Re: [OSL | CCIE_Voice] BLF Speed Dials
Thank you George On Mon, Jan 16, 2012 at 8:09 PM, George Goglidze gogli...@gmail.com wrote: Hi Datucha, Actually for CUPS you need only line association. you do not need to specify owner user id. Misread the question a little bit initially. the owner user id on the phone is for SNR. Cheers, On Mon, Jan 16, 2012 at 4:01 PM, George Goglidze gogli...@gmail.comwrote: correctisimo :-) On Mon, Jan 16, 2012 at 3:25 PM, datucha123 datucha123 datucha...@gmail.com wrote: When configuring the simple BLF Speed Dial, we need to configure the Subscribe CSS for watching Device. So that it could the the Watched DN. But, the Owner User ID, Line Association with End User and other kinds of associations are not required for this BLF, right? Even for Call List Presence. The Owner User ID and Line Association with End User, along with License Capability Assignements are required only for CUPS Presence method. Is it correct? ___ 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 http://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] Cant connect to Sub
Also check the DB Replication Status On Tue, Jan 17, 2012 at 6:17 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Errol , This issue usually mean you have incorrect Host/ processNode files in your PUB or the Sub and so the communication is broken between the PUB and the SUB and you have to find what table we have corrupted and fix it , or you have a big timing difference reported by your NTP , can you check what service you have running on both pub and sub specially the A Cisco DB and let me know what you will get . Ash On Mon, Jan 16, 2012 at 5:15 PM, Errol Abrahams eabraham2...@gmail.com wrote: Hi All, I had a problem with my VMWARE Server and I had to rebuilt the system from scratch. I have reloaded PUB,SUB,CUPS,CUC and CUCCX and all virtual addresses are pingable. When I activate the services for the PUB from the Cisco Unified Serviceability screen, it worked. But, when I try to access the SUB from same screen then it displays'Connection to the Server cannot be established(unable to access Remote Node). Has anybody had a problem like this and how can I fix this problem. Your help is appreciated..thnx. Chhers EA ___ 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 http://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 http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voicemail access during AAR
Hi experts I am working on Vol 2. Lab 7 and try to fully understand the topic of voice mail access during AAR, but I am struggling to have a clear picture. Here is the setup HQ GW - MGCP BR GW - H323 HQ Phones, and BR Phones are both SCCP. The line on both phone been assigned to the same AAR group. Only HQ phone assigned with AAR group and CSS AAR on the device level. voicemail for BR is on CUE which is integrated with CUCM using CTI route point. All integration is configured and tested ok AAR between the phones are working, the HQ Phone will reroute out to PSTN to reach BR Phones when there is not enough bandwidth. Also, HQ Phone can directly dial in to the voicemail pilot (reroute out to PSTN) and reach the CUE log in. BR phone can press the message button and reach the sign-in prompt for CUE (only asking for PIN) However, when HQ phone calls BR phone and BR Phone doesnt pick up, just as it should transfer to voicemail, I get fast busy tone on HQ Phone. I am trying to understand, at this instance, I am still using the AAR CSS on HQ Phone to reach the voice mail pilot right? I imagine if HQ Phone can successfully call to voicemail directly during AAR, it shouldnt be different when it is transferred by BR Phone? Also, I am trying to understand when do we need to assign AAR group and AAR CSS to the gateway? and why? Please help Thanks in advance Vega ___ 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] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
Hello, I have a problem regarding srst. The 2811 router than now is a srst was a acting as a cme in past in a demo lab and the 7960 phone was registered to it with extension . Now, the 7960 phone is registered in UCM with extension 5001 and the 2811 router is configured in telephony service srst mode. The problem is that although the old configuration of the router was erased, when it goes in srst fallback mode, the 7960 gets extension instead of 5001 and the command show telephony-service ephone shows the specified phone with message:This is an srst fallback phone. Thank you for your answer ___ 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] Presence on CUPC
Hello, Based on my testings, CUPC support Presence Status change in its Contacts only when they are imported from AD. Manually created contacts does not support Presence Status Change. Is it correct? ___ 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] Voicemail access during AAR
- Can AAR CSS assigned to HQ DN/Device reach Voicemail Pilot over PSTN? GW AAR CSS is used when there is an incoming call from PSTN to BR IP phone and the phone has call forwarding set to voicemail. Also VM is in other CAC location that has insufficient bandwidth which triggers AAR function. Call is terminated on gw and will do forwarding instead of IP Phone at the branch. Here gw's AAR CSS will route the call toward the Voicemail's pilot. Peter - Original Message - From: Vega Wong vega2...@yahoo.com.au Date: Tuesday, January 17, 2012 2:01 pm Subject: [OSL | CCIE_Voice] Voicemail access during AAR To: ccie_voice@onlinestudylist.com Hi experts I am working on Vol 2. Lab 7 and try to fully understand the topic of voice mail access during AAR, but I am struggling to have a clear picture. Here is the setup HQ GW - MGCP BR GW - H323 HQ Phones, and BR Phones are both SCCP. The line on both phone been assigned to the same AAR group. Only HQ phone assigned with AAR group and CSS AAR on the device level. voicemail for BR is on CUE which is integrated with CUCM using CTI route point. All integration is configured and tested ok AAR between the phones are working, the HQ Phone will reroute out to PSTN to reach BR Phones when there is not enough bandwidth. Also, HQ Phone can directly dial in to the voicemail pilot (reroute out to PSTN) and reach the CUE log in. BR phone can press the message button and reach the sign-in prompt for CUE (only asking for PIN) However, when HQ phone calls BR phone and BR Phone doesnt pick up, just as it should transfer to voicemail, I get fast busy tone on HQ Phone. I am trying to understand, at this instance, I am still using the AAR CSS on HQ Phone to reach the voice mail pilot right? I imagine if HQ Phone can successfully call to voicemail directly during AAR, it shouldnt be different when it is transferred by BR Phone? Also, I am trying to understand when do we need to assign AAR group and AAR CSS to the gateway? and why? Please help Thanks in advance Vega ___ 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] CUE error
Hi While accessing my voicemail via VoiceView, I get the following error when I click on listen Unknown error code {0}. Report this error to your system administrator Has anyone run into this error before? I tried several things but I cant get over this error TR ___ 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] Presence on CUPC
yes it's possible, as long as it's correctly done: http://www.cisco.com/en/US/products/ps6837/products_tech_note09186a00808a2b0d.shtml Unable to View Directory Information Problem With CUPC 8.x, you are unable to view directory information, you are unable to see the status/Presence information for any of the added users, search does not show any results, and you are unable to initiate a chat session with another user. Solution This issue might occur due to an LDAP integration issue. If LDAP is not configured, you cannot search any contact. The reason for the user not able to chat with a manually added user is that when manually adding the user if the proper domain name is not provided, then CUPS will not be able to validate that user with CallManager. So, if you are not using LDAP Integration, then it is suggested to add the user through the CUPS User Option Page to resolve the issue. Perform these steps: 1. Delete the user from CUPC. 2. Add the user from the User Option Page: 1. Enter the CUPS IP address, and click *User Option*. 2. Enter the user ID and credentials of the end user. 3. Now, click on *Settings Contacts Add New* and then type the user ID of the user you want to add. Once these steps are complete, you should be able to see the user ID populated in the CUPC, and you should be able to chat with these users. On Tue, Jan 17, 2012 at 2:52 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Based on my testings, CUPC support Presence Status change in its Contacts only when they are imported from AD. Manually created contacts does not support Presence Status Change. Is it correct? ___ 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] BAT File on Excel 2011 for Mac
Hello Experts, Does anybody ever made the BAT file work just fine on Excel 2011? I am asking this because it is a pain to open my VMWare Fusion, load Windows with Excel 2007 every time I need to do something with the BAT file. Any ideas? *Emanuel Damasceno* CCNP Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Unity Connection Integration
Hi, When adding a new port group to unity connection , it prompts to enter only one CUCM ip address. Should we essentially provide CUCM Publisher here? My Integration is CUCM Sub First CUCM Pub Second as we expect to do in CCIE Exam. When adding a phone system we can provide Subscriber (first) Publisher (second) in order (AXL Server) When adding a new portgroup , it asks for only one ip address. The ip address I provide is displayed under Port Group Configuration (CUCM TFTP) we can later add another CUCM address also. Any guideline? ___ 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] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
Did you enter no create cnf-files create cnf-files on CME Router ? Which srst mode are you running? srst mode auto provision all | dn | none ? Better to post full telephony-service configuration here. (In SRST phone may be downloading previous xml configuration file from the router. Delete it from flash if so) On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan winmasterp...@gmail.comwrote: Hello, I have a problem regarding srst. The 2811 router than now is a srst was a acting as a cme in past in a demo lab and the 7960 phone was registered to it with extension . Now, the 7960 phone is registered in UCM with extension 5001 and the 2811 router is configured in telephony service srst mode. The problem is that although the old configuration of the router was erased, when it goes in srst fallback mode, the 7960 gets extension instead of 5001 and the command show telephony-service ephone shows the specified phone with message:This is an srst fallback phone. Thank you for your answer ___ 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 http://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] Presence on CUPC
No its supporting any kind of users (manual or AD imported). Also, it can be another CUPC or IPPM. Regards, Mohammed Al Baqari Sent from my iPhone On Jan 17, 2012, at 5:52 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Based on my testings, CUPC support Presence Status change in its Contacts only when they are imported from AD. Manually created contacts does not support Presence Status Change. Is it correct? ___ 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] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
Try to do factory reset for the phone and then test again. Regards, Mohammed Al Baqari Sent from my iPhone On Jan 17, 2012, at 5:23 PM, The Masterplan winmasterp...@gmail.com wrote: Hello, I have a problem regarding srst. The 2811 router than now is a srst was a acting as a cme in past in a demo lab and the 7960 phone was registered to it with extension . Now, the 7960 phone is registered in UCM with extension 5001 and the 2811 router is configured in telephony service srst mode. The problem is that although the old configuration of the router was erased, when it goes in srst fallback mode, the 7960 gets extension instead of 5001 and the command show telephony-service ephone shows the specified phone with message:This is an srst fallback phone. Thank you for your answer ___ 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] Unity Connection Integration
I guess it would determine what your requirements are. If, for example, the global requirements were that your subscriber were to be the primary server for all call processing, you might want to take that into account when choosing whether to only use one server or two and which should be the primary and secondary. Earl Hough CCIE #16508 (RS/Security/Voice) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ken Wyan Sent: Tuesday, January 17, 2012 11:42 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity Connection Integration Hi, When adding a new port group to unity connection , it prompts to enter only one CUCM ip address. Should we essentially provide CUCM Publisher here? My Integration is CUCM Sub First CUCM Pub Second as we expect to do in CCIE Exam. When adding a phone system we can provide Subscriber (first) Publisher (second) in order (AXL Server) When adding a new portgroup , it asks for only one ip address. The ip address I provide is displayed under Port Group Configuration (CUCM TFTP) we can later add another CUCM address also. Any guideline? _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ 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] Who is passing and how?
I do not want to mis-inform at anybody. Since I have been through the pain and sufferings I want other to have happiness. Let me get to the point, the LAB is very passable if you can configure with very high speed and know land mines. Again, I could be wrong but this is how I felt. Doing fast outweight knowing more. Knowing all technologies in and out is good but you can still fail if you don't have speed. So, please practice your speed. It is very very important. About land mines, I am not sure how to prepare. There are some who posted already on this and other forums on trick questions. I agreed that most of the questions are very reasonable but there are some which isn't best practice nor a real world problems. So sorry that I can't give you advice on how to prepare but the good thing is there are only a couple crazy questions par LAB. Good luck guys! On Sat, Jan 14, 2012 at 10:31 PM, Edgar Feliz ejzi...@gmail.com wrote: Well I know the proctor I had at RTP would not pass it, I had a issue with something and he had no clue, proctor's use script they are like the telemarketers that call you to bother you. My thought on your main question is this and I do not know how many times you have taken the lab, I have taken it once, be prepared. I know looking back I was not prepared the first time not like I thought I was, I will be going again soon and I feel better prepared for this attempt will I pass hard to say but I will try, there seems to be a lot of variables that no one can explain. The grading of the questions I have heard different things from people, proctors, and instructor saying things are like this but they are saying different things, do you get partial credit or not Someone I know at Cisco that has taken the lab 3 times, last time finished in 6 hours double checked everything said everything was 100% verified and failed worse then her first attempthow can that be? Concentrate on what is giving you trouble why work on the stuff you know as much as the stuff you don't, work harder on what you don't have a good grasp on. I have a lot of friends that have taken the lab 3, 4 and more times and some have stopped because it's expensive and time consuming. I always tell people the reward for passing is a lifetime of work. E On Fri, Jan 13, 2012 at 9:44 PM, cciev wannabe cciev.wann...@gmail.comwrote: Hi All, My questions to you especially to those who has taken the lab: how to pass the lab? I took a few time already and failed each time. It looks very passable after reading all the questions. But for me, time isn't enough to troubleshoot and do the configurations. Not to insult anybody but I sincerely wonder can even proctor pass these lab if they haven't seen it? Any thoughts? Thanks, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unity Connection Integration
That parameter actually does not effect anything. On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl earl.ho...@pcmallservices.comwrote: I guess it would determine what your requirements are. If, for example, the global requirements were that your subscriber were to be the primary server for all call processing, you might want to take that into account when choosing whether to only use one server or two and which should be the primary and secondary. ** ** Earl Hough CCIE #16508 (RS/Security/Voice) ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan *Sent:* Tuesday, January 17, 2012 11:42 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Unity Connection Integration ** ** Hi, When adding a new port group to unity connection , it prompts to enter only one CUCM ip address. Should we essentially provide CUCM Publisher here? My Integration is CUCM Sub First CUCM Pub Second as we expect to do in CCIE Exam. When adding a phone system we can provide Subscriber (first) Publisher (second) in order (AXL Server) When adding a new portgroup , it asks for only one ip address. The ip address I provide is displayed under Port Group Configuration (CUCM TFTP) we can later add another CUCM address also. Any guideline? _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ 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 http://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] Gatekeeper VIAZONE - Could not find an IPIPGW
Hi there, i got some problems with my viazone (CUBE) at HQ-RTR. I already checked the Tech prefix match and it seems to succeed. But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem. I also checked the dial-peers on HQ-RTR. Any help appreciated. Regards, Steven ! *** Begin tech details: HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to BR2 (3006) Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: arqp=0x48F0C08C,crv=0x7, answerCall=0 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name servers Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Tech-prefix match failed. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Matched zone prefix 3 and remainder 006 Jan 17 19:56:37.786: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4793079C Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x47930A08 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=4 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone and z_outvianamep=CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (CUBE) specified. Pick an IP-IP gateway in that viazone. Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, tpp: 0x0, current_endpt: 0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Searched through the entire gateway list. loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Could not find an IPIPGW. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway selection failed for zone CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x805) HQ-RTR#show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone CUBE master gateway list: 10.10.110.1:1720 HQ-RTR Zone UCM master gateway list: 10.10.210.10:44248 gk-trunk_1 10.10.210.11:36641 gk-trunk_2 Prefix: 3#* Zone UCME master gateway list: 10.10.110.3:1720 BR2-RTR HQ-RTR#show gatekeeper zone prefix ZONE PREFIX TABLE = GK-NAME E164-PREFIX --- --- UCME 3... UCM 5... HQ-RTR#show running-config interface loopback 0 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip tech-prefix 1# HQ-RTR#show running-config | section gatekeeper gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia CUBE zone local CUBE ipexpert.com zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unity Connection Integration
You can set it to any server you like initially, but when you go to Server configuration in Port Group page, there you have to set them as necessary and required. On Wed, Jan 18, 2012 at 12:18 AM, datucha123 datucha123 datucha...@gmail.com wrote: That parameter actually does not effect anything. On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl earl.ho...@pcmallservices.com wrote: I guess it would determine what your requirements are. If, for example, the global requirements were that your subscriber were to be the primary server for all call processing, you might want to take that into account when choosing whether to only use one server or two and which should be the primary and secondary. ** ** Earl Hough CCIE #16508 (RS/Security/Voice) ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan *Sent:* Tuesday, January 17, 2012 11:42 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Unity Connection Integration ** ** Hi, When adding a new port group to unity connection , it prompts to enter only one CUCM ip address. Should we essentially provide CUCM Publisher here? My Integration is CUCM Sub First CUCM Pub Second as we expect to do in CCIE Exam. When adding a phone system we can provide Subscriber (first) Publisher (second) in order (AXL Server) When adding a new portgroup , it asks for only one ip address. The ip address I provide is displayed under Port Group Configuration (CUCM TFTP) we can later add another CUCM address also. Any guideline? _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ 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 http://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 error
First of all ensure that the Authentication URL is set correctly. Also you can try to reset the CUE module - just in case. On Tue, Jan 17, 2012 at 6:30 PM, study buddy studybudd...@gmail.com wrote: Hi While accessing my voicemail via VoiceView, I get the following error when I click on listen Unknown error code {0}. Report this error to your system administrator Has anyone run into this error before? I tried several things but I cant get over this error TR ___ 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 http://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] Study Partner wanted
Hi, Looking for a study partner... am working towards taking my first attempt of the Lab. Am in the Geneva/French alps area Thanks D ___ 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] redundancy for SIP dialpeers
When configuring 2 SIP dialpeers for redundancy, together with: sip-ua retry invite 2 This should generate in total 3 INVITES sent to the primary UCM via the first dialpeer, before going over to the second sip dialpeer, right? Doing debug ccsip messages only shows 1 invite sent to the primary, and then 1 invite to the secondary. am I missing something? thanks, 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
[OSL | CCIE_Voice] GK routed calls into UCM
When 2 sites BR1 and BR2 call into the UCM phones via gatekeeper, these 2 sites send H225 call setup to the UCM. In this case there is no way to choose a codec based on calling (= GK trunk on the UCM) and called endpoint for *both situations*? example: GK-trunk in DP BR1calls from BR1 GW will have g711 for calls to BR1 phones. However, calls to BR2 phones that enter the BR2 gw will then use the g729 codec, instead of the intra-site g711 codec Setting the GK trunk in a device pool that speaks g711 with all is not good either, as this would mean a xfer from BR1 to BR2 would create a g711 call between BR1 GW and BR2 phone... Am I missing a valid solution ? Another question: For GK call routing, it is not necessary for the UCM to know the h323 source address of both remote branches, only the gatekeeper needs to be defined. How does UCM know it can accept H225 messages sourced by both gateways - as they are not defined? Is there a 'GK' flag set in the h225 setup that triggers the UCM to consult the GKand therefore accept the call from the undefined h323 gateways? ___ 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] Gatekeeper VIAZONE - Could not find an IPIPGW
Hi Steve, Do you have This in your config? Voice service voip Allow h323 to h323 If not, add it and do no gateway/gateway Your cube should appear as H323 type in show gatek end. Sent from my mobile device, sorry for typos. --- Regards Boris On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote: Hi there, i got some problems with my viazone (CUBE) at HQ-RTR. I already checked the Tech prefix match and it seems to succeed. But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem. I also checked the dial-peers on HQ-RTR. Any help appreciated. Regards, Steven ! *** Begin tech details: HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to BR2 (3006) Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: arqp=0x48F0C08C,crv=0x7, answerCall=0 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name servers Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Tech-prefix match failed. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Matched zone prefix 3 and remainder 006 Jan 17 19:56:37.786: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4793079C Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x47930A08 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=4 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone and z_outvianamep=CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (CUBE) specified. Pick an IP-IP gateway in that viazone. Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, tpp: 0x0, current_endpt: 0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Searched through the entire gateway list. loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Could not find an IPIPGW. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway selection failed for zone CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x805) HQ-RTR#show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone CUBE master gateway list: 10.10.110.1:1720 HQ-RTR Zone UCM master gateway list: 10.10.210.10:44248 gk-trunk_1 10.10.210.11:36641 gk-trunk_2 Prefix: 3#* Zone UCME master gateway list: 10.10.110.3:1720 BR2-RTR HQ-RTR#show gatekeeper zone prefix ZONE PREFIX TABLE = GK-NAME E164-PREFIX --- --- UCME 3... UCM 5... HQ-RTR#show running-config interface loopback 0 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip tech-prefix 1# HQ-RTR#show running-config | section gatekeeper gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia CUBE zone local CUBE ipexpert.com zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown ___ 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
Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW
and do you have at least one h323 dial-peer on the HQ router Leslie Meade - Original Message - From: Boris boris.k...@gmail.com To: Steven forum.ccie.onlinestudyl...@nocer.net Cc: ccie voice ccie_voice@onlinestudylist.com Sent: Tuesday, January 17, 2012 1:19:21 PM Subject: Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW Hi Steve, Do you have This in your config? Voice service voip Allow h323 to h323 If not, add it and do no gateway/gateway Your cube should appear as H323 type in show gatek end. Sent from my mobile device, sorry for typos. --- Regards Boris On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote: Hi there, i got some problems with my viazone (CUBE) at HQ-RTR. I already checked the Tech prefix match and it seems to succeed. But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem. I also checked the dial-peers on HQ-RTR. Any help appreciated. Regards, Steven ! *** Begin tech details: HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to BR2 (3006) Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: arqp=0x48F0C08C,crv=0x7, answerCall=0 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name servers Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Tech-prefix match failed. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Matched zone prefix 3 and remainder 006 Jan 17 19:56:37.786: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4793079C Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x47930A08 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=4 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone and z_outvianamep=CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (CUBE) specified. Pick an IP-IP gateway in that viazone. Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, tpp: 0x0, current_endpt: 0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Searched through the entire gateway list. loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Could not find an IPIPGW. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway selection failed for zone CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x805) HQ-RTR#show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone CUBE master gateway list: 10.10.110.1:1720 HQ-RTR Zone UCM master gateway list: 10.10.210.10:44248 gk-trunk_1 10.10.210.11:36641 gk-trunk_2 Prefix: 3#* Zone UCME master gateway list: 10.10.110.3:1720 BR2-RTR HQ-RTR#show gatekeeper zone prefix ZONE PREFIX TABLE = GK-NAME E164-PREFIX --- --- UCME 3... UCM 5... HQ-RTR#show running-config interface loopback 0 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip tech-prefix 1# HQ-RTR#show running-config | section gatekeeper gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia CUBE zone local CUBE ipexpert.com zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown ___ For more information regarding industry leading CCIE
Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW
If the Voice service Voip commands are all configured...A restart has always helped me ... From: Boris boris.k...@gmail.com To: Steven forum.ccie.onlinestudyl...@nocer.net Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, January 18, 2012 10:19 AM Subject: Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW Hi Steve, Do you have This in your config? Voice service voip Allow h323 to h323 If not, add it and do no gateway/gateway Your cube should appear as H323 type in show gatek end. Sent from my mobile device, sorry for typos. --- Regards Boris On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote: Hi there, i got some problems with my viazone (CUBE) at HQ-RTR. I already checked the Tech prefix match and it seems to succeed. But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem. I also checked the dial-peers on HQ-RTR. Any help appreciated. Regards, Steven ! *** Begin tech details: HQ-RTR#debug gatekeeper main 10 ! tried to call from HQ (5002) to BR2 (3006) Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: arqp=0x48F0C08C,crv=0x7, answerCall=0 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name servers Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Tech-prefix match failed. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Matched zone prefix 3 and remainder 006 Jan 17 19:56:37.786: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4793079C Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x47930A08 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=4 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone and z_outvianamep=CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (CUBE) specified. Pick an IP-IP gateway in that viazone. Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, tpp: 0x0, current_endpt: 0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Searched through the entire gateway list. loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Could not find an IPIPGW. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway selection failed for zone CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x805) HQ-RTR#show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone CUBE master gateway list: 10.10.110.1:1720 HQ-RTR Zone UCM master gateway list: 10.10.210.10:44248 gk-trunk_1 10.10.210.11:36641 gk-trunk_2 Prefix: 3#* Zone UCME master gateway list: 10.10.110.3:1720 BR2-RTR HQ-RTR#show gatekeeper zone prefix ZONE PREFIX TABLE = GK-NAME E164-PREFIX --- --- UCME 3... UCM 5... HQ-RTR#show running-config interface loopback 0 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip tech-prefix 1# HQ-RTR#show running-config | section gatekeeper gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia CUBE zone local CUBE ipexpert.com zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown ___ For
Re: [OSL | CCIE_Voice] QoS question on Workbook2 Lab 10
...I just did a check: in Workbook 2 Lab 6, Tasks 7.1, 7.2 we are trusting the phones+HWIC-4ESW on both BR1 BR2 , the class-map used is class-map match-all wan-rtp match dscp ef etc. etc ...so as I thought, the DSG is not consistent here... On Wed, Jan 18, 2012 at 12:52 PM, Anthony Alba ascanio.al...@gmail.comwrote: Hello, This is what I thought the DSG was pointing too: the HWIC-4ESW is a cheapo low-end device and we're not sure what it does with the markings from the phone so let's re-classify and re-mark at BR1's WAN egress interface to be safe (i.e., don't depend on what phone + HWIC-4ESW passes to us) BTW, I have no knowledge that the HWIC-4ESW spoils markings so this is more a case of being paranoid. Now if you had your phones attached via another 3750 to BR1 then by all means use trust. (I'm not sure the DSG is entirely consistent about this: I'm sure there are other solutions where the phone+HWIC-4ESW is trusted.) On Tue, Jan 17, 2012 at 10:18 AM, John McGaughey (jomcgaug) jomcg...@cisco.com wrote: Hello, ** ** In Workbook 2, Lab 10, question 5.2 it asks you to setup MLP LFI between HQ and BR1. In the solution guide it has you use auto qos trust on the HQ side but does not use trust on the BR1 side. The DSG guide says the reason for not using the trust key word is because of the following: ** ** *Note that we have not done any prior QOS classification/marking on the ESW module therefore we will use class-based marking (no use of the trust keyword when running auto qos).* * * But the phones use the following markings by default. ** ** signaling (SCCP or SIP) - CoS 3 / cs3 media (RTP) - CoS 5 / DSCP 46 (EF) ** ** Why couldn’t we just use the trust keyword on BR1 as well since the phone is already marking the packets correctly? ** ** John ___ 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] QoS question on Workbook2 Lab 10
If the question says to trust packets, should you use the trust keyword or is that there to throw you off. Sent from my iPhone On Jan 16, 2012, at 6:54 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello John , You thinking is correct however as the Packet will traverse over the wan , it will be always subject to get modified by a lot of SW to change specially the DSCP Value and so the QOS SRND recommend to Not trust the packets coming from the wan and so we dont trust in the Branches routers . The right answer for any QOS Question is what is in the SRND and i believe thats why you will find it in the real lap in the desktop Ash On Mon, Jan 16, 2012 at 8:18 PM, John McGaughey (jomcgaug) jomcg...@cisco.com wrote: Hello, In Workbook 2, Lab 10, question 5.2 it asks you to setup MLP LFI between HQ and BR1. In the solution guide it has you use auto qos trust on the HQ side but does not use trust on the BR1 side. The DSG guide says the reason for not using the trust key word is because of the following: Note that we have not done any prior QOS classification/marking on the ESW module therefore we will use class-based marking (no use of the trust keyword when running auto qos). But the phones use the following markings by default. signaling (SCCP or SIP) - CoS 3 / cs3 media (RTP) - CoS 5 / DSCP 46 (EF) Why couldn’t we just use the trust keyword on BR1 as well since the phone is already marking the packets correctly? John ___ 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