branch 1 with the 4 digit DN as the ANI. Before digging into this deeper I want to ensure the IP communicator is capable of supporting the full E.164 number (+14158881002). I found the following quote in Cisco documentation but I cannot find a setting to flip on in the communicator device config/service parameters/etc. *Dial Using Cisco IP Communicator Mode and E. 164 Dialing * If users will use the Dial Using Cisco IP Communicator option to dial numbers in E.164 format, ensure that the Cisco Unified Communications Manager administrator configures Cisco IP Communicator to process E. 164 dialing. Otherwise, calls to these phone numbers will fail. Any ideas? Thanks Scott ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
