Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Dear Hesham, As far as I understand from your email, SiteB is now in SRST mode, which means that SiteB WAN connection is down. In this case, SiteC won't be able to reach SiteB phones over the WAN through GK but you will have to configure a lower preference dial-peer to reach it through PSTN in case the GK rejects or can't reach SiteB phones. Thanks,Ramy Date: Sat, 22 Jun 2013 19:45:41 -0700 From: heshamcentr...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister Dear Experts, SiteC is CME and connected with HQ and SB via GatekeeperGatekeeper is working excellent with HQ and SB I am configuring Call Forward Unregister for SiteB. SiteB has Call-Manager-Fallback mode working excellent Now, I have configured Call Forward Unregisterin the service parameter I changed maximum hops to DN unregister is 1 I have Created a Partitions and CSS for CFURI forward SiteB1 and SiteB2 telephones in unregisted internal and external to be 9723033001 with forward css CFUR-CSS I created Route List to point to HQ Router and create route pattern for CFUR Now gatekeeper is reaching both HQ and SiteB in normal operaitonwhen I put SiteB under call-manager-fallback modewhen I dial from HQ 3001 the CFUR works and shows the E164 number when I dial from SiteC 3001 via gatekeeper it shows unknown number knowing that Gatekeeper is working with SiteB under normal operation but doesn't work with CFUR Any Ideas, Thanks,Hesham ___ 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] Corporate directory issue...
I believe the simplest way is as follows: Change Service Provisioning parameter to External URL on the phone you want to disable Directories. Go to Enterprise Parameters - remove Directories URL (leave it blank) and add Messages URL as (http://10.10.210.10:8080/ccmip/getmessagesmenu.jsp). Where 10.10.210.10 is Publisher. Reset all IP phones. After resetting, once you press Directories button on that phone it'll display No Services Configured. Also, make sure that Directories are displayed on all the other phones as well as the Messages button. Hope this makes it clear. Thanks, Ramy Date: Fri, 12 Oct 2012 15:06:41 -0500 From: whl...@gmail.com To: daqui...@cisco.com CC: ccie_voice@onlinestudylist.com; vir...@rediffmail.com Subject: Re: [OSL | CCIE_Voice] Corporate directory issue... Dan's method will change the order (alphabetical I think) that the service appear on the phone so if order does not matter this method will work. Kevin's method will work well but as he says you need to practice with it to see all the various results Now if those two methods are not enough, say you have to make a custom order or message, then you will need to reference this website: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/all_models/xsi/7_0/english/programming/guide/xsi70xml.html Use this website to create your custom configuration as an xml file, then you will have to upload it to a web server of some kind, your choices in the lab are somewhat limited, and then put in the proper place to have it served up to the phone when it asks for it. Hope this helps On Fri, Oct 12, 2012 at 2:46 PM, Dan Quinlan (daquinla) daqui...@cisco.com wrote: I'd: 1) disable all of the enterprise Directory Services (missed call, placed call, etc) 2) copy each of the services and activate them as Non-Enterprise subscription services 3) create a phone template that is subscribed to each of the newly copied services 4) BAT subscribe all phones but the 1 to the new versions of the services DQ Dan Quinlan, CCIE #36129 daqui...@cisco.com On Oct 12, 2012, at 2:04 PM, Kevin Spicer ke...@kevinspicer.co.uk wrote: You need to understand the different settings for 'Services Provisioning' on the phones and the impact of setting the URLs in enterprise parameters to different values (including invalid ones). Two tips: 1) You need to be aware of the messages URL when you're using external services provisioning. http://x.x.x.x:8080/ccmcip/getmessagesmenu.jsp 2) This stuff only works on recent model phones (7945 / 7965 definitely, I suspect also the 7941/61 but not the 7940/7960) One of those things that is best understood by spending a couple of hours trying the different combinations! On Fri, Oct 12, 2012 at 5:06 PM, virajith vir...@rediffmail.com wrote: hi Guys, If we want only 1 phone to display No services Configured when pressing corporate directory button. What is the easiest way to achieve this config? Could someone guide me with the steps? -Vir From: ccie_voice-requ...@onlinestudylist.com Sent: Wed, 10 Oct 2012 21:31:21 To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 80, Issue 14 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. *** MTP Understanding. *** quote To MTP or (Justin McIntyre) 2. Adding Cisco 7926 in UCM 8.6 (Asad Yasin) 3. Re: Adding Cisco 7926 in UCM 8.6 (Mohammed Nabelsi) -- Message: 1 Date: Tue, 9 Oct 2012 13:35:48 -0400 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] *** MTP Understanding. *** quote To MTP or Message-ID: 0a566999c353b042841473fa64165cd64364281...@exchcluster.corp.bbns.com Content-Type: text/plain; charset=Windows-1252 Think about it this way. When you place hold you are telling the opposite phone/gateway to listen to a new RTP stream. So where is that new RTP stream being sourced from. If it's coming from your PUB/SUB then what device pool is that MOH server in and what MRGL/group is within that Device Pool? If it is structured such that it would be definitive to use the MTP then yes you would also be terminating that RTP stream to the MTP as well. Just a side note , which I'm sure you're already aware of, the PUB and SUB built in software MTPs only support g711. Hope this helps. Thanks, Justin McIntyre This email and any files transmitted with it are
Re: [OSL | CCIE_Voice] CUPC 8.X compatible with 7.X
Hi Nehal, I didn't use this version but if you didn't get the answer on the forum then it's better to confirm that through the compatibility matrix document on Cisco website. Regards,Ramy From: nehal.ah...@msn.com To: ccie_voice@onlinestudylist.com Date: Sat, 18 Aug 2012 00:17:51 +0500 Subject: [OSL | CCIE_Voice] CUPC 8.X compatible with 7.X Hi All, I have windows 7 and CUPC version 8.6.2.XX installed. Will it fully work with the proctorlabs CUCM,UNITY Presence versions ? Thx Nehal ___ 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 - DEAD air
If you have RSVP between SA and SC, make sure that the transcoder at SC is above the MTP in the MRGL assigned to SC. Regards,Ramy From: rrcr...@yahoo.com Date: Wed, 15 Aug 2012 17:26:31 -0700 To: brun...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUE - DEAD air I reloaded CUE and since The tutorial was on so I created a greeting. Sometime after that it started working IdkRc Sent from my iPhone On Aug 15, 2012, at 5:19 PM, Bruno Nonogaki brun...@gmail.com wrote: Have you tried no sccp / sccp? I have already gotten some issues with audio, and it was a bug in the media resources. On Wed, Aug 15, 2012 at 2:37 PM, Randall Crumm rrcr...@yahoo.com wrote: Hello,When I call sc ph2 from another site I am getting dead air once it drops to VM. If I press the messages button from sc ph2 I hear the correct greeting. CUCM -- CUE intergration I see the CTI port on the screen of the calling phone. Any thoughts? 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 ___ 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] sip srst
Voice service voip Sip Bind all source interface vlan240 Registrar server ! Voice register global Mode srst Source-address 10.10.202.1 port 5060 Max-dn 1 Max-pool 1 Create profile ! Voice register pool 1 Id network 10.10.201.0 mask 255.255.255.0 Codec g711ulaw Dtmf-relay rtp-nte ! Regards,Ramy Date: Fri, 17 Aug 2012 17:24:41 -0700 From: jonha...@yahoo.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] sip srst any can sent me a sample sip srst for sip phones? ___ 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] ntp master- is it necessary
No problem at all :-) Ramy Date: Mon, 13 Aug 2012 20:31:42 -0700 From: vinayak_...@yahoo.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary To: ramyoth...@hotmail.com; daqui...@cisco.com CC: ccie_voice@onlinestudylist.com; michael.se...@compucom.com ramy, i apologize for not understanding the question properly... i overlooked at your question.. krishna. From: Ramy Abdelrahim ramyoth...@hotmail.com To: vinayak_...@yahoo.com; daqui...@cisco.com Cc: ccie_voice@onlinestudylist.com; michael.se...@compucom.com Sent: Monday, August 13, 2012 9:08 PM Subject: RE: [OSL | CCIE_Voice] ntp master- is it necessary Krishna, My question was apart from yours, I was trying to sync one of the gateways (Site-B) with its internal clock just for testing to see if it'll provide the same result as when synched with external source. Dan advised that this can be done with using the ntp master command on Site-B GW, just configure the 2 clock commands. Regards,Ramy Date: Mon, 13 Aug 2012 18:50:45 -0700 From: vinayak_...@yahoo.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary To: daqui...@cisco.com; ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com; michael.se...@compucom.com dan Ramy, the question here is hq router is the one that provides the time to ucm..and therefore without putting the command ntp source how can cucm can know it has to synchronize its clock with hq router...i m sure that when you providing time from hq router to cucm phones, then you must use ntp master as well ntp source to which the time on the hq router binds and apparently the cucm can reference the clock from hq router as well... thank youkrishna From: Dan Quinlan (daquinla) daqui...@cisco.com To: Ramy Abdelrahim ramyoth...@hotmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; michael.se...@compucom.com michael.se...@compucom.com Sent: Monday, August 13, 2012 7:33 PM Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary If it is UCM, then the time on the router is only relevant to the phones when in SRST. When registered to the UCM, the phones are getting their time information from the UCM, specifically the date time group configured on the device pool that contains those phones. DQ d...@cisco.com Sent from my iPhone On Aug 13, 2012, at 8:26 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: I think in your scenario you're assuming that this is a CME site. Actually, this site is a UCM remote site not CME. So, in this case do you think the 2 clock commands are sufficient? Regards, Ramy From: daqui...@cisco.com To: ramyoth...@hotmail.com CC: michael.se...@compucom.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Date: Mon, 13 Aug 2012 23:51:17 + You don't need the ntp master or the ntp source commands. Just the two clock commands that you have, the time-zone command under telephony service, and make sure you set the type command under the ephone (eg type 7965). Do a create cnf under telephony service and restart or reset the ephones. DQ d...@cisco.com Sent from my iPhone On Aug 13, 2012, at 6:04 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Dear All, Regarding the same subject, I tried to sync a gateway with its internal clock using ntp master only. The clock was synched but not as what's displayed on the IP phones at that site. I realized also that the stratum was higher than 10. My configuration was as follows: clock timezone EST -5 clock summer-time EDT recurring ntp master I don't know if there's a configuration missing here or not. Regard, Ramy From: michael.se...@compucom.com To: daqui...@cisco.com Date: Sun, 5 Aug 2012 13:32:39 -0400 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Thanks Dan. I'm not trying to make it work yes it works fine without the command, but instead I'm trying to replicate what Cisco might be looking for on the lab. Frankly I don't know what they are looking for in these types of scenarios but don't want to take the chance of losing points because they think the command should be there. Just cause it works doesn't mean you will get the points. Michael Sears -Original Message- From: Dan Quinlan (daquinla) [mailto:daqui...@cisco.com] Sent: Sunday, August 05, 2012 11:27 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Michael, all, You do NOT need the NTP master command on a router whenever that router is synchronizing with an external source, even if other devices are to synchronize with that router. Michael - in your config, you can remove the NTP master command and everything will synchronize with the HQ router just fine. Ntp master should really only be used when
Re: [OSL | CCIE_Voice] ntp master- is it necessary
Dear All, Regarding the same subject, I tried to sync a gateway with its internal clock using ntp master only. The clock was synched but not as what's displayed on the IP phones at that site. I realized also that the stratum was higher than 10. My configuration was as follows: clock timezone EST -5clock summer-time EDT recurringntp master I don't know if there's a configuration missing here or not. Regard,Ramy From: michael.se...@compucom.com To: daqui...@cisco.com Date: Sun, 5 Aug 2012 13:32:39 -0400 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Thanks Dan. I'm not trying to make it work yes it works fine without the command, but instead I'm trying to replicate what Cisco might be looking for on the lab. Frankly I don't know what they are looking for in these types of scenarios but don't want to take the chance of losing points because they think the command should be there. Just cause it works doesn't mean you will get the points. Michael Sears -Original Message- From: Dan Quinlan (daquinla) [mailto:daqui...@cisco.com] Sent: Sunday, August 05, 2012 11:27 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Michael, all, You do NOT need the NTP master command on a router whenever that router is synchronizing with an external source, even if other devices are to synchronize with that router. Michael - in your config, you can remove the NTP master command and everything will synchronize with the HQ router just fine. Ntp master should really only be used when there is no external clock. DQ d...@cisco.com Sent from my iPhone On Aug 5, 2012, at 12:38 PM, michael.se...@compucom.com michael.se...@compucom.com wrote: The only situation in which I use the ntp master command is in a situation where for example HQ is providing clock for Servers and HQ clock is synchronizing with an external reliable source. In my lab I synchronize my PSTN with the Boulder Atomic Clock and synchronize my HQ router with the PSTN. HQ loopback provides clock for Servers, Switch, CUE and other branches. The following is what I use to accomplish this. Other than this scenario I do not use the ntp master command. ntp server 10.1.1.1 ntp source loopback0 ntp master Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, August 05, 2012 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 78, Issue 14 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: Switch QOS query (Justin McIntyre) 2. Re: ntp master- is it necessary (Bruno Nonogaki) 3. Re: Switch QOS query (Bruno Nonogaki) 4. Re: ntp master- is it necessary (Justin McIntyre) -- Message: 1 Date: Sun, 5 Aug 2012 09:10:00 -0400 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Switch QOS query Message-ID: 0aed4c84-05d7-4f60-b91f-eeca67495...@blackbox.com Content-Type: text/plain; charset=us-ascii So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your requirements. Have the priority-queue enabled but then also give queue 1 30% bandwidth. If priority-queue out is enabled then this over-rides the bandwidth command for that queue. I know you had some other questions as well specifically about how to drop certain traffic if a queue were 80% full. My suggestion to you would be to review Vik Mahlis QOS blog on the IPEXPERT website. Go to blog.ipexpert.com and select the voice blog on the left. Then look for the QOS section. I think this will clear up most of your questions and get you on your way. Thanks, Justin McIntyre This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended
Re: [OSL | CCIE_Voice] ntp master- is it necessary
I think in your scenario you're assuming that this is a CME site. Actually, this site is a UCM remote site not CME. So, in this case do you think the 2 clock commands are sufficient? Regards,Ramy From: daqui...@cisco.com To: ramyoth...@hotmail.com CC: michael.se...@compucom.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Date: Mon, 13 Aug 2012 23:51:17 + You don't need the ntp master or the ntp source commands. Just the two clock commands that you have, the time-zone command under telephony service, and make sure you set the type command under the ephone (eg type 7965). Do a create cnf under telephony service and restart or reset the ephones. DQ d...@cisco.com Sent from my iPhone On Aug 13, 2012, at 6:04 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Dear All, Regarding the same subject, I tried to sync a gateway with its internal clock using ntp master only. The clock was synched but not as what's displayed on the IP phones at that site. I realized also that the stratum was higher than 10. My configuration was as follows: clock timezone EST -5 clock summer-time EDT recurring ntp master I don't know if there's a configuration missing here or not. Regard, Ramy From: michael.se...@compucom.com To: daqui...@cisco.com Date: Sun, 5 Aug 2012 13:32:39 -0400 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Thanks Dan. I'm not trying to make it work yes it works fine without the command, but instead I'm trying to replicate what Cisco might be looking for on the lab. Frankly I don't know what they are looking for in these types of scenarios but don't want to take the chance of losing points because they think the command should be there. Just cause it works doesn't mean you will get the points. Michael Sears -Original Message- From: Dan Quinlan (daquinla) [mailto:daqui...@cisco.com] Sent: Sunday, August 05, 2012 11:27 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Michael, all, You do NOT need the NTP master command on a router whenever that router is synchronizing with an external source, even if other devices are to synchronize with that router. Michael - in your config, you can remove the NTP master command and everything will synchronize with the HQ router just fine. Ntp master should really only be used when there is no external clock. DQ d...@cisco.com ! gt; Sent from my iPhone On Aug 5, 2012, at 12:38 PM, michael.se...@compucom.com michael.se...@compucom.com wrote: The only situation in which I use the ntp master command is in a situation where for example HQ is providing clock for Servers and HQ clock is synchronizing with an external reliable source. In my lab I synchronize my PSTN with the Boulder Atomic Clock and synchronize my HQ router with the PSTN. HQ loopback provides clock for Servers, Switch, CUE and other branches. The following is what I use to accomplish this. Other than this scenario I do not use the ntp master command. ntp server 10.1.1.1 ntp source loopback0 ntp master Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, August 05, 2012 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 78, Issue 14 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: Switch QOS query (Justin McIntyre) ! 2.. Re: ntp master- is it necessary (Bruno Nonogaki) 3. Re: Switch QOS query (Bruno Nonogaki) 4. Re: ntp master- is it necessary (Justin McIntyre) -- Message: 1 Date: Sun, 5 Aug 2012 09:10:00 -0400 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Switch QOS query Message-ID: 0aed4c84-05d7-4f60-b91f-eeca67495...@blackbox.com Content-Type: text/plain; charset=us-ascii So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your
[OSL | CCIE_Voice] CUPS - Softphone vs Deskphone mode
Dear All, In Workbook vol2, Lab 4 question 1.2, it's requested to configure CUPC as softphone which I did but still I can go to deskphone mode on the CUPC. Please note that the user is not assigned deskphone capabilities in presence. I configured another CUPC user as Deskphone mode then I tried to switch to softphone and off course it didn't work as expected because there was no UPCXX added in the Device list The question now is:- is it normal for a CUPC user configured as softphone (as requested in the question) to be able to switch and work in Deskphone mode as well? Regards,Ramy ___ 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] CUPS - Softphone vs Deskphone mode
That's right. I forgot that and I assigned a CTI Gateway profile to the softphone user. Thanks Kevin. Regards,Ramy Date: Thu, 9 Aug 2012 18:20:18 +0100 Subject: Re: [OSL | CCIE_Voice] CUPS - Softphone vs Deskphone mode From: ke...@kevinspicer.co.uk To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com This is because you have assigned a CTI profile to the CUPC user in Cups. The deskphone control settings are not relevant to CUPC users. On 9 Aug 2012 18:09, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Dear All, In Workbook vol2, Lab 4 question 1.2, it's requested to configure CUPC as softphone which I did but still I can go to deskphone mode on the CUPC. Please note that the user is not assigned deskphone capabilities in presence. I configured another CUPC user as Deskphone mode then I tried to switch to softphone and off course it didn't work as expected because there was no UPCXX added in the Device list The question now is:- is it normal for a CUPC user configured as softphone (as requested in the question) to be able to switch and work in Deskphone mode as well? Regards, Ramy ___ 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] New Labs - Lab 2
Hi Randall, Please make sure that the Publisher hostname is matching what's configured in the UCM under System Unified CM. Regards,Ramy Date: Thu, 9 Aug 2012 12:16:08 -0700 From: rrcr...@yahoo.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] New Labs - Lab 2 Hello,I am working on lab 2 and the presence server is having an issue which I believe is related to being a VM.The problem I am having is on the SYSTEM\CUCM Publisher menu. The servers sync but failing to conect to the CUCM database. Here is the error: Sync Completed, but currently failed to connect to the CUCM Database Monitor. Retrying... 2012-08-09 19:12:26 Is there some command I need to run on the Presence CLI? 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] New Labs - Lab 2
Try to check the AXL username and password in CUPS and confirm that the password in CUPS and UCM match. The following fields in CUPS should be matching what's configured in UCM: 1. CUCM Publisher Hostname2. CUCM Publisher IP Address3. AXL username4. AXL Password Regards,Ramy CC: ccie_voice@onlinestudylist.com From: rrcr...@yahoo.com Subject: Re: [OSL | CCIE_Voice] New Labs - Lab 2 Date: Thu, 9 Aug 2012 13:24:20 -0700 To: ramyoth...@hotmail.com I changed the host name to the ip address and on the topology page and started the services Rc Sent from my iPhone On Aug 9, 2012, at 12:46 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Hi Randall, Please make sure that the Publisher hostname is matching what's configured in the UCM under System Unified CM. Regards,Ramy Date: Thu, 9 Aug 2012 12:16:08 -0700 From: rrcr...@yahoo.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] New Labs - Lab 2 Hello,I am working on lab 2 and the presence server is having an issue which I believe is related to being a VM.The problem I am having is on the SYSTEM\CUCM Publisher menu. The servers sync but failing to conect to the CUCM database. Here is the error: Sync Completed, but currently failed to connect to the CUCM Database Monitor. Retrying... 2012-08-09 19:12:26 Is there some command I need to run on the Presence CLI? 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] MVA -Your call can not be completed as dialed
Guys, Re-routing CSS is used for SNR but CSS is used for MVA. Therefore, you've to check if the number you're dialing is in a route pattern that the configured CSS in the remote destination profile can reach. I recommend to separate the dial plan for MVA or SNR as follows. - Create a new partition.- Create a new CSS containing the above new partition.- Create a new route list- Create a new route pattern. hope this will help. Regards,Ramy Date: Tue, 7 Aug 2012 16:21:17 -0400 From: ccielab...@gmail.com To: vipji...@cisco.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed b Thank you Vipul. I thought re-routing CSS was used only for the SNR configuration. I will take a look and test again. :) On Tue, Aug 7, 2012 at 3:29 PM, Vipul Jindal (vipjinda) vipji...@cisco.com wrote: It uses the re routing CSS on the remote destination number. If you check the call manager traces, you can easily check it. From: ccielabrat ccielab...@gmail.com Date: Tuesday, August 7, 2012 2:09 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed To All, I'm hoping the group can help me understand the call flow for an MVA call. I'm able to call into the MVA pilot number , have my remote destination number recognized and be prompted for my PIN and to dial . But I get the message Your call can not be completed as dialed for anything I try to call. I understand that the number configured under the mobile voice access page is used as an anchor , as per Vik's vlecture, but I'm unclear what device is referenced regarding CSS and what should and should be reachable. Can anyone please help get closure on this last piece of the puzzle. -Lab Rat ___ 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] iDivert/DND in SRST
Hi Steffen, Even this workaround requires Unified CME as SRST but my configuration is using SRST (call-manager-fallback). Even DND is not there when the phone register to SRST GW. Regards,Ramy Date: Tue, 7 Aug 2012 22:40:01 +0200 Subject: Re: [OSL | CCIE_Voice] iDivert/DND in SRST From: stbruen...@gmail.com To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi Ramy, does the question say that you should preserve this? There is no iDivert softey in CME/SRST 7.X. iDivert is supported for SIP Phones in CME with 8.5 or so but not for phones falling back to SRST. You can only preserve the feature with workarounds like this: Transfer a call directly into cue mailboxhttp://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_tech_note09186a00802ab979.shtml RegardsSteffen 2012/8/7 Ramy Abdelrahim ramyoth...@hotmail.com Dear All, When the phone is registered to UCM it has iDivert softkey button to transfer a call to VM while ringing. When this site goes into SRST, iDivert is not there. Do I have to preserve this feature in SRST? And if it's the case then how? Can anyone help on this? Regards,Ramy ___ 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] iDivert/DND in SRST
But CFNA doesn't require any intervention from the Ip phone user. I tried to find a way to add DND to call-manager-fallback but I couldn't. Regards,Ramy From: daqui...@cisco.com To: daqui...@cisco.com CC: ramyoth...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] iDivert/DND in SRST Date: Tue, 7 Aug 2012 20:37:19 + I should say CFNA. DQ d...@cisco.com Sent from my iPhone On Aug 7, 2012, at 4:31 PM, Dan Quinlan (daquinla) daqui...@cisco.com wrote: If you are required to preserve idivert, you can achieve similar functionality using a DND softkey in SRST. When a call is ringing, pressing DND should forward the call to the RNA number. DQ d...@cisco.com Sent from my iPhone On Aug 7, 2012, at 4:24 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Dear All, When the phone is registered to UCM it has iDivert softkey button to transfer a call to VM while ringing. When this site goes into SRST, iDivert is not there. Do I have to preserve this feature in SRST? And if it's the case then how? Can anyone help on this? Regards, Ramy ___ 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] MVA -Your call can not be completed as dialed
Dear, No, it's not ignored when dialing through MVA. In this situation we have to differentiate between Mobile Connect (Single Number Reach) and Mobile Voice Access (MVA). Rerouting CSS - used when someone calls extension 5000 and we want to ring out to cell phone of user (5000). CSS (under Remote Destination Profile) - used when the user dial in from his cell phone, enter his pin and select 1 to make a call then the number he enter will be routed based on the CSS NOT the reroting CSS. Regards,Ramy Date: Tue, 7 Aug 2012 17:07:00 -0400 Subject: Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed From: ccielab...@gmail.com To: ramyoth...@hotmail.com CC: vipji...@cisco.com; ccie_voice@onlinestudylist.com Ramy, Can you clarify how the re-routing CSS is used? Is the regular CSS ignored when a remote destination phone is dialing through MVA? -Scott On Tue, Aug 7, 2012 at 4:34 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Guys, Re-routing CSS is used for SNR but CSS is used for MVA. Therefore, you've to check if the number you're dialing is in a route pattern that the configured CSS in the remote destination profile can reach. I recommend to separate the dial plan for MVA or SNR as follows. - Create a new partition.- Create a new CSS containing the above new partition.- Create a new route list- Create a new route pattern. hope this will help. Regards,Ramy Date: Tue, 7 Aug 2012 16:21:17 -0400 From: ccielab...@gmail.com To: vipji...@cisco.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed b Thank you Vipul. I thought re-routing CSS was used only for the SNR configuration. I will take a look and test again. :) On Tue, Aug 7, 2012 at 3:29 PM, Vipul Jindal (vipjinda) vipji...@cisco.com wrote: It uses the re routing CSS on the remote destination number. If you check the call manager traces, you can easily check it. From: ccielabrat ccielab...@gmail.com Date: Tuesday, August 7, 2012 2:09 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed To All, I'm hoping the group can help me understand the call flow for an MVA call. I'm able to call into the MVA pilot number , have my remote destination number recognized and be prompted for my PIN and to dial . But I get the message Your call can not be completed as dialed for anything I try to call. I understand that the number configured under the mobile voice access page is used as an anchor , as per Vik's vlecture, but I'm unclear what device is referenced regarding CSS and what should and should be reachable. Can anyone please help get closure on this last piece of the puzzle. -Lab Rat ___ 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] MVA Hangs Up
Please make sure that Redirecting Number IE Delivery in GW page is checked for Inbound calls. Thanks,Ramy Abdelrahim Date: Sun, 6 May 2012 08:13:46 -0400 From: feard...@trinity-health.org To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Hangs Up Any ideas? The MVA application receives the call on the H323 gateway and sends to IVRMainpage.vxml. The call is prompted for PIN and Press 1 to make a call. When the desired number is entered, followed by a # there is an immediate hang up. The Remote Destination Profile CSS has visibility to the number being dialed. What might be the problem or how do I troubleshoot? Thanks, Joe ___ 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] Router Switch port configurations
Hi Guys, Just to cut it short .. If you're on HQ-3750 then configure the phone ports as access ports and if you're configuring a switch module on a voice gateway then make it as trunk port. HQ-3750: int f1/0/1 switchport access vlan 10 switchport mode access switchport voice vlan 20 spanning-tree portfast Switch Module: int f1/1 switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 200 BRgds, Ramy Date: Tue, 24 Apr 2012 05:49:41 +0100 From: ke...@kevinspicer.co.uk To: devsin2...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Router Switch port configurations Actually Chris Bill's post has me worried. How many other people have seen this behavior? There's not a lot to choose between the 2, and if one is more stable... On 24 Apr 2012 04:22, Chris devsin2...@gmail.com wrote: Thanks Juan. Hi ST, What you are saying is not quiet correct. Option 2 will normally be okay. However, if we want to be sure, we could include allowed vlan keyword. This option must always work at least when you have allowed VLAN configured, and vlan pruning is disabled. The switchport voice vlan is Cisco's special for Cisco phone. But some other vendors have also taken up this approach. I my opinion if we can use it option 1 we should use it. From responses from Kevin and Juan, it seems like it works fine. I think we can challenge if this doesn't work in lab. I was originally unsure of if this command would be available on piddly little four port switch module. Chris. On Mon, Apr 23, 2012 at 11:45 PM, Seifeddine Tlili seifeddine.tl...@lvs1.com wrote: But in option 2 you still missing the switchport voice vlan 11 ortherwise you have to specify it manually on each phone, for third party SIP Phone it`s has to be that way Thanks ST From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Juan Lopez Sent: Monday, April 23, 2012 6:33 AM To: Chris Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Router Switch port configurations I always use the second option, never had any issue with it... cheers Juan Op 22 april 2012 15:56 schreef Chris devsin2...@gmail.com het volgende: Hi All , I don't have a 4-port or 9-port POE switch module to try it on. Therefore I would like for some one to confirm if both or one of following port configuration will work on these cards. I do understand the concept, but don't want to find the actual syntax on lab day :). Thanks in advance. Vlan 10 is DATA Vlan 11 is VOICE Preference 1- interface FastEthernet0/1/0 switchport access vlan 10 switchport voice vlan 11 spanning-tree portfast Preference 2- interface FastEthernet0/1/0 switchport trunk native vlan 10 switchport mode trunk Best Regards Chris ___ 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] MVA issue playing the prompts
Hi Juan, Since it's now tomorrow in my region, I think we should say something about this issue :-)) I assume that you've an H323 GW at HQ. If so, please remove both dial-peers and replace with the following one and specify the codec to be g711ulaw. dial-peer voice 2220 pots service mva incoming called-number 2220 dtmf-relay h245-alphanumeric codec g711ulaw no vad Also, make sure that when you configured the MVA service on the HQ router that it has been loaded successfully. application service mva http://177.1.10.10:8080/ccmivr/pages/IVRMainoage.vxml Once you hit enter the router will display a message that it was read successfully. Thanks, Ramy CC: kew...@gmail.com; ccie_voice@onlinestudylist.com From: lopez.hernandez.j...@gmail.com Subject: Re: [OSL | CCIE_Voice] MVA issue playing the prompts Date: Sun, 22 Apr 2012 00:11:32 +0200 To: ramyoth...@hotmail.com Hi Ramy, I did a reboot of everything in the end - no success. Not sure what is going in this time, traces don't show why I dont hear nothing., or I am not looking at the right things... Tomorrow another day :) ? On 21 Apr 2012, at 21:31, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Hi Juan, Did you try to restart the MVA service on the PUB? Thanks Ramy Date: Sat, 21 Apr 2012 14:28:02 +0200 From: lopez.hernandez.j...@gmail.com To: kew...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA issue playing the prompts thanks - did remove the direct inward dial, checked the complete match. Still no prompt to be heard passed the initial welcome to cisco unified comms ' cheers, Juan Op 21 april 2012 12:00 schreef Ken Wyan kew...@gmail.com het volgende: Remove direct-inward-dial from dial-peer 100 In service parameters MVA destination should be complete match (default) Thanks On Sat, Apr 21, 2012 at 2:27 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: Dear all, redoing some lab testing today, and for some odd reason :) I am not able to have MVA playing the prompts. I call in into the MVA number. I hear the lady in the box telling me I'm welcome to Cisco Unified Comms. Then it goes silent apart from that it is working: while no one asks me - enter the user's PIN code 12345, press 1, then an internal number in the none partition, it will connect to that number. Cheking the debug vxml on the router does not tell much. Then checking the trace files either: the trace steps regarding playing the first prompt - which I believe is the one asking for the PIN: controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 5.au and locale = en_US controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 5.au and locale = en_US controller.IVRGetAudioFile - [CCM_IVR]:: getting the file now so no errors, but nothing played. Can anyone help me with this, I cannot seem to find what is wrong. To make sure, I placed the router in the HQ device pool - so it speaks g711 with the UCM -as I am not sure the UCM will be able to play these prompts in g729 too. On the router it's: dial-peer voice 100 pots service mva incoming called-number 2220 direct-inward-dial ! dial-peer voice 1001 voip destination-pattern 2220 voice-class codec 1 session target ipv4:177.1.10.10 dtmf-relay h245-alphanumeric no vad with the voice-class codec speaking g711u/g729r8 PS/ reboot of servers did not help. any help is much appreciated ! Juan ___ 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] MVA issue playing the prompts
Hi Juan, Did you try to restart the MVA service on the PUB? Thanks Ramy Date: Sat, 21 Apr 2012 14:28:02 +0200 From: lopez.hernandez.j...@gmail.com To: kew...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA issue playing the prompts thanks - did remove the direct inward dial, checked the complete match. Still no prompt to be heard passed the initial welcome to cisco unified comms ' cheers, Juan Op 21 april 2012 12:00 schreef Ken Wyan kew...@gmail.com het volgende: Remove direct-inward-dial from dial-peer 100 In service parameters MVA destination should be complete match (default) Thanks On Sat, Apr 21, 2012 at 2:27 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: Dear all, redoing some lab testing today, and for some odd reason :) I am not able to have MVA playing the prompts. I call in into the MVA number. I hear the lady in the box telling me I'm welcome to Cisco Unified Comms. Then it goes silent apart from that it is working: while no one asks me - enter the user's PIN code 12345, press 1, then an internal number in the none partition, it will connect to that number. Cheking the debug vxml on the router does not tell much. Then checking the trace files either: the trace steps regarding playing the first prompt - which I believe is the one asking for the PIN: controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 5.au and locale = en_US controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 5.au and locale = en_US controller.IVRGetAudioFile - [CCM_IVR]:: getting the file now so no errors, but nothing played. Can anyone help me with this, I cannot seem to find what is wrong. To make sure, I placed the router in the HQ device pool - so it speaks g711 with the UCM -as I am not sure the UCM will be able to play these prompts in g729 too. On the router it's: dial-peer voice 100 pots service mva incoming called-number 2220 direct-inward-dial ! dial-peer voice 1001 voip destination-pattern 2220 voice-class codec 1 session target ipv4:177.1.10.10 dtmf-relay h245-alphanumeric no vad with the voice-class codec speaking g711u/g729r8 PS/ reboot of servers did not help. any help is much appreciated ! Juan ___ 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] Fwd: UCCX Script
Hi Chris, Make sure that there's a transcoder at HQ if you're using G729r8 between HQ and BR1. UCCX only uses G711. ThanksRamy Date: Sun, 8 Apr 2012 20:07:30 +1000 From: devsin2...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Fwd: UCCX Script The attached is screen capture of the icd script and script. Task is to redirect the call to BR1PH2 1002, if there are no agents logged in. I reset the application and script few times. but call still doesn't get redirected to 1002. Can someone have a look at the script and tell where is the mistake. Best RegardsChris ___ 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] UCCX and Media Resources
Hi Chris, 1. Make sure calls from BR1/BR2 to HQ are working properly. If so, then RSVP is working fine.2. Reset the xcoder in the CUCM and do no sccp/sccp in HQ GW. Thanks,Ramy Date: Tue, 17 Apr 2012 18:17:24 +1000 From: devsin2...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX and Media Resources My UCCX is in HQ device pool. The DP has MRGL allocated to with registered transcoder resources. However, when I try to dial from BR1/BR2. The call fails to connect. The SDI traces on the call manager show following messages: 04/17/2012 15:28:30.231 CCM|MediaManager(9)::disconnOnResourceAllocationFailure, ERROR disconnOnResourceAllocationFailure - fails to allocate MTP/XCoder,connCount=2|CLID::StandAloneClusterNID::10.10IP::10.10.100.14DEV::UCCX_5701LVL::ErrorMASK::0800 Xcoder resource is configured as Transcoding Oper State: ACTIVE - Cause Code: NONEActive Call Manager: 10.10.100.12, Port Number: 2000TCP Link Status: CONNECTED, Profile Identifier: 1Reported Max Streams: 6, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 MTP Oper State: ACTIVE - Cause Code: NONEActive Call Manager: 10.10.100.12, Port Number: 2000TCP Link Status: CONNECTED, Profile Identifier: 3Reported Max Streams: 20, Reported Max OOS Streams: 0 Supported Codec: pass-thru, Maximum Packetization Period: N/ASupported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30RSVP : ENABLED MRGL Can someone tell me what am I doing wrong. ThanksChris ___ 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 attachment: image.png___ 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 CLI is Blank
Press Ctrl+C. Regards,Ramy Date: Sun, 15 Apr 2012 12:26:54 +0530 From: kew...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE CLI is Blank When I initially log-in to CUE , it shows blank. When I press ENTER , cursor goes to next line , but no text displayed. Any quicker solution before reloading / resetting CUE ? R3#service-module se R3#service-module service-Engine 0/0 sess R3#service-module service-Engine 0/0 session Trying 10.10.112.2, 2194 ... Open R3#service-module service-Engine 0/0 session Trying 10.10.112.2, 2194 ... % Connection refused by remote host R3#service-module service-Engine 0/0 session clear [confirm] [OK] R3# [Resuming connection 1 to 10.10.112.2 ... ] [Connection to 10.10.112.2 closed by foreign host] R3# R3#service-module service-Engine 0/0 session Trying 10.10.112.2, 2194 ... Open ___ 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] switch-QoS-Quick Question
Hi Steven, It seems to be correct but the policy-map should be applied to the output queue not the input of fa0/1. ThanksRamy Date: Wed, 28 Mar 2012 11:11:24 +1100 From: smoran...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] switch-QoS-Quick Question I’m preparing for the exam and as you are all aware question interpretation is really important. Below is a practice question plus my config on how to approach it. I would appreciate it if anyone could comment on my approach to the question and see if the answer meets the brief. I considered running auto qos on the phone and server ports to mark the traffic at source but this seems excessive for the question. Question On port fa0/1 which is connected to HQ router, guarantee 16k for MGCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. mls qos ! mls qos map cos 0 8 16 24 32 46 48 56 ! mls qos map policed-dscp 24 to 8 ! ip access-list extended 100 permit tcp any any eq 2428 permit udp any any eq 2427 ! class-map class-mgcp match access-group 100 ! policy-map policy-mgcp class class-mgcp set dscp cs3 police 16000 8000 exceed-action policed-dscp-transmit ! interface fa0/1 service input policy-mgcp ___ 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] HQ Phones - DHCP Problem
Actually, I changed the question and I was trying to receive DHCP for HQ/BR1 from CUCM PUB and BR2 from BR2-RTR. I'll try the cdp advertise and update you all. Thanks Ramy CC: ccie_voice@onlinestudylist.com From: whl...@gmail.com Subject: Re: [OSL | CCIE_Voice] HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 16:52:59 -0600 To: ramyoth...@hotmail.com On switch you have no cdp adv, this command prevents the phones from getting vlan set No dhcp pool on router so your br1 is most likely using cached addressing Bill On Feb 26, 2012, at 2:54 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Kindly find attached the full config of HQ-3750 and HQ-RTR for your reference. Thanks Ramy From: ramyoth...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 19:53:34 + Hi All, In the five-lab ebook, lab 2, there's a requirement that HQ/BR1 phones must receive an IP address from the DHCP server configured on HQ-RTR. BR1 phones didreceive IP address successfully but HQ did not. Can anyone help on this? Thanks in advance. Ramy HQ-3750.txtHQ-RTR.txt___ 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] HQ Phones - DHCP Problem
Dear All, I tried now to enable cdp advertise-v2 on HQ-3750 switch and it worked. HQ phones were able to receive IP address from the DHCP server on CUCM. Thanks and appreciated :-) Wish you all the best and good luck! Ramy Date: Tue, 28 Feb 2012 21:43:05 +0100 From: forum.ccie.onlinestudyl...@nocer.net To: ccie_voice@onlinestudylist.com CC: ramyoth...@hotmail.com Subject: Re: [OSL | CCIE_Voice] HQ Phones - DHCP Problem Hi Ramy, cdp is only needed for the VLAN information. If your phone doesn't get an ip address you have a different problem. Enable service dhcp on HQ-RTR and check with debug ip udp if port 68 and 69 are getting unicast-ed to the CUCM. Regards, Steven Am 28.02.2012 21:00, schrieb Ramy Abdelrahim: Actually, I changed the question and I was trying to receive DHCP for HQ/BR1 from CUCM PUB and BR2 from BR2-RTR. I'll try the cdp advertise and update you all. Thanks Ramy CC: ccie_voice@onlinestudylist.com From: whl...@gmail.com Subject: Re: [OSL | CCIE_Voice] HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 16:52:59 -0600 To: ramyoth...@hotmail.com On switch you have no cdp adv, this command prevents the phones from getting vlan set No dhcp pool on router so your br1 is most likely using cached addressing Bill On Feb 26, 2012, at 2:54 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Kindly find attached the full config of HQ-3750 and HQ-RTR for your reference. Thanks Ramy From: ramyoth...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 19:53:34 + Hi All, In the five-lab ebook, lab 2, there's a requirement that HQ/BR1 phones must receive an IP address from the DHCP server configured on HQ-RTR. BR1 phones didreceive IP address successfully but HQ did not. Can anyone help on this? Thanks in advance. Ramy HQ-3750.txt HQ-RTR.txt ___ 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] HQ Phones - DHCP Problem
Hi Bill, Thank you very much for your advice and this valuable information. Appreciated :-) Ramy CC: ccie_voice@onlinestudylist.com From: whl...@gmail.com Subject: Re: [OSL | CCIE_Voice] HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 16:52:59 -0600 To: ramyoth...@hotmail.com On switch you have no cdp adv, this command prevents the phones from getting vlan set No dhcp pool on router so your br1 is most likely using cached addressing Bill On Feb 26, 2012, at 2:54 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Kindly find attached the full config of HQ-3750 and HQ-RTR for your reference. Thanks Ramy From: ramyoth...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 19:53:34 + Hi All, In the five-lab ebook, lab 2, there's a requirement that HQ/BR1 phones must receive an IP address from the DHCP server configured on HQ-RTR. BR1 phones didreceive IP address successfully but HQ did not. Can anyone help on this? Thanks in advance. Ramy HQ-3750.txtHQ-RTR.txt___ 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] HQ Phones - DHCP Problem
Hi All, In the five-lab ebook, lab 2, there's a requirement that HQ/BR1 phones must receive an IP address from the DHCP server configured on HQ-RTR. BR1 phones didreceive IP address successfully but HQ did not. Can anyone help on this? Thanks in advance. Ramy ___ 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] HQ Phones - DHCP Problem
Kindly find attached the full config of HQ-3750 and HQ-RTR for your reference. Thanks Ramy From: ramyoth...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: HQ Phones - DHCP Problem Date: Sun, 26 Feb 2012 19:53:34 + Hi All, In the five-lab ebook, lab 2, there's a requirement that HQ/BR1 phones must receive an IP address from the DHCP server configured on HQ-RTR. BR1 phones didreceive IP address successfully but HQ did not. Can anyone help on this? Thanks in advance. Ramy SiteA-Switch#sh run Building configuration... Current configuration : 2455 bytes ! ! Last configuration change at 15:26:46 UTC Sun Feb 26 2012 ! NVRAM config last updated at 15:26:47 UTC Sun Feb 26 2012 ! version 12.2 no service pad service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname SiteA-Switch ! boot-start-marker boot-end-marker ! ! no aaa new-model switch 1 provision ws-c3750-24p system mtu routing 1500 ip subnet-zero ip routing ! ! ! ! ! ! ! ! ! ! spanning-tree mode pvst spanning-tree extend system-id ! vlan internal allocation policy ascending ! ! ! ! interface FastEthernet1/0/1 description TRUNK to HQ-RTR switchport trunk encapsulation dot1q switchport trunk native vlan 10 switchport mode trunk speed 100 duplex full queue-set 2 ! interface FastEthernet1/0/2 description HQ PHONE 1- 7960 phone switchport access vlan 10 switchport mode access switchport voice vlan 20 spanning-tree portfast ! interface FastEthernet1/0/3 ! interface FastEthernet1/0/4 description SERVER port- do not change switchport access vlan 30 switchport mode access duplex half spanning-tree portfast ! interface FastEthernet1/0/5 ! interface FastEthernet1/0/6 ! interface FastEthernet1/0/7 ! interface FastEthernet1/0/8 ! interface FastEthernet1/0/9 ! interface FastEthernet1/0/10 ! interface FastEthernet1/0/11 ! interface FastEthernet1/0/12 ! interface FastEthernet1/0/13 ! interface FastEthernet1/0/14 ! interface FastEthernet1/0/15 ! interface FastEthernet1/0/16 ! interface FastEthernet1/0/17 ! interface FastEthernet1/0/18 ! interface FastEthernet1/0/19 ! interface FastEthernet1/0/20 ! interface FastEthernet1/0/21 ! interface FastEthernet1/0/22 ! interface FastEthernet1/0/23 description HQ PHONE 2- 7962 phone shutdown ! interface FastEthernet1/0/24 description *** DO NOT CHANGE - THIS IS YOUR L3 CONNECTION TO YOUR VPN!!! *** switchport access vlan 10 switchport mode access speed 100 duplex full no cdp enable ! interface GigabitEthernet1/0/1 ! interface GigabitEthernet1/0/2 ! interface Vlan1 no ip address ! interface Vlan10 ip address 10.10.100.3 255.255.255.0 ! interface Vlan101 no ip address ! ip classless ip route 0.0.0.0 0.0.0.0 10.10.100.1 ip http server no ip http secure-server ! ! no cdp advertise-v2 ! control-plane ! ! line con 0 exec-timeout 0 0 logging synchronous line vty 0 4 exec-timeout 0 0 privilege level 15 logging synchronous no login line vty 5 15 exec-timeout 0 0 privilege level 15 logging synchronous no login ! end SiteA-Switch# SiteA-RTR#sh run Building configuration... Current configuration : 2533 bytes ! ! Last configuration change at 15:21:47 EST Sun Feb 26 2012 ! NVRAM config last updated at 15:21:47 EST Sun Feb 26 2012 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname SiteA-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 clock timezone EST -5 clock summer-time EDT recurring no network-clock-participate wic 0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! ! voice-card 0 no dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 description WAN CONNECTION- DO NOT CHANGE ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/0.101 ! interface FastEthernet0/0.102 ! interface FastEthernet0/0.103 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0
[OSL | CCIE_Voice] DHCP Timeout
Dear All, I faced a scenario on workbook 2 that requests to have HQ and BR1 phones acquire their IP addresses from UCM-PUB. What happened was BR1 phones were able to get IP addresses from the UCM-PUB but HQ phones were not. The Switch and HQ router configuration is as follows. I appreciate if anyone can help on that. NOTE: The UCM-PUB is pingable from the switch and the HQ-RTR. Switch: vlan 10 name DATA!vlan 20 name PHONES!vlan 30 name SERVERS!interface fastethernet 1/0/1 -- To HQ router switchport trunk encapsulation dot1q switchport mode trunk switchport trunk native vlan 10!interface fastethernet 1/0/2 -- HQ Phone 1 switchport access vlan 10 switchport mode access switchport voice vlan 20 spantree portfast!/// HQ-RTR: interface fastethernet 0/0.10 encapsulation dot1q 10 native ip address 10.10.100.1 255.255.255.0!interface fastethernet 0/0.20 encapsulation dot1q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10!interface fastethernet 0/0.30 encapsulation dot1q 30 ip address 10.10.210.1 255.255.255.0!/// Thanks Ramy ___ 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] DHCP Timeout
CUCM is pingable from both the switch and the HQ-RTR. HQ-RTR#ping 10.10.210.10 source 10.10.200.3 Type escape sequence to abort.Sending 5, 100-byte ICMP Echos to 10.10.210.10, timeout is 2 seconds:Packet sent with a source address of 10.10.200.3!Success rate is 100 percent (5/5), round-trip min/avg/max = 1/2/4 ms Date: Fri, 10 Feb 2012 08:17:42 -0600 Subject: Re: [OSL | CCIE_Voice] DHCP Timeout From: whl...@gmail.com To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com OK so BR1 is working but HQ is not. Difference is that BR1 the switch ports have direct access to the routing, they are clearly being routed. My guess is that the voice vlan on HQ is not able to reach the CUCM. See if you can ping the CUCM from the switch. If this does not work, then you will have to find your routing issue. 2012/2/10 Ramy Abdelrahim ramyoth...@hotmail.com Dear All, I faced a scenario on workbook 2 that requests to have HQ and BR1 phones acquire their IP addresses from UCM-PUB. What happened was BR1 phones were able to get IP addresses from the UCM-PUB but HQ phones were not. The Switch and HQ router configuration is as follows. I appreciate if anyone can help on that. NOTE: The UCM-PUB is pingable from the switch and the HQ-RTR. Switch: vlan 10 name DATA!vlan 20 name PHONES!vlan 30 name SERVERS!interface fastethernet 1/0/1 -- To HQ router switchport trunk encapsulation dot1q switchport mode trunk switchport trunk native vlan 10!interface fastethernet 1/0/2 -- HQ Phone 1 switchport access vlan 10 switchport mode access switchport voice vlan 20 spantree portfast !/// HQ-RTR: interface fastethernet 0/0.10 encapsulation dot1q 10 native ip address 10.10.100.1 255.255.255.0!interface fastethernet 0/0.20 encapsulation dot1q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10!interface fastethernet 0/0.30 encapsulation dot1q 30 ip address 10.10.210.1 255.255.255.0 !/// Thanks Ramy ___ 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] DHCP Timeout
CDP is working as shown below and CDP version is 2. HQ-3750#sh cdp neCapability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge S - Switch, H - Host, I - IGMP, r - Repeater, P - Phone Device IDLocal Intrfce HoldtmeCapability Platform Port IDCUC7-Pub Fas 1/0/4 177H none foun eth0SiteA-RTRFas 1/0/1 133 R S I 2811 Fas 0/0UCMPub Fas 1/0/4 168H none foun eth0UCMSub Fas 1/0/4 162H none foun eth0SEP001BD4C6C195 Fas 1/0/2 135H IP Phone Port 1UCCX7-PUB. Fas 1/0/4 144H Win2000 S Eth 2/2CUPS-Pub Fas 1/0/4 161H none foun eth0HQ-3750# HQ-3750#sh cdpGlobal CDP information:Sending CDP packets every 60 secondsSending a holdtime value of 180 secondsSending CDPv2 advertisements is enabled Date: Fri, 10 Feb 2012 16:16:29 + Subject: Re: [OSL | CCIE_Voice] DHCP Timeout From: ke...@kevinspicer.co.uk To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com; whl...@gmail.com Is CDP working on the switch so that the phones learn the voice vlan? Check the CDP version too. On 10 Feb 2012 16:02, Ramy Abdelrahim ramyoth...@hotmail.com wrote: CUCM is pingable from both the switch and the HQ-RTR. HQ-RTR#ping 10.10.210.10 source 10.10.200.3 Type escape sequence to abort.Sending 5, 100-byte ICMP Echos to 10.10.210.10, timeout is 2 seconds: Packet sent with a source address of 10.10.200.3!Success rate is 100 percent (5/5), round-trip min/avg/max = 1/2/4 ms Date: Fri, 10 Feb 2012 08:17:42 -0600 Subject: Re: [OSL | CCIE_Voice] DHCP Timeout From: whl...@gmail.com To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com OK so BR1 is working but HQ is not. Difference is that BR1 the switch ports have direct access to... ___ 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
