Re: [OSL | CCIE_Voice] MVA
Thx Dan - will check it out once back in my lab - but from memory, indeed, I think that the EPNM was left empty @RDP Thx a lot for the tip! Juan On 21 May 2012, at 13:21, Dan Quinlan (daquinla) daqui...@cisco.com wrote: Check the external phone number mask on the line on the RDP (not the phone itself.). I think you'll find that it's blank. When you alter a DN for a shared line, you need to propagate the changes. DQ d...@cisco.com Sent from my iPhone On May 21, 2012, at 2:08 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: Hi Steven, not sure this was a reply to my question. If so, i was talking about the ANI, not the CNAM :) As for your answer, are you able to show the CNAM when calling à PSTN phone? Can you confirm, as to my knowledge this is not supported - i can only show if for internal calls once answered. Cheers, Juan On 20 May 2012, at 19:23, steven moran smoran...@gmail.com wrote: is the call name missing in the rining or answered state? answer the call on thePSTN side and see if the name is there, if not make sure you have gone to the primary extension number and propagated the call line values to the RDP extension, the user is assocaited with both lines and that the owner ID for the primary phones is the user. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MVA
Hi all, when running MVA, I can call a PSTN number. But the Calling number is not sent in the ISDN setup message.. Anybody an idea where to look ? When I dial directly from the deskphone associated with the remote destination profile (whose remote destination is matched for MVA), the call is sent with the calling number I am sure the ANI needs to be sent for MVA calls - I can't seem to find the root cause. 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
Re: [OSL | CCIE_Voice] Lab 7 Lan QOS
not sure about that Ken. If we use auto qos on a switch, it applies a service policy in combination with mls qos statements on the same port. These mls qos statements I believe are used for anything not mathching the service policy cheers, Juan 2012/5/14 Ken Wyan kew...@gmail.com If we specify a service-policy ,then mls qos trust commands won't have any effect. Then you have to consider rtp traffic in ACL. On Sun, May 13, 2012 at 1:32 PM, san r luv...@gmail.com wrote: Since they specifically asked for 'CUPC Signalling ' I don't think you need to mark the RTP traffic On May 13, 2012 11:53 AM, Nazeer rahiman nazs...@yahoo.com wrote: For LAN QoS I got below question All servers are connected (running on vmware) to SW int G 1/0/4. CUPC is running in UCCX and test pc. They asked to configure one in softphone mode other one is desktop mode. also configre voice mail on both clients QoS question was - In Gig 1/0/4 , make sure all incoming CUPC signaling traffic to mark CS3 and gurantee 32k BW. anythung exess should be mark down to DSCP 8 and retransmit. My ans was - mls qos mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos map policed-dscp 24 26 to 8 ip access-list extended voice-rtp permit udp any any range 16384 32767 ip access-list extended cupc-sig permit tcp any any eq 5060 permit tcp any any eq 5060 permit tcp any eq 5060 any permit udp any any eq 5060 permit udp any eq 5060 any permit tcp any any eq 143 permit tcp any eq 143 any permit tcp any any eq 80 permit tcp any eq 80 any permit tcp any any eq 443 permit tcp any eq 443 any permit tcp any any eq 993 permit tcp any eq 993 any permit tcp any any eq 7993 permit tcp any eq 7993 any permit tcp any any eq 389 permit tcp any eq 389 any permit tcp any any eq 2748 permit tcp any eq 2748 any config)#class-map voice-rtp (config-cmap)#match access-group name voice-rtp (config)#class-map match any cupc-sig (config-cmap)#match access-group name cupc-sig (config-cmap)#policy-map cupc (config-pmap)#class voice-rtp (config-pmap-c)#set dscp ef (config-pmap)#class cupc-sig (config-pmap-c)#police 32000 8000 exceed-action policed-dscp-transmit (config-pmap-c)#set dscp cs3 (config)#interface GigabitEthernet1/0/4 config-if)#service-policy input cupc Phone ports mls qos trust cos mls qos trust device cisco phone Server ports mls qos trust dscp I got 0 marks for this question - any body can clarify where it's wrong ? ___ 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 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] blind transfer at ucm
yes, but that is in essence still a consultative transfer... thanks for the replies! Juan 2012/5/2 Dan Quinlan daqui...@cisco.com Correct, it’s the service parameter. When set to true, you can hit transfer, dial the number, and hang up. On 5/2/12 7:45 AM, khaled Saholy khaled_sah...@hotmail.com wrote: I think it's the setting in the service parameter configuration Transfer on-hook enabled , set it to True and see the difference. -- From: nsam...@staff.iinet.net.au To: lopez.hernandez.j...@gmail.com; ccie_voice@onlinestudylist.com Date: Wed, 2 May 2012 09:23:49 + Subject: Re: [OSL | CCIE_Voice] blind transfer at ucm Hit the transfer button directly after dialling the number? There is a direct transfer softkey but that’s used to connect two parties currently both on hold (allowing you to drop out without conferencing) *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Juan Lopez *Sent:* Wednesday, 2 May 2012 5:20 PM *To:* CCIE Study *Subject:* [OSL | CCIE_Voice] blind transfer at ucm hi, anyone an idea how to make a blind transfer in UCM instead of consultative? cheers 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 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/ -- *Dan Quinlan Collaboration Engineering PA Territory * *Cisco Systems, Inc. *323 North Shore Drive Suite 300 Pittsburgh, PA 15212 United States Cisco.com http://www.cisco.com Phone: *412.237.6268 *d...@cisco.com Think before you print. This e-mail may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply e-mail and delete all copies of this message. ___ 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] Built-in-Bridge
And remember: ucm cfb does not allow for LBR participants. Cheers, juan On 29 Apr 2012, at 16:36, Mohamed Hassan mrmha...@gmail.com wrote: I agree but what is the problem to use also the built in bridge as extra resources. On Sun, Apr 29, 2012 at 12:49 PM, Ken Wyan kew...@gmail.com wrote: If we disable built-in-bridge of a phone , it uses conference resources available through MRGL for ad-hoc conferences Barge/cBarge . These external Conference resources may be hardware CFB or CUCM ipvoicemediastreamingapp software CFB resources. Is this 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 -- Engineer / Mohamed Rabea Unified communication engineer ___ 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 Synchronization Route Point
did you change any parameters, like eg phone number mask ,AAR dst ... ? cheers, Juan 2012/4/28 Ken Wyan kew...@gmail.com After configuring UCCX server ( applications with trigger , scripts , .) finally if I run Unified CM Data Synchronization check ; UCCX shows that Route Points (created by UCCX itself corresponding to telephony triggers in CUCM) have a data inconsistency between CUCM UCCX. Even we don't check it , there's no problem. But why does it show like this ? ___ 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] RSVP and CFB
hi Ken, sorry - indeed, quite confusing. Must be momentary lapse of reason down here... Trying to figure out/re-engineering what I was trying to do... please disregard for now ;) - I'm going for a break :) ps: correct to say the leaving the location setting on the device level = hub_none, will make the UCM to go look at the next level, ie: the location setting on the device pool. This is what I see for phones, gateways, vm ports, cti ports ... cheers, Juan Op 27 april 2012 07:57 schreef Ken Wyan kew...@gmail.com het volgende: Your question isn't clear. For RSVP , we need 2 mtp s (rsvp enabled) at each. They should be in MRGL of each end device respectively. Are you going to use CFB as rsvp agent as well? On Fri, Apr 27, 2012 at 12:10 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: all, just found out and wonder if someone can confirm: unless you place a CFB in a location (instead of being placed in a location by means of the CFB's device pool location setting) - RSVP will not be triggered ?? So if I leave the location at the CFB set to hub_none (where it then should check the device pool's location setting - correct me if I'm wrong), and with the location on the device pool set to HQ, no RSVP will be triggered. When setting the location on CFB pages to 'HQ' , RSVP will be triggered. Do you also see this in your labs? thanks for the feedback, 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 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] ANI based Call Routing
no problemo ! are you sure about the incoming called . cannot test it right now, but the incoming called . on the POTS will not be affecting the 'answer-address' on the VOIP dialpeer - I think... let me know, will try it tomorrow if I find some time Of course, no other VOIP dialpeer incoming called . to be configured in that case cheers, Juan Op 27 april 2012 16:06 schreef Juan Carlos Anzola juancarlosanz...@gmail.com het volgende: Hi Juan! Great answer, that is the way i found it possible. I already tested in Lab environment and works fine. But i have one concern, to successfully implement this solution there can't be ANY Dial-Peer configured with Incoming-Called number . I believe i can be more specific for my Incoming PSTN Dial-Peer and it should work fine! I will test this and send the final config. Regards. On Fri, Apr 27, 2012 at 12:49 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: Juan, I would - if allowed - catch the 2XXX numbers on IOS dialpeer 1 (answer-address) and 3XXX on dialpeer 2. On both of these, I would manipulate/tag the called number. For example dialpeer 1 adds 123 to the DNIS, dialpeer 2 adds 321 to the dialpeer. This then can match on 2 different outbound dialpeers, pointing to the different ITSPs where I would do the corresponding DNIS manipulation cheers, Juan Op 26 april 2012 21:20 schreef Juan Carlos Anzola juancarlosanz...@gmail.com het volgende: Hi Guys, Consider the following scenario: I have a CUCM Cluster with the Following Extensions: 2XXX: Sales 3XXX: Engineering I have a single H.323 or SIP PSTN GW. I have 2 different ITSP: 10.2.2.2 and 10.3.3.3 Right now, All calls are routing properly throug 10.2.2.2 I want calls from 2XXX to be routed out 10.2.2.2 and calls from 3XXX to be routed out 10.3.3.3 The requirement is to do this without modifying anything in CUCM. Thanks in advance, -- Juan Carlos Anzola ___ 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/ -- Juan Carlos Anzola ___ 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 Scripts End Step
Hi Ken, Gurpreet/Peter are correct - you misunderstand as far as the script behavour goes: the script goes to the next step if nothing is specified under the 'menu' step's 'substep' (eg: no step defined under the 'timeout', 'unsuccessful'...will take it to the next step defined after the menu step) Therefore it is best to have a 'Goto' defined below the 'connect' step so the script can 'terminate' using the 'terminate' step... hope this somewhat clear - not easy to explain ... ;) cheers, Juan Op 26 april 2012 08:58 schreef Ken Wyan kew...@gmail.com het volgende: Hi Gurpeet, Thanks for your recommendation. Take an example , we use a menu step to redirect a call based on a user input. this shows many output flows such as timeout , busy , invalid , unsuccessful. As per your logic , does script hangs in that place if such an (undesired) event occur. I thought script moves forward through next listed steps. Do we need only a goto end step , only under successful call-contact ? Ken On Wed, Apr 25, 2012 at 6:45 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hi Ken, I would go with Peter here. A terminate step is needed to make sure that the session is not stuck and the port is released. We use it as a best practice. You can use a Go To Step with the Connected/Connect step and send it to the Terminate Label you'll create before the End step. Regards Gurpreet On Wed, Apr 25, 2012 at 5:48 AM, Ken Wyan kew...@gmail.com wrote: But UCCX default scripts don't have terminate step ? reason? On Wed, Apr 25, 2012 at 3:03 PM, Farkas Péter wormh...@sch.bme.huwrote: You can insert a Goto step to direct the contact to the final and single End step. I would also put a Terminate step just before the End to free up the IVR port. Peter - Original Message - From: Ken Wyan kew...@gmail.com Date: Wednesday, April 25, 2012 11:26 am Subject: [OSL | CCIE_Voice] UCCX Scripts End Step To: ccie_voice@onlinestudylist.com Sometimes , our scripts send calls to an agent at the middle of the script. In that case should we include an End step right below call-contact step ? Is it recommended / not recommended to have multiple End steps in a single script? 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 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 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] Unity Connection Services
Ken, AXL for import users from UCM, serv. reporter to run reports from serv. pages, dirsync to import users directly from AD (I believe) cheers! Op 24 april 2012 09:38 schreef Ken Wyan kew...@gmail.com het volgende: Unity Connection Serviceability has following services. Cisco AXL Web Service Cisco Serviceability Reporter Cisco Dirsync Even if all above services are deactivated , Unity connection seems works fine (integrated with CUCM / Presence) What above services are for? Ken ___ 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] 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 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] MVA issue playing the prompts
Hi Ramy, I took some day off :) but yes, everything is checked, and removed an reinstated the MVA application in IOS, where it is indeed loaded without issues. The MVA router is indeed h323 - and I placed it into DP=HQ to rule out any codec issues. There is no CAC wahtsoever. When I call into from a non-recognized number, it asks me to enter the remote destination number. From there on, silence again, just like when dialing with a number that is defined as remote destination. I have a complete match also on the RD. And in fact, all functionality is there, but no prompts. gonna leave it for the moment - might be that a return to snapshot of the UCM will whipe away this odd behaviour. cheers, Juan Op 22 april 2012 09:12 schreef Ramy Abdelrahim ramyoth...@hotmail.com het volgende: 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.auand locale = en_US controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 5.auand 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 http://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
Re: [OSL | CCIE_Voice] Router Switch port configurations
oops - I meant : option 1: switchport voice vlan x switchport access vlan y spanning-tree portfast Op 23 april 2012 15:45 schreef Seifeddine Tlili seifeddine.tl...@lvs1.comhet volgende: 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 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] MVA issue playing the prompts
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
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 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] MVA issue playing the prompts
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] Lab Location Decission
maybe a stupid Q - but as I've not been in the lab yet: do we have telnet sessions available - or all console (low speed, specially for debug)? If so, is it on the same the local PC where we can use notepad? I guess the UCCX won't have a putty client on it - so using notepad over there is not ideal... cheers, Juan Op 19 april 2012 08:53 schreef Ken Wyan kew...@gmail.com het volgende: I didn't have problem accessing notepad in local PC. But never try to use notepad via VNC to candidate PC it's slow.. I think , we can use Remote Desktop to connect to UCCX use notepad there , rather than struggling with Test PC (through VNC small display) On Wed, Apr 18, 2012 at 8:51 PM, Mathew Miller miller.mat...@gmail.comwrote: You can use notepad on the test PC but it is not enabled on the PC you are sitting at. So basically you have to use notepad through VNC which sucks. 2012/4/18 Farkas Péter wormh...@sch.bme.hu Notepad is not enabled by default at each location? Peter - Original Message - From: Mathew Miller miller.mat...@gmail.com Date: Wednesday, April 18, 2012 5:04 pm Subject: Re: [OSL | CCIE_Voice] Lab Location Decission To: Juan Carlos Anzola juancarlosanz...@gmail.com, Online Study ccie_voice@onlinestudylist.com I think it depends on how early you like to get up and how close you are to each. RTP Test starts at 7:10. You get a 20 minutes lunch in a conference room and it is catered in and are done by 3:45. SJ Test starts at 8:30. You get a 40-45 minute lunch in a cafeteria with lots of choices. You are done with the test about 5:05. Computers are about the same, but you get access to notepad on your computer in RTP but not in SJ. I felt like the proctor at RTP is more helpful. From: Juan Carlos Anzola juancarlosanz...@gmail.com Date: Wed, 18 Apr 2012 10:44:36 -0400 To: Online Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab Location Decission Hi Guys, I am scheduling my first attempt today. I have heard many myths and rumors about different locations. I am trying to decide between RTP and San Jose. Someone want to share te pros and cons? (In case they really exist) Regards, -- Juan Carlos Anzola ___ 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 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] uccx AAR
somehow I cannot find my note s back on UCCX - dough !!! what I cannot seen to remember: for AAR to work for UCCX, I remember setting the AAR dst mask on the CTI ports RP to the +E164 number on the UCM side. For the CTI RP it is simply the +E164 of the CTI RP. But what to use for the CTI ports? Is it better to reference the CTI RP's +E164 or shall I need to configure each CTI port with it's own +E164? The first option I believe poses a problem when you have eg. more than 1 CTP - pointing the CTI ports to 1 specific CTI RP +E614 number seems weird in that case 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
Re: [OSL | CCIE_Voice] UCCX and Media Resources
Weird... you see rtmt trying to allocate the HQ xcoder? (OOR incrementing or something) To rule out other things: enough BW available on wan link? Rsvp active? Just thinking out loud. Cti ports also in HQ device pool referencing HQ xcoder ? On 17 Apr 2012, at 11:38, Chris devsin2...@gmail.com wrote: Yes. This is all true. I had double checked before my original post. But I did it again after seeing your response. Best Regards Thanks Chris On Tue, Apr 17, 2012 at 7:27 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hi Chris, I would verify the following: 1) Region setting between HQ and Branch sites uses G.729. 2) All your CTI Route Points should show in HQ DP on the CM and the CCX. 3) The Media Resource (Xcoder) should be configured on the HQ router. 4) The codec selected on the CCX is G.711. 5) Your IP phones show in the correct Device Pool. Last but not the least, make sure that the CTI Route point you dial should also be in the HQ DP with an Xcoder in the MRGL of HQ DP. Let me know if all the above stands true. Regards Gurpreet On Tue, Apr 17, 2012 at 4:17 AM, Chris devsin2...@gmail.com wrote: 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: NONE Active Call Manager: 10.10.100.12, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 6, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 MTP Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.10.100.12, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 3 Reported Max Streams: 20, Reported Max OOS Streams: 0 Supported Codec: pass-thru, Maximum Packetization Period: N/A Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 RSVP : ENABLED MRGL image.png Can someone tell me what am I doing wrong. Thanks 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
Re: [OSL | CCIE_Voice] viazone GK - something missing
Hi Kevin, thank you - you were right! So the cube needs to register with tech-prefixe if the called number has a tech-prefix included. So only time viazone cube does not need to register with tech-prefix, is when the called number does not include any tech prefix, which might be the case when using a default-techology tech-prefix. Thanks al lot ! Juan Op 15 april 2012 00:25 schreef Kevin Spicer ke...@kevinspicer.co.uk het volgende: Hi, I can't find what you're referring to in the SRND (that page numbers something else in my copy). What happens if you set the tech prefix of the CUBE to to same as the destination gateway 115# On Sat, Apr 14, 2012 at 10:46 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: thx Kevin, but I don't think the cube needs to register with a tech-prefix in the viazone - cfr p8-40 in the SRND. I just tried registering the cube with a tech-prefix, and defined that tech-prefix as default tech-prefix in the GK - to make sure - but: no, that does not help Op 14 april 2012 23:02 schreef Kevin Spicer ke...@kevinspicer.co.ukhet volgende: Don't you need a technology prefix match, even for a via-zone? Your CUBE isn't registering a tech-prefix and you don't have a default tech-prefix set. On Sat, Apr 14, 2012 at 9:27 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: all, can someone point me to the missing config for my viazone GK? I'm sure it must be some detail I'm overlooking... can't seem to find the reason why the GK is not able to find the IPIPgw that is registered in the viazone IPIPzone thx!! Juan gatekeeper zone local SEA ccie.com 177.1.254.1 outvia IPIPzone zone local AMS ccie.com outvia IPIPzone zone local IPIPzone ccievoice.com no zone subnet AMS default enable zone subnet AMS 177.1.254.3/32 enable no zone subnet AMS 177.1.10.10/32 enable no zone subnet AMS 177.1.10.20/32 enable zone subnet AMS 177.3.11.0/24 enable zone prefix SEA 1* gw-priority 10 SEA-CUCM-GW_2 zone prefix SEA 2* gw-priority 10 SEA-CUCM-GW_2 zone prefix AMS 3* bandwidth interzone default 48 bandwidth session default 16 no shutdown CorpHQ#sh gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 177.1.11.1 1720 177.1.11.1 64510 *IPIPzone* H323-GW H323-ID: *IPIPgw *Voice Capacity Max.= Avail.= Current.= 0 177.1.10.10 1720 177.1.10.10 32952 SEA VOIP-GW H323-ID: SEA-CUCM-GW_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 33095 SEA VOIP-GW H323-ID: SEA-CUCM-GW_2 Voice Capacity Max.= 1 Avail.= 1 Current.= 0 177.1.254.3 1720 177.1.254.3 51010 AMS H323-GW H323-ID: AMS-CME-GW Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 CorpHQ#sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 225#* Zone AMS master gateway list: 177.1.254.3:1720 AMS-CME-GW Prefix: 115#* Zone SEA master gateway list: 177.1.10.20:1720 SEA-CUCM-GW_2 177.1.10.10:1720 SEA-CUCM-GW_1 Zone SEA prefix 2* priority gateway list(s): Priority 10: 177.1.10.20:1720 SEA-CUCM-GW_2 Priority 5: 177.1.10.10:1720 SEA-CUCM-GW_1 Zone SEA prefix 1* priority gateway list(s): Priority 10: 177.1.10.20:1720 SEA-CUCM-GW_2 Priority 5: 177.1.10.10:1720 SEA-CUCM-GW_1 Dialpeers: dial-peer voice 90 voip *(calls to SEA UCM)* destination-pattern 115#[12]...$ session target ras incoming called-number 225#3... dtmf-relay h245-alphanumeric no vad dial-peer voice 91 voip *(calls to AMS CME) * destination-pattern 225#3...$ session target ras incoming called-number 115# dtmf-relay h245-alphanumeric no vad BUT : CorpHQ# ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup ////GK/gk_rassrv_arq: arqp=0x491BC938,crv=0x22, answerCall=0 ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC //FFA1735A80B9/FFA1735A80BB/GK/gk_dns_query: No Name servers //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo: (115#1001) Matched tech-prefix 115# //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo: (115#1001) Matched zone prefix 1 and remainder 001 ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4928DB2C //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: matched zone is AMS, and z_invianamelen=0 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x480A5050 //FFA1735A80B9/FFA1735A80BB/GK
[OSL | CCIE_Voice] viazone GK - something missing
all, can someone point me to the missing config for my viazone GK? I'm sure it must be some detail I'm overlooking... can't seem to find the reason why the GK is not able to find the IPIPgw that is registered in the viazone IPIPzone thx!! Juan gatekeeper zone local SEA ccie.com 177.1.254.1 outvia IPIPzone zone local AMS ccie.com outvia IPIPzone zone local IPIPzone ccievoice.com no zone subnet AMS default enable zone subnet AMS 177.1.254.3/32 enable no zone subnet AMS 177.1.10.10/32 enable no zone subnet AMS 177.1.10.20/32 enable zone subnet AMS 177.3.11.0/24 enable zone prefix SEA 1* gw-priority 10 SEA-CUCM-GW_2 zone prefix SEA 2* gw-priority 10 SEA-CUCM-GW_2 zone prefix AMS 3* bandwidth interzone default 48 bandwidth session default 16 no shutdown CorpHQ#sh gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 177.1.11.1 1720 177.1.11.1 64510 *IPIPzone* H323-GW H323-ID: *IPIPgw *Voice Capacity Max.= Avail.= Current.= 0 177.1.10.10 1720 177.1.10.10 32952 SEA VOIP-GW H323-ID: SEA-CUCM-GW_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 33095 SEA VOIP-GW H323-ID: SEA-CUCM-GW_2 Voice Capacity Max.= 1 Avail.= 1 Current.= 0 177.1.254.3 1720 177.1.254.3 51010 AMS H323-GW H323-ID: AMS-CME-GW Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 CorpHQ#sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 225#* Zone AMS master gateway list: 177.1.254.3:1720 AMS-CME-GW Prefix: 115#* Zone SEA master gateway list: 177.1.10.20:1720 SEA-CUCM-GW_2 177.1.10.10:1720 SEA-CUCM-GW_1 Zone SEA prefix 2* priority gateway list(s): Priority 10: 177.1.10.20:1720 SEA-CUCM-GW_2 Priority 5: 177.1.10.10:1720 SEA-CUCM-GW_1 Zone SEA prefix 1* priority gateway list(s): Priority 10: 177.1.10.20:1720 SEA-CUCM-GW_2 Priority 5: 177.1.10.10:1720 SEA-CUCM-GW_1 Dialpeers: dial-peer voice 90 voip *(calls to SEA UCM)* destination-pattern 115#[12]...$ session target ras incoming called-number 225#3... dtmf-relay h245-alphanumeric no vad dial-peer voice 91 voip *(calls to AMS CME) * destination-pattern 225#3...$ session target ras incoming called-number 115# dtmf-relay h245-alphanumeric no vad BUT : CorpHQ# ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup ////GK/gk_rassrv_arq: arqp=0x491BC938,crv=0x22, answerCall=0 ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC //FFA1735A80B9/FFA1735A80BB/GK/gk_dns_query: No Name servers //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo: (115#1001) Matched tech-prefix 115# //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo: (115#1001) Matched zone prefix 1 and remainder 001 ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4928DB2C //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: matched zone is AMS, and z_invianamelen=0 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x480A5050 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: matched zone is SEA, and z_outvianamelen=8 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone and z_outvianamep=IPIPzone //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: *Received ARQ for a zone (SEA) that has an outviazone (IPIPzone) specified. Pick an IP-IP gateway in that viazone. *////GK/gk_gw_select_ipipgw_random: zonep: 0x4921EC04, tpp: 0x48FB789C, current_endpt: 0 ////GK/gk_gw_select_ipipgw_random: Gateway selection will start at the top of the linked list. use_count=0, current_endpt=0 ////GK/gk_gw_select_ipipgw_random: qelemp=0x0, loop_count=0 ////GK/gk_gw_select_ipipgw_random: Could not find an IPIPGW. CorpHQ# //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo(115#1001): *Viazone gateway selection failed for zone IPIPzone *//FFA1735A80B9/FFA1735A80BB/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x805) CorpHQ# ___ 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] viazone GK - something missing
thx Kevin, but I don't think the cube needs to register with a tech-prefix in the viazone - cfr p8-40 in the SRND. I just tried registering the cube with a tech-prefix, and defined that tech-prefix as default tech-prefix in the GK - to make sure - but: no, that does not help Op 14 april 2012 23:02 schreef Kevin Spicer ke...@kevinspicer.co.uk het volgende: Don't you need a technology prefix match, even for a via-zone? Your CUBE isn't registering a tech-prefix and you don't have a default tech-prefix set. On Sat, Apr 14, 2012 at 9:27 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: all, can someone point me to the missing config for my viazone GK? I'm sure it must be some detail I'm overlooking... can't seem to find the reason why the GK is not able to find the IPIPgw that is registered in the viazone IPIPzone thx!! Juan gatekeeper zone local SEA ccie.com 177.1.254.1 outvia IPIPzone zone local AMS ccie.com outvia IPIPzone zone local IPIPzone ccievoice.com no zone subnet AMS default enable zone subnet AMS 177.1.254.3/32 enable no zone subnet AMS 177.1.10.10/32 enable no zone subnet AMS 177.1.10.20/32 enable zone subnet AMS 177.3.11.0/24 enable zone prefix SEA 1* gw-priority 10 SEA-CUCM-GW_2 zone prefix SEA 2* gw-priority 10 SEA-CUCM-GW_2 zone prefix AMS 3* bandwidth interzone default 48 bandwidth session default 16 no shutdown CorpHQ#sh gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 177.1.11.1 1720 177.1.11.1 64510 *IPIPzone* H323-GW H323-ID: *IPIPgw *Voice Capacity Max.= Avail.= Current.= 0 177.1.10.10 1720 177.1.10.10 32952 SEA VOIP-GW H323-ID: SEA-CUCM-GW_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 33095 SEA VOIP-GW H323-ID: SEA-CUCM-GW_2 Voice Capacity Max.= 1 Avail.= 1 Current.= 0 177.1.254.3 1720 177.1.254.3 51010 AMS H323-GW H323-ID: AMS-CME-GW Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 CorpHQ#sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 225#* Zone AMS master gateway list: 177.1.254.3:1720 AMS-CME-GW Prefix: 115#* Zone SEA master gateway list: 177.1.10.20:1720 SEA-CUCM-GW_2 177.1.10.10:1720 SEA-CUCM-GW_1 Zone SEA prefix 2* priority gateway list(s): Priority 10: 177.1.10.20:1720 SEA-CUCM-GW_2 Priority 5: 177.1.10.10:1720 SEA-CUCM-GW_1 Zone SEA prefix 1* priority gateway list(s): Priority 10: 177.1.10.20:1720 SEA-CUCM-GW_2 Priority 5: 177.1.10.10:1720 SEA-CUCM-GW_1 Dialpeers: dial-peer voice 90 voip *(calls to SEA UCM)* destination-pattern 115#[12]...$ session target ras incoming called-number 225#3... dtmf-relay h245-alphanumeric no vad dial-peer voice 91 voip *(calls to AMS CME) * destination-pattern 225#3...$ session target ras incoming called-number 115# dtmf-relay h245-alphanumeric no vad BUT : CorpHQ# ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup ////GK/gk_rassrv_arq: arqp=0x491BC938,crv=0x22, answerCall=0 ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC //FFA1735A80B9/FFA1735A80BB/GK/gk_dns_query: No Name servers //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo: (115#1001) Matched tech-prefix 115# //FFA1735A80B9/FFA1735A80BB/GK/rassrv_get_addrinfo: (115#1001) Matched zone prefix 1 and remainder 001 ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4928DB2C //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: matched zone is AMS, and z_invianamelen=0 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x480A5050 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: matched zone is SEA, and z_outvianamelen=8 //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone and z_outvianamep=IPIPzone //FFA1735A80B9/FFA1735A80BB/GK/rassrv_arq_select_viazone: *Received ARQ for a zone (SEA) that has an outviazone (IPIPzone) specified. Pick an IP-IP gateway in that viazone. *////GK/gk_gw_select_ipipgw_random: zonep: 0x4921EC04, tpp: 0x48FB789C, current_endpt: 0 ////GK/gk_gw_select_ipipgw_random: Gateway selection will start at the top of the linked list. use_count=0, current_endpt=0 ////GK/gk_gw_select_ipipgw_random: qelemp=0x0, loop_count=0 ////GK
Re: [OSL | CCIE_Voice] IPexpert lab 10: F/R hub-and-spoke topology vs. RSVP not applied to the GK DP's location
Hi Mohammed, I just checked in my lab, and without the passthrough enabled on the RSVP agent, the calls SCCP-SIP get negotiated to g729r8 with the MTPs in between. Makes sense, the SCCP phones do speak each variant of g729 that I know of, so with the RSVP MTP in there, they will use the g729r8 instead. Regarding the CAC : not sure - my idea is that you can use CUCM CAC for internal calls (br1-HQ), but for offnet calls (br1/hq - br2) going via a GK, you do CAC on the GK. Example scenario: intercluster call from UCM to CME max 5 calls allowed (enforced by GK CAC: bandwidth interzone). Within UCM cluster max 2 calls between BR1 and HQ, using RSVP. - Placing the GK in hub_none with no RSVP (like IPExpert suggests in lab10) allows up to 5 calls BR1-BR2, until the GK sends a ARJ. This oversubscribes the BR1-HQ link with 3 calls...Correct ? - Placing the GK in hub_none with RSVP enabled (set to 'mandatory' between hub_none and br1) would allow to restrict it to 2 calls over the BR1-HQ link, while the GK (bandwidth interzone...) still allows for an additional max of 3 calls for any remainingHQ-BR2 calls if needed. Result: no oversubscription of br1-hq link, and max 5 calls between UCM and CME. Correct? Basically, how do you limit the amount of calls to 2 over the br1-hq link when placing a call from br1 to br2 by placing the GK trunk in a non-RSVP enabled location (like ipexpert suggests) - please do tell me - I feel like I miss something crucial here if it's not correct what I am saying. cheers, Juan Op 31 maart 2012 23:32 schreef Mohammed Al Baqari baqari.voic...@gmail.comhet volgende: Hi Juan, ** ** Regarding GK separate DP, this is required to apply CAC when calls are initiated from both BR1 HQ to BR2. As you know CAC won’t apply for intra-location. Assume GK has HQ DP, in this case BR1-BR2 calls will have CAC applied, but HQ-BR2 calls won’t have CAC (same location). It’s always good to have GK in separate DP to have more control over CAC rules between locations. ** ** Regarding pass through, I believe you need to keep it. Consider this scenario: ** ** HQ is calling BR2 using G729. HQ is having SIP and SCCP phones. As you know, MTP can have one codec. Let’s assume that you configured G729r8. Now the problem is CUCM uses g729br8 for SCCP phones and g729r8 for SIP phones (SIP phones don’t support annex-b). Without pass-through, calls from SCCP won’t have MTP allocated since the codec isn’t matching. ** ** Please correct me if I am wrong and share your input. ** ** Regards, Mohammed Al Baqari ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Juan Lopez *Sent:* Thursday, March 29, 2012 8:26 AM *To:* George Goglidze *Cc:* CCIE Study *Subject:* Re: [OSL | CCIE_Voice] IPexpert lab 10: F/R hub-and-spoke topology vs. RSVP not applied to the GK DP's location ** ** Thx George - indeed what I read in SRND about asage of the passthrough. As for the lab (I understand your feedback is general, not lab-specific), there is no sRTP nor video, so my question was why passthrough would be included in the IPExpert solution guides. for the hub_none/RSVP: I'll need to continue searching I think - as indeed it might be a bad idea, but the lab doesn't care for good practices. so for the given topology, I wonder why the ipexpert solutions create a seperate DP for the GK. Anyone out there with some clear thoughts on this? Op 28 maart 2012 23:41 schreef George Goglidze gogli...@gmail.com het volgende: Hi, I'll just write regarding pass-through codec on MTP. You must have pass-through codec when the voice payload is not supported. For example, in case of SRTP traffic, the MTP resource without pass-through codec would not support encryption, but if you use pass-through codec it does not care if the media is encrypted or not, it will just substitute the IP's around in ip header without looking in payload. same goes for video streams using MTP! no pass-through no video, no game :) Now, if it's a good thing in a lab, I guess I'll let you decide :-) It's still good practice in my opinion to always have pass-through as first option. Regarding CAC on hub_none location, bad idea in general, if you need to limit one location just create different location for it. I'm not familiar with the questions, so can't say if you should be limiting that location or not. I guess I'd try NOT to overconfigure. and just do what's asked for. Hope this helps, ** ** On Wed, Mar 28, 2012 at 10:36 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: thx Baktha, I don't see a reason to use the pass-through for the lab purposes either... let's wait and see if someone comes up with some other opinion.* *** for the CAC: well, if the requirement states to have x calls between BR1 and HQ, my guess is that you need to have RSVP applied
Re: [OSL | CCIE_Voice] policy-map counters showing 24kbps for g729 with cRTP
Hi Datucha! Thx, but that is true for calculation of BW - but interface policy map counters should reflect values as being put on the link, I would guess. But from command reference, this output does not look uniform across platforms/features/ios - so i guess I should not be bothered with it too much. Cheers mate! On 01 Apr 2012, at 13:33, datucha123 datucha123 datucha...@gmail.com wrote: RSVP CAC does not care about the Compressed RTP (cRTP), like CUCM Locations Based CAC. On Sun, Apr 1, 2012 at 1:10 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: all, can someone explain where the value of 24kbps could come from for a g729 call with cRTP and going over MLP? From the output below, you can see 1* g729, cRTP OK, and MTP in between for RSVP. 5 minute offered rate 24000 bps. Only value I know of 24kbps equals g729, excluding L2 overhead but then NO cRTP however... cheers, Juan CorpHQ#sh policy-map int Virtual-Access3 Virtual-Access3 Service-policy output: WAN-EDGE1 Class-map: Voice (match-all) 19408 packets, 1207116 bytes 5 minute offered rate 24000 bps, drop rate 0 bps Match: ip dscp ef (46) Queueing Strict Priority Output Queue: Conversation 136 Bandwidth 62 (kbps) Burst 1550 (Bytes) (pkts matched/bytes matched) 0/0 (total drops/bytes drops) 0/0 compress: header ip rtp UDP/RTP (compression on, IPHC, RTP) Sent:19380 total, 19379 compressed, 735122 bytes saved, 427678 bytes sent 2.71 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps CorpHQ#sh ip rsvp reserv ToFrom Pro DPort Sport Next Hop I/F Fi Serv BPS 177.1.11.1177.2.11.1UDP 17156 17342 177.1.11.1 FF LOAD 24K 177.2.11.1177.1.11.1UDP 17342 17156 177.0.101.2 Vi3 FF LOAD 24K CorpHQ# CorpHQ#sh call active voice brie ... Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 2 CorpHQ#sh policy-map int Virtual-Access3 Virtual-Access3 Service-policy output: WAN-EDGE1 Class-map: Voice (match-all) 19408 packets, 1207116 bytes 5 minute offered rate 24000 bps, drop rate 0 bps Match: ip dscp ef (46) Queueing Strict Priority Output Queue: Conversation 136 Bandwidth 62 (kbps) Burst 1550 (Bytes) (pkts matched/bytes matched) 0/0 (total drops/bytes drops) 0/0 compress: header ip rtp UDP/RTP (compression on, IPHC, RTP) Sent:19380 total, 19379 compressed, 735122 bytes saved, 427678 bytes sent 2.71 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps CorpHQ#sh ip rsvp reserv ToFrom Pro DPort Sport Next Hop I/F Fi Serv BPS 177.1.11.1177.2.11.1UDP 17156 17342 177.1.11.1 FF LOAD 24K 177.2.11.1177.1.11.1UDP 17342 17156 177.0.101.2 Vi3 FF LOAD 24K CorpHQ# CorpHQ#sh call active voice brie ... Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 2 0: 33 13247580ms.1 +0 pid:0 Originate connecting dur 00:06:52 tx:20612/412240 rx:20614/412280 IP 177.1.11.30:25244 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a 0: 34 13247590ms.1 +0 pid:0 Originate connecting dur 00:06:52 tx:20614/412280 rx:20612/412240 IP 177.2.11.1:17342 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 2 ___ 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
This causes some confusion. I would like to agree on this, but from the Switch QOS configuration chapter, I copy: When you enter the auto qos voip cisco-phone command on a port at the network edge connected to a Cisco IP Phone, the switch enables the trusted boundary feature. If the packet does not have a DSCP value of 24, 26, or 46 or is out of profile, the switch changes the DSCP value to 0. When there is no Cisco IP Phone, the ingress classification is set to not trust the QoS label in the packet. *The policing is applied to the traffic matching the policy-map classification before the switch enables the trust boundary feature*. (aka. before using the 'mls qos trust cos' in tandem with 'mls qos trust device cisco-phone') So as far as I can decypher the above: for 'auto qos voip cisco-phone' this would mean that the policer is active. After policing, the trust boundary feature should be taken a look at. So with a phone connected: ok, the 'mls qos trust cos' would set the DSCP correctly *using the cos-dscp map * (not the setting from the policy map) and phone traffic has correct DSCP values. But for any PC traffic behind the phone... it is mostly untagged, and as we trust COS (mls qos trust cos) and becuase it's untagged: we use the default port cos =0, from which the DSCP is derived: DSCP =0. Result of this all: EF and CS3/AF31 policed, phone traffic has correct DSCP, PC traffic (like VTA, CUPC traffic) : DSCP 0... Does indeed not seem right... So therefore I think Vik is indeed correct in his blog stating the 'mls qos trust cos' is not used for any traffic matching the policy map (including CUPC, VTA traffic from PC) - but it contradicts a bit what I read in the Switch config guide QOS chapter. But hey, thing is that for QOS there is so much different approaches/explanations to be found on CCO. No so much uniformity there... From the same blog and looking in the lab, it also looks like the policy-map behaviour is different on a router compared to a switch: for a switch, DSCP is set to 0 (untrusted) if not explicitly set to trust within a class, whereas for a policy-map on a router, the DSCP is by default trusted within a class (not set to DSCP 0). But then: when we use the 'auto qos voip on a router', it creates the 'remark' class which sets DSCP to default = 0. Why not use the default class for this? Or better: why is there nowhere a reference stating set dscp default in the class class-default in the SRNDs, as I believe this is one of the things that the default class should be doing after all ? This would eliminate the need to have this 'remark' class, as the 'rogue' ef/cs3/af31 traffic would fall into the class-default class. So what is the reason not to find this command in the default class, and having auto qos generating the 'remark' class? hope to hear something from the experts on this :) Thx for the great blog on QOS and the links in them ! cheers, Juan Op 29 maart 2012 23:24 schreef George Goglidze gogli...@gmail.com het volgende: yes, that's correct. if you have MQC applied then trust does not take effect any more. On Thu, Mar 29, 2012 at 6:53 PM, Baktha Muralidharan muralic...@gmail.com wrote: Hello, Isn't it true that the trust stuff applies ONLY to those packets that are not caught by the class (in the qos policy)? For packets that are processed by the policy-map, you do a set cs3 (or whatever) under the policy. thanks, /Baktha -- Message: 1 Date: Thu, 29 Mar 2012 08:06:53 +0200 From: George Goglidze gogli...@gmail.com To: Chris devsin2...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] switch-QoS-Quick Question Message-ID: 4557c162-94d1-4c7e-963f-1ec20a440...@gmail.com Content-Type: text/plain; charset=us-ascii You have to trust DSCP on interface connected to the router. Routers do not set cos bits in dot1q header!!! Same goes for interface connected to CUCM. ___ 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
Re: [OSL | CCIE_Voice] switch-QoS-Quick Question
Steven, As far as I know, you cannot do policing on the egress fa0/1 towards the router... Only way to make sure MGCP towards the router is policed and down-marked, is by applying a policer on the ingress access port connecting the server I think. Is it a hard requirement to have it done on the fa0/1 uplink to the router? Let me know cheers, Juan Op 28 maart 2012 02:11 schreef steven moran smoran...@gmail.com het volgende: 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 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] policy-map counters showing 24kbps for g729 with cRTP
all, can someone explain where the value of 24kbps could come from for a g729 call with cRTP and going over MLP? From the output below, you can see 1* g729, cRTP OK, and MTP in between for RSVP. 5 minute offered rate 24000 bps. Only value I know of 24kbps equals g729, excluding L2 overhead but then NO cRTP however... cheers, Juan CorpHQ#sh policy-map int Virtual-Access3 Virtual-Access3 Service-policy output: WAN-EDGE1 Class-map: Voice (match-all) 19408 packets, 1207116 bytes 5 minute offered rate 24000 bps, drop rate 0 bps Match: ip dscp ef (46) Queueing Strict Priority Output Queue: Conversation 136 Bandwidth 62 (kbps) Burst 1550 (Bytes) (pkts matched/bytes matched) 0/0 (total drops/bytes drops) 0/0 compress: header ip rtp UDP/RTP (compression on, IPHC, RTP) Sent:19380 total, 19379 compressed, 735122 bytes saved, 427678 bytes sent 2.71 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps CorpHQ#sh ip rsvp reserv ToFrom Pro DPort Sport Next Hop I/F Fi Serv BPS 177.1.11.1177.2.11.1UDP 17156 17342 177.1.11.1 FF LOAD 24K 177.2.11.1177.1.11.1UDP 17342 17156 177.0.101.2 Vi3 FF LOAD 24K CorpHQ# CorpHQ#sh call active voice brie ... Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 2 CorpHQ#sh policy-map int Virtual-Access3 Virtual-Access3 Service-policy output: WAN-EDGE1 Class-map: Voice (match-all) 19408 packets, 1207116 bytes 5 minute offered rate 24000 bps, drop rate 0 bps Match: ip dscp ef (46) Queueing Strict Priority Output Queue: Conversation 136 Bandwidth 62 (kbps) Burst 1550 (Bytes) (pkts matched/bytes matched) 0/0 (total drops/bytes drops) 0/0 compress: header ip rtp UDP/RTP (compression on, IPHC, RTP) Sent:19380 total, 19379 compressed, 735122 bytes saved, 427678 bytes sent 2.71 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps CorpHQ#sh ip rsvp reserv ToFrom Pro DPort Sport Next Hop I/F Fi Serv BPS 177.1.11.1177.2.11.1UDP 17156 17342 177.1.11.1 FF LOAD 24K 177.2.11.1177.1.11.1UDP 17342 17156 177.0.101.2 Vi3 FF LOAD 24K CorpHQ# CorpHQ#sh call active voice brie ... Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 2 0: 33 13247580ms.1 +0 pid:0 Originate connecting dur 00:06:52 tx:20612/412240 rx:20614/412280 IP 177.1.11.30:25244 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a 0: 34 13247590ms.1 +0 pid:0 Originate connecting dur 00:06:52 tx:20614/412280 rx:20612/412240 IP 177.2.11.1:17342 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 2 ___ 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] IPexpert lab 10: F/R hub-and-spoke topology vs. RSVP not applied to the GK DP's location
In lab 10A, HQ devices are placed in DP_HQ with location HQ. GK/SIP trunk are placed in seperate DP, with location hub_none and no RSVP applied. So when calling from BR1 phone to CME over GK trunk, no CAC is enforced at the UCM. But the calls go over the FR link, as the FR setup is hub-and-spoke. Considering this topology (versus MPLS) : Shouldn't CAC be enforced (RSVP enabled on the hub_none location) to limit the amount of calls from BR1 to HQ - by extension to CME? I understand GK CAC can be used to contol the amount of calls between the UCM cluster and CME - but in this case we need to have CAC on the link BR1 HQ too. Also, what is the use of codec pass-through in the RSVP MTP ? Calls use g729 over the WAN (normally) - so what scenario exsits to have codec pass-through configured with g729 as fallback? why not simply use g729r8 as codec, without the passthrough (considering no video is involved here) thx for sharing thoughts! ___ 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] BW values used in lab10 - QOS
Is it possible the values listed in the PG are not correct (or better: other ones can be used based on QOS SRND) 28kbps for g729 when using MLP? When looking in the QOS SRND, page 33, it gives 30kbps for MLP (not taking FR overhead into consideration. With MLPoFR I get 30,8kbps) - and 28kbps for FR w/FRF12. So what to use: UCM SRND or QOS SRND? I'm used to calculate 17B for MPL-FR : 13B for MLP and 4B for FR. ___ 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] IPexpert lab 10: F/R hub-and-spoke topology vs. RSVP not applied to the GK DP's location
thx Baktha, I don't see a reason to use the pass-through for the lab purposes either... let's wait and see if someone comes up with some other opinion. for the CAC: well, if the requirement states to have x calls between BR1 and HQ, my guess is that you need to have RSVP applied between BR1 and HQ, and that GK trunk needs to be in the location HQ. If the GK is placed in a seperate DP with a location that has no RSVP with any other locations (like in the ipexpert proctor guide), then calls going from BR1 to CME over GK trunk will not be subject to RSVP cac , although these calls traverse the BR1 link to HQ.. I don't see how the rquirement can be met in that case... cheers, Juan Op 28 maart 2012 19:01 schreef Baktha Muralidharan muralic...@gmail.comhet volgende: Hi Juan As for CAC, guess one would have to consider - peak call volume - WAN QoS to decide if location-based CAC is [also] needed. If WAN QoS guarantees enough bandwidth for the anticipated peak volume, then, not sure we need the local-based CAC [RSVP or otherwise] As for pass-through, my experience is you only need the codec that will be used, in this case, the g.729r8. pass-through would be needed if you are plan to do such things as T.38 fax. thanks, /Batkha Message: 3 Date: Wed, 28 Mar 2012 16:33:29 +0200 From: Juan Lopez lopez.hernandez.j...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: Message-ID: CANpj6cyy3bDV8OvmNei= yqm5zpvbghxaalvpjyduku1wlqi...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 In lab 10A, HQ devices are placed in DP_HQ with location HQ. GK/SIP trunk are placed in seperate DP, with location hub_none and no RSVP applied. So when calling from BR1 phone to CME over GK trunk, no CAC is enforced at the UCM. But the calls go over the FR link, as the FR setup is hub-and-spoke. Considering this topology (versus MPLS) : Shouldn't CAC be enforced (RSVP enabled on the hub_none location) to limit the amount of calls from BR1 to HQ - by extension to CME? I understand GK CAC can be used to contol the amount of calls between the UCM cluster and CME - but in this case we need to have CAC on the link BR1 HQ too. Also, what is the use of codec pass-through in the RSVP MTP ? Calls use g729 over the WAN (normally) - so what scenario exsits to have codec pass-through configured with g729 as fallback? why not simply use g729r8 as codec, without the passthrough (considering no video is involved here) thx for sharing thoughts! -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120328/94dfa20d/attachment-0001.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 73, Issue 107 *** ___ 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] IPexpert lab 10: F/R hub-and-spoke topology vs. RSVP not applied to the GK DP's location
Thx George - indeed what I read in SRND about asage of the passthrough. As for the lab (I understand your feedback is general, not lab-specific), there is no sRTP nor video, so my question was why passthrough would be included in the IPExpert solution guides. for the hub_none/RSVP: I'll need to continue searching I think - as indeed it might be a bad idea, but the lab doesn't care for good practices. so for the given topology, I wonder why the ipexpert solutions create a seperate DP for the GK. Anyone out there with some clear thoughts on this? Op 28 maart 2012 23:41 schreef George Goglidze gogli...@gmail.com het volgende: Hi, I'll just write regarding pass-through codec on MTP. You must have pass-through codec when the voice payload is not supported. For example, in case of SRTP traffic, the MTP resource without pass-through codec would not support encryption, but if you use pass-through codec it does not care if the media is encrypted or not, it will just substitute the IP's around in ip header without looking in payload. same goes for video streams using MTP! no pass-through no video, no game :) Now, if it's a good thing in a lab, I guess I'll let you decide :-) It's still good practice in my opinion to always have pass-through as first option. Regarding CAC on hub_none location, bad idea in general, if you need to limit one location just create different location for it. I'm not familiar with the questions, so can't say if you should be limiting that location or not. I guess I'd try NOT to overconfigure. and just do what's asked for. Hope this helps, On Wed, Mar 28, 2012 at 10:36 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: thx Baktha, I don't see a reason to use the pass-through for the lab purposes either... let's wait and see if someone comes up with some other opinion. for the CAC: well, if the requirement states to have x calls between BR1 and HQ, my guess is that you need to have RSVP applied between BR1 and HQ, and that GK trunk needs to be in the location HQ. If the GK is placed in a seperate DP with a location that has no RSVP with any other locations (like in the ipexpert proctor guide), then calls going from BR1 to CME over GK trunk will not be subject to RSVP cac , although these calls traverse the BR1 link to HQ.. I don't see how the rquirement can be met in that case... cheers, Juan Op 28 maart 2012 19:01 schreef Baktha Muralidharan muralic...@gmail.comhet volgende: Hi Juan As for CAC, guess one would have to consider - peak call volume - WAN QoS to decide if location-based CAC is [also] needed. If WAN QoS guarantees enough bandwidth for the anticipated peak volume, then, not sure we need the local-based CAC [RSVP or otherwise] As for pass-through, my experience is you only need the codec that will be used, in this case, the g.729r8. pass-through would be needed if you are plan to do such things as T.38 fax. thanks, /Batkha Message: 3 Date: Wed, 28 Mar 2012 16:33:29 +0200 From: Juan Lopez lopez.hernandez.j...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: Message-ID: CANpj6cyy3bDV8OvmNei= yqm5zpvbghxaalvpjyduku1wlqi...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 In lab 10A, HQ devices are placed in DP_HQ with location HQ. GK/SIP trunk are placed in seperate DP, with location hub_none and no RSVP applied. So when calling from BR1 phone to CME over GK trunk, no CAC is enforced at the UCM. But the calls go over the FR link, as the FR setup is hub-and-spoke. Considering this topology (versus MPLS) : Shouldn't CAC be enforced (RSVP enabled on the hub_none location) to limit the amount of calls from BR1 to HQ - by extension to CME? I understand GK CAC can be used to contol the amount of calls between the UCM cluster and CME - but in this case we need to have CAC on the link BR1 HQ too. Also, what is the use of codec pass-through in the RSVP MTP ? Calls use g729 over the WAN (normally) - so what scenario exsits to have codec pass-through configured with g729 as fallback? why not simply use g729r8 as codec, without the passthrough (considering no video is involved here) thx for sharing thoughts! -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120328/94dfa20d/attachment-0001.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 73, Issue 107 *** ___ 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
Re: [OSL | CCIE_Voice] CUPC Signalling
not sure as just starting with this, but I believe the CUPC, when logging in, will download the VM profile setup in CUPS, and then use a direct connection to CUC over HTTPS to access the VM/MWI - whether it uses SIP (softphone mode) or CTI (deskphone mode) to talk to CUCM. Op 22 maart 2012 07:33 schreef Ken Wyan kew...@gmail.com het volgende: Take typical integration of CUCM to Unity Connection sccp integration CUPC client is used to check voicemail / mwi CUPC Client --sip signalling- CUPS Server CUCM Server --sccp signalling- Unity Connection CUPC Client --sip signalling- CUCM Server CUCM Server -sip signalling- CUPS Server CUPC Client - ? signalling Unity Connection When CUPC client access voice mail mwi indications , does it use SCCP signalling ? OR does CUCM acts as a signalling proxy between CUPC client Unity connection server for sccp/sip translation? Ken ___ 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] CUPC no contact details -
I cannot make calls to listed contacts, from the CUPC in softphone mode (contacts added through the CUPC user pages) Dialing from the keypad however is not a problem (so I can make calls manually from the CUPC in softphone mode). Few hours later now with no solution so does anyone have an idea? In CUPC the phone button and VM (enveloppe) are greyed out. The server health shows all is ok/green (apart for secure messaging and ldap - which is indeed not configured). Config for user YYY: 1) CUPC phone added on UCM, owner ID = YYY (per deployment guide, not sure why this is) 2) CUPC line is shared with deskphone 3) line appearance of CUPC phone is not associated with user YYY (only the line appearance of the deskphone is associated with user YYY) -- so for user YYY only presence state of the line on his deskphone is communicated 4) user YYY only associated with deskphone (not with CUPC, as deskphone/CTI control is not intended for the CUPC) When looking on the CUPS user management for user YYY: 1) I do not see the Phone/Manager fields filled in (they are in UCM) 2) I see 2 devices for that user: the 7962 phone, and the CUPC. The CUPC capabilites show only : 'supports instant messaging' - and no lines (no idea why the CUPC is listed anyway - as that user is not associated with the CUPC device on UCM) I tried all kinds of configuration variartions - the above is what according to me it should be just to get the softphone working. Any advise/guidance is welcome ! cheers, 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] cups...
CUPS - thought it was gonna be easy ;) but I do not seem to get it right - it's like it will not fit together And the more I try, to less clear it becomes... mainly about: line appearance for a deskphone is associated with a user: OK, to provide presence info to CUPS server. But from the CUPS deployment guide, to configure the CUPC client, you should also associate the line appearance of the CUPC device with the user (p3-18). No problem, but when you then take a look at the presence viewer @ CUPS, you notice that the CUPC does not have a line nor presence info, just an indication of it's (IM only) capabilities. (IM only I guess because there's no AD integration (?) and I cannot call contacts added via the Sync Agent (CUPC user pages) - but that's another story) When you on the contrary associate the CUPC too with the user on the CUCM user pages, then the line + presence state appears in the presence viewer from CUPS. So do you also need to associate the CUPC device with the user altogether ? And shouldn't you then see the presence state of each device (CUPC deskphone) seperately for that user in another CUPC client - I can only see the 'aggregated' presence state... any thoughts are more than welcome. I call it a day... ;) the morning will be wiser than the evening was (I hope) ___ 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] Fwd: CUC and AAR....need some help
Still investigating this but can't seem to find the answer. Form the SRND it states: By default, the directory number configuration retains the AAR leg of the call in the call history, which ensures that the AAR forward to the voice messaging system will select the proper voice mailbox This I understand, and gives an indication to what I see: that the AAR function retains the RDNIS of the called party by default. So when I press the VM button on SB phone, and AAR is invoked, CUC's HP external number mask/AAR group is used to get the call over to CUC, but the RDNIS of the called party (CUC's HP) is sent along. I think I am not grasping something basic here - I do not find anything stating that AAR in combination with the VM button does not work. Or I am not seeing things clear anymore. So feedback is appreciated. So if anyone can point me in a good direction it would be great - getting despirate over here ;-D cheers, Juan -- Doorgestuurd bericht -- Van: Juan Lopez lopez.hernandez.j...@gmail.com Datum: 19 maart 2012 21:58 Onderwerp: CUC and AARneed some help Aan: CCIE Study ccie_voice@onlinestudylist.com All, I need some help: I have CUC setup at the HQ. When using AAR between SB and HQ, when the SB phones presses the 'voicemail' button, I would have thought this constitutes a direct call (aka: no RDNIS sent) to voicemail. However, I notice the call is being sent out with the RDNIS of the voicemail hunt pilot. So the caller is connected to the welcome prompt instead... this really strikes me - I did not see that one coming... Is this normal? What can be done so that a direct call to VM from a SB phone - to listen to his/her voicemail messages - is sent to the subscribers' inbox instead? (I cannot uncheck de delivery of RDNIS at the SB GW, as this needs to be sent for forwarded calls) any help is greatly 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
Re: [OSL | CCIE_Voice] Fwd: CUC and AAR....need some help
Hi Kevin, thx for the feedback - that is indeed the workaround I was just hoping to get more feedback on this - sharing knowledge instead of figuring everything out by yourself, is what the idea of a public forum like OSL is/should be about - but apparently apart from Baktha and you it was unidirectional silence unfortunately. Or there are not that many going for the ccie voice lately, that could also be the case ;) cheers thx, Juan Op 20 maart 2012 14:31 schreef Kevin Spicer ke...@kevinspicer.co.uk het volgende: Hi Juan, Just playing with this in the lab and I think a workaround is to add a routing rule (in CUC Call Management Call Routing Forwarded Routing Rules). Add to the top of the list a rule that sends to the 'Attempt Sign In' Conversation where the forwarding station is the voicemail pilot number (i.e. the number which shows as the redirtecting number in the q931 trace when you hit the messages key). Initial testing this seems to work okay. Kevin On Tue, Mar 20, 2012 at 8:04 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: Still investigating this but can't seem to find the answer. Form the SRND it states: By default, the directory number configuration retains the AAR leg of the call in the call history, which ensures that the AAR forward to the voice messaging system will select the proper voice mailbox This I understand, and gives an indication to what I see: that the AAR function retains the RDNIS of the called party by default. So when I press the VM button on SB phone, and AAR is invoked, CUC's HP external number mask/AAR group is used to get the call over to CUC, but the RDNIS of the called party (CUC's HP) is sent along. I think I am not grasping something basic here - I do not find anything stating that AAR in combination with the VM button does not work. Or I am not seeing things clear anymore. So feedback is appreciated. So if anyone can point me in a good direction it would be great - getting despirate over here ;-D cheers, Juan -- Doorgestuurd bericht -- Van: Juan Lopez lopez.hernandez.j...@gmail.com Datum: 19 maart 2012 21:58 Onderwerp: CUC and AARneed some help Aan: CCIE Study ccie_voice@onlinestudylist.com All, I need some help: I have CUC setup at the HQ. When using AAR between SB and HQ, when the SB phones presses the 'voicemail' button, I would have thought this constitutes a direct call (aka: no RDNIS sent) to voicemail. However, I notice the call is being sent out with the RDNIS of the voicemail hunt pilot. So the caller is connected to the welcome prompt instead... this really strikes me - I did not see that one coming... Is this normal? What can be done so that a direct call to VM from a SB phone - to listen to his/her voicemail messages - is sent to the subscribers' inbox instead? (I cannot uncheck de delivery of RDNIS at the SB GW, as this needs to be sent for forwarded calls) any help is greatly 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 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] Vik's blog on CFUR - manipulate XML display of called number on calling device.
Baktha, I read your response on Vik's blog for CFUR. I try to manipulate the called number on the XML display of the caller's phone so that it would look like an internal call - by setting a called party transformation at the RP used by CFUR - like you suggest. Only thing is that this does not work whenever you have called party transformations at the CUCM egress gateway - these even do overwrite the XML display for the called number on the calling device according my tests. So how does this work taking your response into consideration, where you say to work with called party transformations on the egress gateway? Does this work for you? - if so, would you want to share how you setup that part of the dialplan? cheers, Juan Op 19 maart 2012 18:47 schreef Baktha Muralidharan muralic...@gmail.comhet volgende: Steve Congratulations!! Enjoy the well-deserved break! /Baktha ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com 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] [OSL | CCIE_VOICE] SRST and MWI unsolicited notify.
Justin, a reload of cue often solves these kinds of issues in my case. Or you might try to bump the mwi by setting it to solicited, then unsolicited on the cue, with refresh mwi. I dont see anything wrong on the config. On 18 Mar 2012, at 04:43, Justin McIntyre justin.mcint...@blackbox.com wrote: I am wondering if anyone one can give me some hints on where to troubleshoot swi updates not working with unsolicited notify. I have my ccn subsystem output and my running config input below. My issue is that I do not see any Sip Notify messages being sent out. SHOW CCN SUB SIP se-10-10-115-2# show ccn subsystem sip SIP Gateway:10.10.115.1 SIP Port Number:5060 DTMF Relay: sip-notify,sub-notify MWI Notification: unsolicited MWI Envelope Info: disabled Transfer Mode: bye-also SIP RFC Compliance: Pre-RFC3261 Running config…. Current configuration : 9319 bytes ! ! Last configuration change at 03:20:35 GMT Sun Mar 18 2012 by justin ! NVRAM config last updated at 03:20:30 GMT Sun Mar 18 2012 by justin ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname SiteC-RTR ! boot-start-marker boot-end-marker ! logging message-counter syslog ! no aaa new-model memory-size iomem 20 clock timezone GMT 0 network-clock-participate wic 3 network-clock-select 1 E1 0/3/0 dot11 syslog ip source-route ! ! ip dhcp excluded-address 10.10.202.1 10.10.202.119 ip dhcp excluded-address 10.10.202.130 10.10.202.254 ! ip dhcp pool SiteC-Static host 10.10.202.130 255.255.255.0 client-identifier 0100.1930.5d0b.d7 default-router 10.10.202.1 option 150 ip 10.10.210.11 10.10.210.10 ! ip dhcp pool SiteC-PHONES network 10.10.202.0 255.255.255.0 default-router 10.10.202.1 option 150 ip 10.10.210.11 10.10.210.10 ! ! ip cef ip domain name ipexpert.com no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip ! ! ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone exit dualtone conference cadence 400 ! voice class custom-cptone entry dualtone conference cadence 200 ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 2 rule 1 /^4...$/ /7796\0/ ! voice translation-rule 900 rule 1 /^4...$/ /+144207796\0/ ! ! voice translation-profile 4digitDNIS translate called 1 ! voice translation-profile 8digitANI translate calling 2 ! voice translation-profile e164ANI translate calling 900 ! ! voice-card 0 dsp services dspfarm ! ! crypto pki trustpoint TP-self-signed-1655997933 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-1655997933 revocation-check none rsakeypair TP-self-signed-1655997933 ! ! crypto pki certificate chain TP-self-signed-1655997933 certificate self-signed 01 3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 31363535 39393739 301E 170D3132 30323131 31343135 30385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 36353539 39373933 3330819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100E6E4 0307318B 5E2C94CB 7E2A83CF 6F99AE89 10D93A2F 38BDEB71 95C5695E 4BEA4075 6AE144A6 961F0630 CCECF324 EDB7E128 64BA6A7F 289758F4 8C5268BF C36E7746 40F8CDEE 8D5EE734 0BADF088 8B1B933F BFC9CD9C B25CC8C7 D68AFDEC FC8AC19F 6200D364 7F82FD03 7B43C688 DF02DF00 31F09D24 A21421D8 26CA303C ACDF0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603 551D1104 18301682 14425232 2D525452 2E697065 78706572 742E636F 6D301F06 03551D23 04183016 80142071 A4496B36 760E3BB9 7BA7ECB2 3441D434 EA54301D 0603551D 0E041604 142071A4 496B3676 0E3BB97B A7ECB234 41D434EA 54300D06 092A8648 86F70D01 01040500 03818100 342F96C6 47F5E13E 1EB508A2 6A614A3F 9C975E35 B6690F3A 74E75E4D E88F802B 6A09E40D 3E86128D BDFD34EC D2C0FF33 E3DDB0B8 495F5600 A1921326 11E4851E DED6D532 C2B597B9 1755F18E 8A71C86B A6D3D77A
Re: [OSL | CCIE_Voice] DND/HLog huntgroup
pls disregard - found there is a difference on how the DNd is invoked: through SK : ringer muted, through auto-logout: huntgroup call does not get presented anymore. Op 13 maart 2012 12:59 schreef Juan Lopez lopez.hernandez.j...@gmail.comhet volgende: Dear all, probably something stupid, but when in CME SRST, the only way to really logout of a huntgroup is by using the HLog softkey. The DND only turns the ringer off. Per the CME admin guide, when the phone is in DND state, nor a huntgroup call nor the direct call to that phone should be presented - which is not what happens. anyone an idea? there is 1 shared line on the huntgroup phones, but that shared line is not part of the huntgroup. cheers, 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
Re: [OSL | CCIE_Voice] CME as SRST ... giving unforeseen issues... please help
The same issue. so after digging into old posts - I find this is a known issue when configuring auto-provison-all? A reboot is needed to have the DNs register back if the phone ever got unregistered (back to UCM or manual reset after ephone/ephone-dn customization). There *seems* to be a lot of bugs related to CME as SRST when going through the old postings: bugs related to ephone-template usage, privacy settings not working under telephony-service etc. Would there be any chance someone out there has a compiled list of what is not working as it should for CME as SRST? II am sure everyone going through CME SRST has faced this kind of question ... cheers, Juan Op 10 maart 2012 21:14 schreef chase mergenthal cm3_...@hotmail.com het volgende: Try removing the following and test: srst ephone template 1 srst dn template 1 -Chase -- If winners never quit and quitters never win, then who coined the phrase, Quit while you’re still ahead.? -- Date: Sat, 10 Mar 2012 20:49:29 +0100 From: lopez.hernandez.j...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME as SRST ... giving unforeseen issues... please help So I must be doing something wrong for CME as SRST: I let the phones be auto-provisioned. No problem there. After going into fallback, if the phone resets, it does loose all of its lines - or when waiting a long time, it might get one of it's lines back - the rest shows INVALID : ephone-3[2] Mac:001E.4A92.4FDE TCP socket:[5] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:12 privacy:0 IP:177.2.11.246 39701 7962 keepalive 0 max_line 6 available_line 6 button 1: dn 4 number 2002 CH1 INVALID CH2 INVALID CH3 INVALID CH4 INVALID CH5 INVALID CH6 INVALID CH7 INVALID CH8 INVALID button 2: dn 3 number 2010 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared Preferred Codec: g711ulaw What am I forgetting ? My CME as SRST config: telephony-service sdspfarm units 3 sdspfarm transcode sessions 2 sdspfarm tag 1 br1-cfb sdspfarm tag 2 br1-xcode no privacy privacy-on-hold conference hardware srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 5 max-dn 50 preference 2 ip source-address 177.1.254.2 port 2000 no service directed-pickup timeouts interdigit 7 system message Your CME current options time-zone 23 time-format 24 date-format dd-mm-yy max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files version-stamp Jan 01 2002 00:00:00 Please help 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 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] CME as SRST ... giving unforeseen issues... please help
Hi Mohd, that works :) but the idea is to know what is not working as expected when using a certain approach... - like the needed reboot for the issue listed below. eg: I would think that cBarge works for auto-provision none, as no learned ephone/ephone-dn customization is needed (only creating a conference ephone-dn, but this stands apart from what is considered to be auto-provisioned). However, from Vik's blog it does not look like this - no idea why. therefore: has anyone compiled something that lists the known bugs for CME as SRST? thanks for the help! cheers, Juan Op 10 maart 2012 21:52 schreef Mohd Baqari baqari.voic...@gmail.com het volgende: Try auto provision none and test. Regards, Mohammed Al Baqari Sent from my iPhone On Mar 10, 2012, at 11:49 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: So I must be doing something wrong for CME as SRST: I let the phones be auto-provisioned. No problem there. After going into fallback, if the phone resets, it does loose all of its lines - or when waiting a long time, it might get one of it's lines back - the rest shows INVALID : ephone-3[2] Mac:001E.4A92.4FDE TCP socket:[5] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:12 privacy:0 IP:177.2.11.246 39701 7962 keepalive 0 max_line 6 available_line 6 button 1: dn 4 number 2002 CH1 INVALID CH2 INVALID CH3 INVALID CH4 INVALID CH5 INVALID CH6 INVALID CH7 INVALID CH8 INVALID button 2: dn 3 number 2010 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared Preferred Codec: g711ulaw What am I forgetting ? My CME as SRST config: telephony-service sdspfarm units 3 sdspfarm transcode sessions 2 sdspfarm tag 1 br1-cfb sdspfarm tag 2 br1-xcode no privacy privacy-on-hold conference hardware srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 5 max-dn 50 preference 2 ip source-address 177.1.254.2 port 2000 no service directed-pickup timeouts interdigit 7 system message Your CME current options time-zone 23 time-format 24 date-format dd-mm-yy max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files version-stamp Jan 01 2002 00:00:00 Please help 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 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] VPIM
how do people train this part with their own HW if VPIM is not in the CUC demo license? Is there a workaround? cheers, Juan 2012/3/9 Cisco Nut rafayc...@gmail.com Hi Vik Its my CUC, I am able to add location in my CUE, when I send message to 2125002 from 3002, I hear its telling me sending message to 5002 location 212 but message never gets deliverd, instead I get a message in 3002 that message is not delivered to 5002. I guess its due to the fact CUC dont have VPIM license and it wont accept or send VPIM messages. Regards Rafay On Thu, Mar 8, 2012 at 11:05 PM, Vik Malhi vma...@ipexpert.com wrote: Is this your CUC or CUE? The demo license on CUC does not allow you to add VPIM locations. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 8, 2012, at 4:51 PM, Cisco Nut wrote: Hi I am running a Demo license on my CUE server, when I add VPIM location it gives me an error that VPIM is a license feature, Please let me kow how you guys are working on VPIM in your home labs. Please see below exact error I get when I tried adding VPIM location. Regards Status [image: error] The requested operation would result in a license violation. [image: error] Unable to create VPIM Location ___ 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] CME CBarge Called phone's display info
When using cBarge under CME, the admin guide states that the 3rd party sees to Barge on the display. In my case I see the ad-hoc conference ephone dn's name instead. Is there something that can be done to have it changed to 'To Barge' instead of 'From xxx'? Also, when the barger leaves and the CFB is tore down, the call is converted back to a 2party call, but the display for the 3rd party is not updated to reflect the other side's CNAM. Is that normal? thx, 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] directory services
Is there a way to have a user created service on UCM been placed before the internal Corporate Directory service on the phone display? Aka: determine the order in which the subscribed services appear? In my setup I try to use the UCM directory for CME, and therefore disabled the missed/places/received services in the UCM (they would appear twice at the CME phone otherwise). But when recreating those services on the UCM and then resubscribing the necessary UCM phones, these 'missed/placed/recieved' calls appear after the Corporate and Personal Directory listing... Secondly, on CME: is there a way to manipulate the XML page that is presented on the phone when pressing the Directory button? Maybe it would be a better idea to remove the missed/places/received calls provisioned by CME. The only reference I find about this is that you can disable the Local Directory (no service local-directory). thx! 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
Re: [OSL | CCIE_Voice] Mobile Voice Access
very good to know ! thx for the tip Vik, I was unaware of this. So the workaround is: set RD = ANI and eventually use application dial rules to map the RD to the route plan used. 2012/3/7 Vik Malhi vma...@ipexpert.com Welcome Mathew. Juan- sporadic in all 7.x. Sometimes works, sometimes doesn't. My advice- don't go anywhere near partial matching for the lifetime of your CCIE-V Lab prep. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 6, 2012, at 9:26 PM, Juan Lopez wrote: is this a general issue? I did not have this kind of problem: RP set to +e164 and partial match did not give any problems in my case... Used UCM version is 7.0.1.11000-2 thx! Juan 2012/3/6 Mathew Miller miller.mat...@gmail.com Thanks much Vik! I have confirmed that if I have a 10 digit RD and get 7 digit ANI and set to Partial Match that every time I experience this behavior with the annunciator. When I changed it to full match with a 7 digit RD it fixed my problem. On Tue, Mar 6, 2012 at 11:08 AM, Vik Malhi vma...@ipexpert.com wrote: I'm guessing you are using partial matching of the ANI versus the RD. Make it a complete match- in other words whatever the ANI is (from the PSTN in the Q931 debug) make this the RD number. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 6, 2012, at 8:29 AM, Mathew Miller wrote: Hello All, I have setup mobile voice access for use to be able to dial extensions using enterprise access in my home lab. 100 times out of 100 it works just peachy… I setup my dial-peer on my router, I setup the IVR service on the router, I setup my RDP and RD, along with my Mobile Voice Access number and set my Service Parameters. In my lunch dates I set this up exactly like I have done 100 times at home and everything seems to work fine until I try to dial the the extension I want to call and I get the Annunciator telling me my call can't be completed as dialed. I have checked my re-routing CSS and all the steps in setup and have access to internal extensions so I don't know what I am doing wrong. I have tried to create the issue in my home lab and cant seem to do it. It work EVERY time in my home lab. Can anyone think of something I may be overlooking? ___ 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
Re: [OSL | CCIE_Voice] directory services
Hi Mathew, this helped a LOT - I just finished testing as I wanted to nail this topic down once and forever as for me it was never clear. What I learned here is that when using the external service provisioning method - you also disable the VM button(!) I never paid attention to this. But mainly what you do is using the 'directory URL' and the 'messages URL' defined in enterprise pars. So if these are empty - and for messages it is by default - you indeed get nothing. Internal service provisioning just gives you everything the phones is subscribed to: enterprise and manually added services - all done through the phone services pages. So if there is someone out there that can tell how CME phones resolve the ' http://CME/locadirectory' so I can then find those CME files in flash/nvram, it would be great to have a look if it's possible to remove the missed/places/received calls). thanks for that feedback. Juan 2012/3/7 Mathew Miller miller.mat...@gmail.com Hi Juan, I just worked through this in my lab the other day. Here are some things that helped me. Subscribed services (ie non-enterprise) will be put in alphabetical order. Some things that helped me out were the following: Understand the difference between Internal/External Provisioning for services. If you have setup phones for external provisioning they will use the directories URL in Enterprise Parameters. That will also break the voicemail button. The External URL really is only a pointer to whatever enterprise services you have setup and are currently enabled. So for example if you have the enterprise URL left at default xmldirectory.jsp and you deleted or disabled all of your enterprise services nothing will show up. Internal will use the Internal services you have setup that have the enterprise flag checked. Both will use the external URL that you have setup and will also use the External Directory URL in enterprise parameters. If this is the default and you disabled all of your enterprise URLs except voicemail then nothing will show up. If you set your phone to use both for service provisioning you can create a custom XML directory and order your directories however you like. You just need a place to host it. (think of a place that has a webserver built in) This page gives part of a code sample for access directories example http://www.netcraftsmen.net/component/content/article/70-unified-communications/714-how-to-disable-corporate-directory-in-cucm-7x.html See the section that says (Added 8/31/2010) To fill in the rest of this download the Cisco IP Phone SDK and open the supporting PDF to figure out the format for creating a CiscoIPPhoneDirectory HtH Matt From: Juan Lopez lopez.hernandez.j...@gmail.com Date: Wed, 7 Mar 2012 09:26:09 +0100 To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] directory services Is there a way to have a user created service on UCM been placed before the internal Corporate Directory service on the phone display? Aka: determine the order in which the subscribed services appear? In my setup I try to use the UCM directory for CME, and therefore disabled the missed/places/received services in the UCM (they would appear twice at the CME phone otherwise). But when recreating those services on the UCM and then resubscribing the necessary UCM phones, these 'missed/placed/recieved' calls appear after the Corporate and Personal Directory listing... Secondly, on CME: is there a way to manipulate the XML page that is presented on the phone when pressing the Directory button? Maybe it would be a better idea to remove the missed/places/received calls provisioned by CME. The only reference I find about this is that you can disable the Local Directory (no service local-directory). thx! 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 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] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE
Datucha, thanks again for the clarification and for the very useful tip you gave about the MTP restricting the use of dtmf notify - and the need to remove the RTP NTE from the rtp stream when going from inband to out of band DTMF. I am sure this will come in very useful once I start with adding CUE to the picture. cheers, Juan 2012/3/4 datucha123 datucha123 datucha...@gmail.com There is a little restriction for SIP Notify DTMF for CUCM. Juan is correct - You need to enable the accept unsollicited notify in sip security profile so that CUCM will be able to receive Notify DMTFs. But if MTP is checked, then the Notify option will not work. Also for outbound Notify DTMF from CUCM, it is not necessary to enable accept unsollicited notify. It will still work. As for KPML and Notify internetworking - it is supported of CUCME (CUBE). If you are using Inband (RFC2833) on one side and any out of band DTMF on other side, then you have to configure the following command on RFC2833 side dial-peer, so that the Router will strip out the inband DTMF's and leave only out of band for outgoing dial-peer: *dtmf-relay rtp-nte digit-drop* On Sun, Mar 4, 2012 at 10:23 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: for option 1 below - you could try to set under the sip security profile accept unsollicited notify so that on the BR2 side, you use sip-notify as DTMF relay on both CUE and UCM SIP dialpeers. Let us know if thay might help cheers, Juan 2012/3/4 Juan Lopez lopez.hernandez.j...@gmail.com Hi Justin, from reading the mail it looks like on the SIP dialpeers on the BR2, you use the rtp-nte (inband) dtmf-relay method? can you try and let us know: 1) use SIP-NOTIFY on both SIP dialpeers at BR2? (not sure if UCM supports this - in SRND it states a UCM SIP trunk uses RTP-NTE or possibly SIP-KPML) if 1 does not work: 2) use sip notify on CUE dialpeer and sip-kpml on sip dialpeer to UCM. Not sure here whether the CUBE at branch 2 supports notify - kpml dtmf interworking... The idea is to have DTMF between UCM and CME out of band... From SRND I read that SIP at UCM uses RTP-NTE or possibly SIP-KPML, so it rules out to use the SIP-NOTIFY on the dialpeer at branch2 pointing to UCM (not tested yet) to keep it all out of band - but this is the way to rule out an MTP 2012/3/3 Justin McIntyre justin.mcint...@blackbox.com Ok. For those who are interested I have resolved my issue. By selecting the Media Termination Point Required option within the SIP trunk I was able to resolve my media stream to an MTP prior to connection to the CME. This allowed in-band/Out of Band DTMF traversal. Note that when you select the MTP required option within your sip trunk to pay special attention to the device pool and region settings upon with the MTP that you will resolve to will lie. The MTP will not inherit the Device Pool settings from the Sip trunk depending on your configuration. This was a really good learning experience and if anyone is curious as to any further details please let me know. I am however un-clear on one thing and maybe someone can help me out. I remember using Sip-Notify within my CUE dial-peer and within CUE configuration the last time I ran this lab. For some reason I could not get SIP-Notify to work in any case at all that I tried this time around. If anyone has any clarity on this I would be most appreciative, I'd hate to see, please configure a sip trunk between UCM and CME location at to reach the CUE VM pilot. Note: use of an MTP on the SIP trunk is not allowed in the lab. Plus who knows when a customer site may encounter this situation. Thanks everyone. *!*!*!*Thanks to Chase and Vik as they were pertinent in my resolution.*!*!*!* 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 recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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
Re: [OSL | CCIE_Voice] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE (Juan Lopez)
it still have to try it out with CUE, but from debug ccsip messages at BR2, it looks like the UCM is sending NOTIFY to the BR2 and that these are understood by the BR2 router ... so I'm not sure why this would not work: (debug kpml all and debug voip rtp session named-event do not produce any output, so only notify method is used) 2012/3/4 Justin McIntyre justin.mcint...@blackbox.com Hello, I just wanted to update that the SIP-Notify did not work when set at the UCM SIP trunk, the BR2 CUE Dial-peer and also within CUE configuration. At this point it only seems to be working when using the MTP to terminate between UCM SIP trunk and CME dial-peer. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 Please note new e-mail address justin.mcint...@blackbox.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, March 04, 2012 1:23 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 73, Issue 8 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: service-policy on trunk ports (Vik Malhi) 2. Re: CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE (Juan Lopez) 3. Re: CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE (Juan Lopez) -- Message: 1 Date: Sat, 3 Mar 2012 19:00:34 -0800 From: Vik Malhi vma...@ipexpert.com To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] service-policy on trunk ports Message-ID: cada9c18-6add-4058-b744-194805b8d...@ipexpert.com Content-Type: text/plain; charset=windows-1252 This is and has been for a long time been a limitation on the 3750- the show policy-map command doesn't work:-( Vik Malhi ? CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 3, 2012, at 12:59 PM, Ken Wyan wrote: I have following scenario (Tested in Proctorlabs Rack). HQ Switch Fa1/0/1 (trunk port) ---connect to-- HQ Router Fa0/0 (with sub-interfaces) I want to apply a service-policy to mgcp packets going through this link. I configured access-list , class-map , policy-map applied to switch interface. But I can't see any mgcp packets matching HQ-3750#show policy-map interface fastEthernet 1/0/1 FastEthernet1/0/1 Service-policy input: mgcp Class-map: mgcp (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 100 Class-map: class-default (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any 0 packets, 0 bytes 5 minute rate 0 bps interface FastEthernet1/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan 10 switchport mode trunk speed 100 duplex full mls qos trust dscp service-policy input mgcp Now same thing I configured on HQ Router ( Fa0/0 interface) , then I can see packets are matching with service policy. What can be the reason? (Switch accepts service-policy in input direction only , hence I applied service-policy in output direction on Router port) Can this be a limitation for trunk (multi-vlan) ports on switches ? Ken ___ 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/ -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120303/1ed256a1/attachment-0001.html -- Message: 2 Date: Sun, 4 Mar 2012 06:34:42 +0100 From: Juan Lopez lopez.hernandez.j...@gmail.com To: Justin McIntyre justin.mcint...@blackbox.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE Message-ID: canpj6cze1nfeesk2ou6pgudzgo9cbyd5nuffqboxkdzdptk...@mail.gmail.com Content-Type: text
Re: [OSL | CCIE_Voice] Answer too soon timer on Remote Destination (for SNR)
Baktha, I'd think so - the delay before ringing gives a timeout after which the Q931 setup is sent to the PSTN. No use to take the call back to UCM before that message is even sent. If the PSTN sends a connect within the default 1.5s after this setup message, then it would make sense to take the call back- as the mobile probably not reachable. cheers, Juan 2012/3/4 Baktha Muralidharan muralic...@gmail.com Just a confirmation request-- Is it true that the Answer too soon timer on Remote Destination config starts *AFTER* Delay Before Ringing timer. It seems to, but wanted to be sure. Help page is not very clear about it. thanks in advance, /Baktha ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com 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] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE
Hi Justin, from reading the mail it looks like on the SIP dialpeers on the BR2, you use the rtp-nte (inband) dtmf-relay method? can you try and let us know: 1) use SIP-NOTIFY on both SIP dialpeers at BR2? (not sure if UCM supports this - in SRND it states a UCM SIP trunk uses RTP-NTE or possibly SIP-KPML) if 1 does not work: 2) use sip notify on CUE dialpeer and sip-kpml on sip dialpeer to UCM. Not sure here whether the CUBE at branch 2 supports notify - kpml dtmf interworking... The idea is to have DTMF between UCM and CME out of band... From SRND I read that SIP at UCM uses RTP-NTE or possibly SIP-KPML, so it rules out to use the SIP-NOTIFY on the dialpeer at branch2 pointing to UCM (not tested yet) to keep it all out of band - but this is the way to rule out an MTP 2012/3/3 Justin McIntyre justin.mcint...@blackbox.com Ok. For those who are interested I have resolved my issue. By selecting the Media Termination Point Required option within the SIP trunk I was able to resolve my media stream to an MTP prior to connection to the CME. This allowed in-band/Out of Band DTMF traversal. Note that when you select the MTP required option within your sip trunk to pay special attention to the device pool and region settings upon with the MTP that you will resolve to will lie. The MTP will not inherit the Device Pool settings from the Sip trunk depending on your configuration. This was a really good learning experience and if anyone is curious as to any further details please let me know. I am however un-clear on one thing and maybe someone can help me out. I remember using Sip-Notify within my CUE dial-peer and within CUE configuration the last time I ran this lab. For some reason I could not get SIP-Notify to work in any case at all that I tried this time around. If anyone has any clarity on this I would be most appreciative, I'd hate to see, please configure a sip trunk between UCM and CME location at to reach the CUE VM pilot. Note: use of an MTP on the SIP trunk is not allowed in the lab. Plus who knows when a customer site may encounter this situation. Thanks everyone. *!*!*!*Thanks to Chase and Vik as they were pertinent in my resolution.*!*!*!* 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 recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE
for option 1 below - you could try to set under the sip security profile accept unsollicited notify so that on the BR2 side, you use sip-notify as DTMF relay on both CUE and UCM SIP dialpeers. Let us know if thay might help cheers, Juan 2012/3/4 Juan Lopez lopez.hernandez.j...@gmail.com Hi Justin, from reading the mail it looks like on the SIP dialpeers on the BR2, you use the rtp-nte (inband) dtmf-relay method? can you try and let us know: 1) use SIP-NOTIFY on both SIP dialpeers at BR2? (not sure if UCM supports this - in SRND it states a UCM SIP trunk uses RTP-NTE or possibly SIP-KPML) if 1 does not work: 2) use sip notify on CUE dialpeer and sip-kpml on sip dialpeer to UCM. Not sure here whether the CUBE at branch 2 supports notify - kpml dtmf interworking... The idea is to have DTMF between UCM and CME out of band... From SRND I read that SIP at UCM uses RTP-NTE or possibly SIP-KPML, so it rules out to use the SIP-NOTIFY on the dialpeer at branch2 pointing to UCM (not tested yet) to keep it all out of band - but this is the way to rule out an MTP 2012/3/3 Justin McIntyre justin.mcint...@blackbox.com Ok. For those who are interested I have resolved my issue. By selecting the Media Termination Point Required option within the SIP trunk I was able to resolve my media stream to an MTP prior to connection to the CME. This allowed in-band/Out of Band DTMF traversal. Note that when you select the MTP required option within your sip trunk to pay special attention to the device pool and region settings upon with the MTP that you will resolve to will lie. The MTP will not inherit the Device Pool settings from the Sip trunk depending on your configuration. This was a really good learning experience and if anyone is curious as to any further details please let me know. I am however un-clear on one thing and maybe someone can help me out. I remember using Sip-Notify within my CUE dial-peer and within CUE configuration the last time I ran this lab. For some reason I could not get SIP-Notify to work in any case at all that I tried this time around. If anyone has any clarity on this I would be most appreciative, I'd hate to see, please configure a sip trunk between UCM and CME location at to reach the CUE VM pilot. Note: use of an MTP on the SIP trunk is not allowed in the lab. Plus who knows when a customer site may encounter this situation. Thanks everyone. *!*!*!*Thanks to Chase and Vik as they were pertinent in my resolution.*!*!*!* 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 recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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] MOH issue
Locations cac allows for 80k to BR1 ? On 29 Feb 2012, at 13:02, Brian Valentine bkvalent...@gmail.com wrote: Sorry... read too fast. I see it there on the subinterfaces now. Are you testing via set to set or via pstn? Pstn should need a loopback ro work. On Feb 29, 2012 6:59 AM, Brian Valentine bkvalent...@gmail.com wrote: Yes. You arent making pim neighbor relationships over the wan link. Enable pim on your wan interfaces. On Feb 29, 2012 6:44 AM, Ashwani ash_r...@hotmail.com wrote: Juan, Yes igmp-snooping is disabled on switches. I forgot to paste my switch configuration. Also I have Trans-coder setup for CorpHQ and BR1 site. Also CUCM MoH is setup for G711 only and Region for CorpHQ and BR1 is allowed G729. Thanks for looking into this issue. Ashwani On 2/29/2012 12:14 AM, Juan Lopez wrote: Ashwani, have you turned off 'ip igmp snooping' on the switch? The silence indicates a network issue, not a ucm/resource issue. Also note, no mmoh over IPSec tunnels. On 29 Feb 2012, at 05:27, Ashwani ash_r...@hotmail.com wrote: Hello Experts, I am trying to do MoH lab but I am not seeing the result as expected. Here is my scenario, relevant configuration and my test results MOH Audio Stream Number MOH Audio Source File Play Continuously Allow Multicasting 1 Source 1 Check Not Check 2 Source 2 Check Check 3 Source 3 Check Not Check MOH Server Name Device Pool Enable Multicast Selected Multicast Audio Sources Max Hops MOH_PUB DP_CorpHQ Check Source 2 5 MOH_SUB DP_Branch1 Check Source 2 5 MRG Name Selected Media Resources Use Multicast for MOH Audio MRG_Pub-SW-MMoH MOH_PUB (Multicast) Check MRG_Pub-SW-UMoH MOH_PUB (Multicast) Not Check MRG_Sub-SW-MMoH MOH_SUB (Multicast) Check MRG_Sub-SW-UMoH MOH_SUB (Multicast) Not Check MRGL Name Selected MRG’s MRGL_CorpHQ-Devices MRG_Pub-SW-MMoH MRG_Sub-SW-MMoH MRGL_Branch1-Devices MRG_Pub-SW-UMoH MRG_Sub-SW-UMoH Phone Device Pool MRGL User Hold Audio Source Network Hold Audio Source CorpHQ PH1 1001 DP_CorpHQ MRGL_CorpHQ-Devices Source 2 Source 1 BR1 PH1 2001 DP_Branch1 MRGL_Branch1-Devices Source 3 Source 1 Gateways Device Pool CorpHQ_GW DP_CorpHQ Branch1_GW DP_Branch1 Test Results:- Holder Holdee User Hold MoH Played Network MoH Played RTMT Result CorpHQ PH1 BR1 PH1 Yes (Source 2) Yes (Source 1) User Hold (Unicast), Network Hold (Unicast) BR1 PH1 CorpHQ PH1 Yes (Source 3) No (Silence) User Hold (Unicast), Network Hold (Multicast) CorpHQ PH1 CorpHQ PH2 Yes (Source 2) No (Silence) User Hold (Multicast), Network Hold (Multicast) BR1 PH1 BR1 PH2 Yes (Source 3) Yes (Source 1) User Hold (Unicast), Network Hold (Unicast) Here is my CorpHQ Router Configuration…. Ip multicast-routing ! interface FastEthernet0/0 description == To SW1 no ip address duplex auto speed auto ! interface FastEthernet0/0.10 description == Server VLAN encapsulation dot1Q 10 ip address 10.10.100.1 255.255.255.0 ip pim dense-mode ! interface FastEthernet0/0.11 description == Voice VLAN encapsulation dot1Q 11 ip address 10.10.200.3 255.255.255.0 ip pim dense-mode ! interface FastEthernet0/0.12 description == Data VLAN encapsulation dot1Q 12 ip address 10.10.210.1 255.255.255.0 ! interface Serial0/1/0 description == Frame-Relay Circuit to WAN no ip address encapsulation frame-relay no fair-queue service-module t1 timeslots 1-24 cdp enable no frame-relay inverse-arp frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point description == FR To BR1 ip address 10.10.111.1 255.255.255.0 ip pim dense-mode snmp trap link-status frame-relay interface-dlci 101 ! interface Serial0/1/0.2 point-to-point description == FR To BR2 ip address 10.10.112.1 255.255.255.0 ip pim dense-mode snmp trap link-status frame-relay interface-dlci 201 ! Here is my Branch1 Router Configuration…. Ip multicast-routing ! interface FastEthernet0/0.20 description == Data VLAN encapsulation dot1Q 22 ip address 10.10.101.1 255.255.255.0 ! interface FastEthernet0/0.21 description == Voice VLAN encapsulation dot1Q 21 ip address 10.10.201.1 255.255.255.0 ip pim dense-mode ! interface Serial0/1/0 description == Frame-Relay Circuit to WAN no ip address encapsulation frame-relay no fair-queue service-module t1 timeslots 1-24 cdp enable no frame-relay inverse-arp frame-relay lmi-type ansi ! interface
Re: [OSL | CCIE_Voice] MOH issue
Also: ccm-manager music-on-hold configured on br1? On 29 Feb 2012, at 13:02, Brian Valentine bkvalent...@gmail.com wrote: Sorry... read too fast. I see it there on the subinterfaces now. Are you testing via set to set or via pstn? Pstn should need a loopback ro work. On Feb 29, 2012 6:59 AM, Brian Valentine bkvalent...@gmail.com wrote: Yes. You arent making pim neighbor relationships over the wan link. Enable pim on your wan interfaces. On Feb 29, 2012 6:44 AM, Ashwani ash_r...@hotmail.com wrote: Juan, Yes igmp-snooping is disabled on switches. I forgot to paste my switch configuration. Also I have Trans-coder setup for CorpHQ and BR1 site. Also CUCM MoH is setup for G711 only and Region for CorpHQ and BR1 is allowed G729. Thanks for looking into this issue. Ashwani On 2/29/2012 12:14 AM, Juan Lopez wrote: Ashwani, have you turned off 'ip igmp snooping' on the switch? The silence indicates a network issue, not a ucm/resource issue. Also note, no mmoh over IPSec tunnels. On 29 Feb 2012, at 05:27, Ashwani ash_r...@hotmail.com wrote: Hello Experts, I am trying to do MoH lab but I am not seeing the result as expected. Here is my scenario, relevant configuration and my test results MOH Audio Stream Number MOH Audio Source File Play Continuously Allow Multicasting 1 Source 1 Check Not Check 2 Source 2 Check Check 3 Source 3 Check Not Check MOH Server Name Device Pool Enable Multicast Selected Multicast Audio Sources Max Hops MOH_PUB DP_CorpHQ Check Source 2 5 MOH_SUB DP_Branch1 Check Source 2 5 MRG Name Selected Media Resources Use Multicast for MOH Audio MRG_Pub-SW-MMoH MOH_PUB (Multicast) Check MRG_Pub-SW-UMoH MOH_PUB (Multicast) Not Check MRG_Sub-SW-MMoH MOH_SUB (Multicast) Check MRG_Sub-SW-UMoH MOH_SUB (Multicast) Not Check MRGL Name Selected MRG’s MRGL_CorpHQ-Devices MRG_Pub-SW-MMoH MRG_Sub-SW-MMoH MRGL_Branch1-Devices MRG_Pub-SW-UMoH MRG_Sub-SW-UMoH Phone Device Pool MRGL User Hold Audio Source Network Hold Audio Source CorpHQ PH1 1001 DP_CorpHQ MRGL_CorpHQ-Devices Source 2 Source 1 BR1 PH1 2001 DP_Branch1 MRGL_Branch1-Devices Source 3 Source 1 Gateways Device Pool CorpHQ_GW DP_CorpHQ Branch1_GW DP_Branch1 Test Results:- Holder Holdee User Hold MoH Played Network MoH Played RTMT Result CorpHQ PH1 BR1 PH1 Yes (Source 2) Yes (Source 1) User Hold (Unicast), Network Hold (Unicast) BR1 PH1 CorpHQ PH1 Yes (Source 3) No (Silence) User Hold (Unicast), Network Hold (Multicast) CorpHQ PH1 CorpHQ PH2 Yes (Source 2) No (Silence) User Hold (Multicast), Network Hold (Multicast) BR1 PH1 BR1 PH2 Yes (Source 3) Yes (Source 1) User Hold (Unicast), Network Hold (Unicast) Here is my CorpHQ Router Configuration…. Ip multicast-routing ! interface FastEthernet0/0 description == To SW1 no ip address duplex auto speed auto ! interface FastEthernet0/0.10 description == Server VLAN encapsulation dot1Q 10 ip address 10.10.100.1 255.255.255.0 ip pim dense-mode ! interface FastEthernet0/0.11 description == Voice VLAN encapsulation dot1Q 11 ip address 10.10.200.3 255.255.255.0 ip pim dense-mode ! interface FastEthernet0/0.12 description == Data VLAN encapsulation dot1Q 12 ip address 10.10.210.1 255.255.255.0 ! interface Serial0/1/0 description == Frame-Relay Circuit to WAN no ip address encapsulation frame-relay no fair-queue service-module t1 timeslots 1-24 cdp enable no frame-relay inverse-arp frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point description == FR To BR1 ip address 10.10.111.1 255.255.255.0 ip pim dense-mode snmp trap link-status frame-relay interface-dlci 101 ! interface Serial0/1/0.2 point-to-point description == FR To BR2 ip address 10.10.112.1 255.255.255.0 ip pim dense-mode snmp trap link-status frame-relay interface-dlci 201 ! Here is my Branch1 Router Configuration…. Ip multicast-routing ! interface FastEthernet0/0.20 description == Data VLAN encapsulation dot1Q 22 ip address 10.10.101.1 255.255.255.0 ! interface FastEthernet0/0.21 description == Voice VLAN encapsulation dot1Q 21 ip address 10.10.201.1 255.255.255.0 ip pim dense-mode ! interface Serial0/1/0 description == Frame-Relay Circuit to WAN no ip address encapsulation frame-relay no fair-queue service-module t1 timeslots 1-24 cdp enable no frame-relay inverse-arp frame-relay lmi-type ansi
Re: [OSL | CCIE_Voice] 150 ms latency
also, the 64k is one-way. But witin QOS this is ok to take only this into consideration : you do the BW provisioning on the outbound direction to the provider. Inbound shaping/policing/priority queueing is done at the other side of the link. cheers! 2012/2/27 donny f f.faraday...@gmail.com hi all, are the 150 ms latency and 64 kpbs for one way or 2 way ? tks 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 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] MVA Calling Name
datucha, from my notes I can see the calling name and number from RDP were shown for any calls made through MVA - internal and external... I don't have that lab set up anymore at the moment, so cannot verify for 100% can someone confirm? 2012/2/16 datucha123 datucha123 datucha...@gmail.com Hello, When PSTN Phone makes call to MVA number (IVR) and then makes a call to HQ or any other site internal extensions, the Calling Name of the RDP Profile is not visible on the called Phone. Is it normal? or I have misconfigured something? ___ 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] built-in bridge support for iLBC g729
can anyone confirm the phone built-in bridge does support g729? I'm running it in my lab, no HW CFB are configured and I put all SW CFB in their own MRG, to be sure they are not used: admin:show perf query class Cisco SW Conference Bridge Device ==query class : - Perf class (Cisco SW Conference Bridge Device) has instances and values: CFB_2 - AllocatedResourceCannotOpenPort = 0 CFB_2 - OutOfResources = 0 CFB_2 - ResourceActive = 0 CFB_2 - ResourceAvailable = 48 CFB_2 - ResourceTotal = 48 CFB_2 - SWConferenceActive = 0 CFB_2 - SWConferenceCompleted = 0 CFB_3 - AllocatedResourceCannotOpenPort = 0 CFB_3 - OutOfResources = 0 CFB_3 - ResourceActive = 0 CFB_3 - ResourceAvailable = 48 CFB_3 - ResourceTotal = 48 CFB_3 - SWConferenceActive = 0 CFB_3 - SWConferenceCompleted = 0 admin:show perf query class Cisco HW Conference Bridge Device ==query class : - Perf class (Cisco HW Conference Bridge Device) has instances and values: no values are returned Nevertheless, the barge function is working: barger and bargee use g722, the 3rd party uses g729. same goes is 3rd party would use iLBC. 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
Re: [OSL | CCIE_Voice] MVA problem
Datucha, for what its worth, i'm currently set stuck a bit unfortunately with practicing labs : check the location settings - I've seen something when i was last practicing which I did not understand:when using the xfer to mobile softkey combined with locations CAC. It's a wild guess, but a small chance this might be helpful with respect to the toh: as long as the xfer is not completed, I remember counting 80k extra on rtmt, no matter the RDP's DP. If not useful, pls disregard this feedback On 08 Feb 2012, at 20:28, datucha123 datucha123 datucha...@gmail.com wrote: I have the following kind of probem: I am using SLRG for Mobile Connect calls, so that that calls to users mobiles are done through local gateway (this is just for test). Now, the HQ phone has the RDP assinged with RD of his mobile phone. Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. Also I have noticed that when the Mobile Phone picks up the call faster then the Desk Phone, the codec negotiated is g711 from BR1 phone to its local gateway through which the call went out. But if the Desk Phone at HQ picked up the call first, and then made Send calls to mobile phone, the codec stays at G729 on BR1 phone, even though the call is going out throuhg local BR1 gateway where it should use G711. ___ 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] Calling Party Transformation set at the egress GW
thanks - I was glad to read about what I overlooked at first glance... But the explanation is not correct I'm afraid: Calls from UCME drop of at the HQ gw (teho). The calling number sent out by the IOS for this teho call is also 10digit 3432143002 (like a teho call from BR1 going out over the HQ gw (ANI = 6178631002)), but in this case it is shown as 'international' like I set it up to be on the UCM - so here it is not overwritten bu the IOS... 2012/2/1 datucha123 datucha123 datucha...@gmail.com That is because of IOS. IOS detects the US Dialplan, and sets the Types accordingly in H323 gateway. It is not possible to disable that feature. So you have to use voice translation rules to change the ANI Type. On Wed, Feb 1, 2012 at 2:01 PM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: Hi all, I'm doing lab 5 Vol1 with some extra things into it and found the following: Has anyone noticed that when using a H323 GW in a backup fashion (example: RG-BR1 contains BR1-GW as primary and HQ-GW as secondary), the calling number type is ALWAYS set to national, unless you use a prefix in the calling party transformation? For most cases this is ok, but not for all (example: international TEHO for calls from the UCME) any feedback is welcome ! grts, 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 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] MVA probably stupid Q...
Hi all, when setting up MVA, it provides the menu to press 2 or 3, but for some reason, it does not present the option 1 to make an outbound call through UCM. I'm sure I overlooked somethigm but can't find it ... someone willing to throw in some 'nice to have done' ;) to have this option being presented? Greatly appreciated ! Some details if this matters: The RDP etc is being matched (do only have to enter PIN), the RDP's CSS is set to be able to dial anything. On the H323 I translate everythig to 4 digit on the voice port (5XXX): application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 5999 pots service cmm incoming called-number 5999 no digit-strip dial-peer voice 1000 voip destination-pattern 5...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.11 incoming called-number . dtmf-relay h245-alphanumeric no vad ___ 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] Publisher failure
am afraid you will need to do disaster recovery on pub (rebuild and import backup) , then replicate it's DB to the sub afterwards. 2012/1/19 khaled Saholy khaled_sah...@hotmail.com Hi, What can we do in case of the Publisher failed to boot and the Subscriber is still working but I can't do any configuration on it? The Publisher vm crashed and I could not recover it. Is there a way to restore the db from the sub to the pub or the replication will fix that? Thanks for your inputs. Regards. Khaled ___ 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] Voicemail during AAR Redux
would 'call-forward busy 570.' work under fallback config? This needs also that the 570X range is part of the HQ's DID range. If this is the case, this inbound call on HQ can be matched on a CTI RP witf CFA to VM and with a VM mask set. from the top of my head I do not see how to manipulate the RDNIS so it dynamically reflects the forwarding station at the branch. If it only needs to work for 1 specific forwarding number, then you can manipulate it statically I guess 2012/1/19 Anthony Alba ascanio.al...@gmail.com Hello, this issue has surfaced in the past but no one email seems to summarize the exact requirements to get Voicemail to work during AAR. I'd like to give a go and get your feedback: Task: BR1, a H.323 GW, is in AAR, Voicemail must work 1. BR1 Ph2 dials Voicemail external PSTN DID directly: Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002 Solution: Configure 415888100N as alternate extension for all BR1 lines 100N 2. BR1 Ph2 presses messages service button or dials 5600 (the VM pilot) Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002, RDNIS 5600 Solution: This is the task that seems to cause the most confusion, you hit the Unity Connection Opening Greeting rather than the users Attempt Sign-In; this is due to the RDNIS 5600 which isn't a mailbox on the system. Unlike some reports which stated that the 10D CallerID as alternate extension worked for them. I found that the RDNIS matching wins, it is a non-mailbox extension, so I always get Unity Connection Opening Greeting. Can you guys confirm that this is the expected behaviour for RDNIS = 5600 (VM pilot) and CallerID = 4158881002 (1002 alternate extension). My solution is to add a Fowarded Routing Rule with Forwarding Station = 5600 and Send Call To = Attempt Sign-In I have only read one report that suggested this and I find I need this; yet nobody else seemed to need this. Hence I really like to hear your thoughts: is the Forwarded Routing Rule mandatory? 3. PSTN, Internal users call BR1 Ph2 Note: HQ-RTR sees CalledID 4087775600, CallerID 123456789, RDNIS 1002 Solution: this task works with no further configuration because the RDNIS is already 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] Voicemail during AAR Redux
my mistake - I didn't read the question well - apologies 2012/1/19 George Goglidze gogli...@gmail.com Juan, We are discussing AAR not fallback mode... that's completely different. On Thu, Jan 19, 2012 at 9:07 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: would 'call-forward busy 570.' work under fallback config? This needs also that the 570X range is part of the HQ's DID range. If this is the case, this inbound call on HQ can be matched on a CTI RP witf CFA to VM and with a VM mask set. from the top of my head I do not see how to manipulate the RDNIS so it dynamically reflects the forwarding station at the branch. If it only needs to work for 1 specific forwarding number, then you can manipulate it statically I guess 2012/1/19 Anthony Alba ascanio.al...@gmail.com Hello, this issue has surfaced in the past but no one email seems to summarize the exact requirements to get Voicemail to work during AAR. I'd like to give a go and get your feedback: Task: BR1, a H.323 GW, is in AAR, Voicemail must work 1. BR1 Ph2 dials Voicemail external PSTN DID directly: Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002 Solution: Configure 415888100N as alternate extension for all BR1 lines 100N 2. BR1 Ph2 presses messages service button or dials 5600 (the VM pilot) Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002, RDNIS 5600 Solution: This is the task that seems to cause the most confusion, you hit the Unity Connection Opening Greeting rather than the users Attempt Sign-In; this is due to the RDNIS 5600 which isn't a mailbox on the system. Unlike some reports which stated that the 10D CallerID as alternate extension worked for them. I found that the RDNIS matching wins, it is a non-mailbox extension, so I always get Unity Connection Opening Greeting. Can you guys confirm that this is the expected behaviour for RDNIS = 5600 (VM pilot) and CallerID = 4158881002 (1002 alternate extension). My solution is to add a Fowarded Routing Rule with Forwarding Station = 5600 and Send Call To = Attempt Sign-In I have only read one report that suggested this and I find I need this; yet nobody else seemed to need this. Hence I really like to hear your thoughts: is the Forwarded Routing Rule mandatory? 3. PSTN, Internal users call BR1 Ph2 Note: HQ-RTR sees CalledID 4087775600, CallerID 123456789, RDNIS 1002 Solution: this task works with no further configuration because the RDNIS is already 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 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] CUCM version to study upgrade to latest 7.0.x release?
all, I found the version on proctorlabs (7.0.1.11002-2) is giving me quite some issues with dialrules on the 7962. Is it a good thing to upgrade to the latest 7.0.x release to study, without being out of sync with the tested UCM version? Or should I simply upgrade the phone firmwares instead? what is the best way to prepare for the real exam? thx, 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] 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
[OSL | CCIE_Voice] lab 4a shouldn't take too much time, right?
someone of you guys/girls walked into following weird behaviour ? I for sure wasted too much time with lab 4a, 2 remote rack days :(( , where a SIP HQ phone (5002) is supposed to dial 3...@ipxcme.com (SIP CME phone) The issue was when picking up the call ath the CME phone, it disconnected. So a codec issue. Per lab 4a, the call at HQ phone leaves the trunk, both are in HQ device pool. Inbound dialpeer at CME has voice-class codec with g711 and g729. Registered SIP CME phone has g711 configured on voice register pool. So call should negotiate to g711: supported by the region configuration at UCM, on the inbound call leg and the final CME phone. But it fails. Putting g729 on voice register pool call succeeds as g729... So in g729 it works Suspicion: is the correct inbound dialpeer matched? YES, the one with the voice-class codec. So that cannot be the cause why the call MUST be g729. What is? debug ccsip mess is not very descriptive: the CME router sends BYE to the UCM, with *some* cause code, when the CME phone is configured to have g711. Not familiar zith UCM traces unfortunately (hopefully this will improve, quickly...) After a *very long* journey in the dark, I reset the UCM SIP phone. Problem automagically gone :-( Question: - does this happen often with sip phones? - what trace should I have looked into on the UCM, to find more UCM info that the phone ONLY tried to negotiate the g729? Can someone point me in a good direction what trace to enable, and what text files to investigate? Is there some good info/examples available on this topic :S ? 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
Re: [OSL | CCIE_Voice] Unity 4.0(5) - VM Selection
is it possible to add a CTI RP into a line group? thought that didn't work, but can be wrong of course cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Jost (grjost) Sent: Friday, May 16, 2008 6:52 PM To: Christian Narvaez; Mark Snow Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity 4.0(5) - VM Selection Use AC PP/Hunt Group containing CTI RP w/CFA to VM and change the following TCD service parameter: Keep Original Called Party If Forwarded : This parameter determines whether the directory number (DN) of the original called party gets reset to the redirected number if the call is forwarded to the Pilot Point. Valid values specify True (do not reset the original called party) or False (reset the original called party during forwarding). This is a required field. Default: true. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Narvaez Sent: Thursday, May 15, 2008 8:26 AM To: Mark Snow Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity 4.0(5) - VM Selection Mark, I see your lets say Super-Hint is valid, but you mentioned that there are multiple ways to achieve the same. I would like to find one where just CCM and UNITY be involved, avoiding to use IPCC Scripting because is too time consuming to be applied in the Exam, besides it would be dependant of the IPCC be working correctly (I have the impression most of the people dont make work the IPCC in their first attempts) thankful of your wise guidance. -Original Message- From: Mark Snow [mailto:[EMAIL PROTECTED] Sent: Wed 5/14/2008 4:29 PM To: Christian Narvaez Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity 4.0(5) - VM Selection Foward the call to 7 where is the other VM Box you want it to hit (for this example maybe 2001). 7 (literally just 7) is the trigger for a UCCX application that preserves whatever '' was in a variable (in this case 2001). Then a new number is formed to make it like # + -variable so the new number is #2001. Then forward #2001 back to the UCM where it matches a CTI RP that is # and FwdsAll to VM with a New VMProfile that has only the mask (so it strips the #) - then it forwards properly into the correct Subscriber box. Sorry - got to typing and didn't really give a hint - gave the answer - sometimes I get carried away in thinking up a way to solve it that I forgo the fishing lesson for the fish :) Ah well - next time perhaps. -- Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On May 14, 2008, at 4:22 PM, Christian Narvaez wrote: I agree with you that stuff about fishing :). Actually now I am stuck trying to get around how to manipulate the Redirect Number in Callmanager, never done that before (What I have seen is just manipulations of called and calling number). And trying to figure out what kind of variables you meant for that. H, maybe another hint would not be so bad to learn fish a barracuda like this :) -Original Message- From: Mark Snow [mailto:[EMAIL PROTECTED] Sent: Wed 5/14/2008 4:03 PM To: Christian Narvaez Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity 4.0(5) - VM Selection If that wasn't enough of a hint - let me know - I will be happy to give you a bit more of a hint of HOW to do it in UCCX - but I am one of those that is of the mindset of teaching people to fish - if you get the euphemism :) -- Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On May 14, 2008, at 3:48 PM, Christian Narvaez wrote: Thanks Mark for the clarification !, any little clue how to resolve it would be nice :) . I take the rest. -Original Message- From: [EMAIL PROTECTED] on behalf of Mark Snow Sent: Wed 5/14/2008 2:24 PM To: Christian Narvaez
Re: [OSL | CCIE_Voice] 3550 trust L2 and police
So is it correct to say that the input policy map on the port to which an IP phone is connected, and with conditional trust, the COS will be mapped to internal DSCP, which then is used to classify the traffic according to the class-maps of the policy map? cheers, Juan From: Devildoc [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 22, 2008 3:56 PM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); Gregory Jost (grjost); CCIE Maillist Subject: RE: [OSL | CCIE_Voice] 3550 trust L2 and police When conditional trust is configured as in mls qos trust device cisco-phone, the switchport will trust the COS value from the IP Phone device only when the switchport detects (via CDP) that an IP phone is connected to its port. If no IP phone is dectected, then the switchport becomes untrusted. The COS value will be used to schedule for INGRESS queue treatment. The COS value will then be mapped to an internal DSCP value (according to the configured COS-DSCP map) for EGRESS queue treatment. JD Subject: RE: [OSL | CCIE_Voice] 3550 trust L2 and police Date: Mon, 21 Apr 2008 22:51:00 +0200 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com if you have a conditional trust, what is trusted when the switch is hooked up to an IP phone? Is it the COS or DSCP? Either way, it will be converted to internal DSCP which then is used in the class maps of the policy map attached to the interface. Correct? Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devildoc Sent: Monday, April 21, 2008 5:19 PM To: Gregory Jost (grjost); CCIE Maillist Subject: Re: [OSL | CCIE_Voice] 3550 trust L2 and police You cannot trust COS and police it at the same time on the 3550. As you have found out, when you applied the policer policy, the trust COS statement is replaced with the trust device cisco-phone. As for your 2nd question, the configuration looks valid. You don't need to set ip dscp 24 in the policy-map again since the packets have already been matched with the statement match ip dscp 24 in the class-map. Setting ip dscp 24 again is just redundant. JD Date: Sat, 19 Apr 2008 11:22:01 -0700 From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 3550 trust L2 and police Is it even possible to trust L2 markings AND police on the 3550? A policy-map replaces mls qos trust cos, but not mls qos trust device cisco-phone. Is the following configuration valid, or do you need set ip dscp 24 in the policy-map (not trusting phone cos-dscp mapping)? 3550: mls qos map policed-dscp 24 to 0 mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos ! class-map match-all Control match ip dscp 24 ! policy-map Policer class Control police 32000 12000 exceed-action policed-dscp-transmit ! interface FastEthernet0/2 mls qos trust device cisco-phone service-policy input Policer Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 Pack up or back up-use SkyDrive to transfer files or keep extra copies. Learn how. http://www.windowslive.com/skydrive/overview.html?ocid=TXT_TAGLM_WL_Ref resh_skydrive_packup_042008 Express yourself wherever you are. Mobilize! http://www.gowindowslive.com/Mobile/Landing/Messenger/Default.aspx?Loca le=en-US?ocid=TAG_APRIL
Re: [OSL | CCIE_Voice] Question
yep, indeed... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Jost (grjost) Sent: Saturday, April 19, 2008 9:24 PM To: CCIE Maillist Subject: [OSL | CCIE_Voice] Question Would anyone out there besides me rather take a claw hammer to the face than deal with L2 QoS? Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] 3550 trust L2 and police
if you have a conditional trust, what is trusted when the switch is hooked up to an IP phone? Is it the COS or DSCP? Either way, it will be converted to internal DSCP which then is used in the class maps of the policy map attached to the interface. Correct? Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devildoc Sent: Monday, April 21, 2008 5:19 PM To: Gregory Jost (grjost); CCIE Maillist Subject: Re: [OSL | CCIE_Voice] 3550 trust L2 and police You cannot trust COS and police it at the same time on the 3550. As you have found out, when you applied the policer policy, the trust COS statement is replaced with the trust device cisco-phone. As for your 2nd question, the configuration looks valid. You don't need to set ip dscp 24 in the policy-map again since the packets have already been matched with the statement match ip dscp 24 in the class-map. Setting ip dscp 24 again is just redundant. JD Date: Sat, 19 Apr 2008 11:22:01 -0700 From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 3550 trust L2 and police Is it even possible to trust L2 markings AND police on the 3550? A policy-map replaces mls qos trust cos, but not mls qos trust device cisco-phone. Is the following configuration valid, or do you need set ip dscp 24 in the policy-map (not trusting phone cos-dscp mapping)? 3550: mls qos map policed-dscp 24 to 0 mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos ! class-map match-all Control match ip dscp 24 ! policy-map Policer class Control police 32000 12000 exceed-action policed-dscp-transmit ! interface FastEthernet0/2 mls qos trust device cisco-phone service-policy input Policer Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 Pack up or back up-use SkyDrive to transfer files or keep extra copies. Learn how. http://www.windowslive.com/skydrive/overview.html?ocid=TXT_TAGLM_WL_Ref resh_skydrive_packup_042008
Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail...
I'm missing the point why we need the AAR CSS and AAR group on the remote GW for redirected calls? Cheers, Juan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Wednesday, April 16, 2008 10:43 PM To: [EMAIL PROTECTED] Cc: CCIE Voice Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... That makes sense... Now, what about when a GK denies the call to insufficient bandwidth? The only thing I can think of is to put an H.323 gateway second the route-list for the GK-controlled trunk (from CCM) and turn off the stop hunting on unallocated/busy/etc... Jonathan On Wed, Apr 16, 2008 at 3:12 PM, Vik Malhi [EMAIL PROTECTED] wrote: AAR cannot be down and AAR can't deny a call. Your entire CallManager cluster is either unavailable (SRST is the solution) or it is available (AAR being the solution when the WAN is saturated). If CallManager is active and alive then AAR cannot fail unless off course it is misconfigured or you are out of B-channels on your PSTN connection. Assuming the CallManager is operational and the remote sites phones and gateway are still registered then AAR is the solution for when Locations CAC blocks the call. So on the remote phon you hit the Messages button and CallManager determines there is no Locations bandwidth available. The External Number Mask and AAR Group needs to be configured on the Hunt Pilot for Voicemail. The AAR CSS and AAR Group needs to be configured on the calling (remote) phone. Also in the case of a Call Forward from the remote phone, the remote gateway needs an AAR CSS and AAR Group (+Redirecting # outbound). Assuming the CallManager is not available (WAN outage) then SRST is your only option. Nothing on CallManager works including AAR. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 16, 2008 1:01 PM To: [EMAIL PROTECTED] Cc: Onur Tufekci; CCIE Voice Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... what I meant was SRST config to reroute the call if AAR is down... So, if I am at the remote site, AAR is denying calls and I hit the messages button, what are my options? Jonathan On Wed, Apr 16, 2008 at 2:57 PM, Vik Malhi [EMAIL PROTECTED] wrote: You can configure both- but only one of AAR and SRST will be active at any one point. When the WAN is saturated signaling still traverses the WAN. When there is a WAN outage then you lose signaling to the remote sites and SRST kicks in. From your original question you indicate that SRST might be a solution for AAR which it isn't. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED]Sent: Wednesday, April 16, 2008 12:41 PMTo: [EMAIL PROTECTED]Cc: Onur Tufekci; CCIE VoiceSubject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... Well the goal is two-fold. First, if the WAN is saturated, someone at HQ, should be able to callBR2 via the PSTN (using AAR), and a CFNA/CFB should route back to vmail and the correct box In the event of a WAN outage, the phones in SRST should also CFNA/CFB to voicemail and the correct boxJonathan On Wed, Apr 16, 2008 at 2:23 PM, Vik Malhi [EMAIL PROTECTED] wrote: Good point- AAR requires you to configure an AAR CSS, AAR Group on the remote site gateway in addition to checking the Redirecting Number Outbound checkbox (if this is MGCP then no mgcp/mgcp on the IOS). You need to check the Redirecting Number Inbound checkbox on the HQ gateway. You also need an external number mask and aar group on the hunt pilot. It should work a treat having done this. The Redirecting Number is good since CCM builds this in the case of AAR.
Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail...
Thx Jonthan. According to Vik's reply earlier on this, it looks like AAR/UCM builds the RDNIS correctly, and passes it on back to Unity. The *only* thing is of course that the ISDN cloud, or Telco cloud in general needs to pass this info too. If the telco cloud doesn't relay this RDNIS (example: no ISDN), I'm not aware of another solution for this... For SRST there's the VM-integration stuff, but that doesn't work in this case. But I believe that was also what you were referring to right Jonathan? -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Thursday, April 17, 2008 4:06 PM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Cc: Gregory Jost (grjost); CCIE Voice Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... Well, what is actually happening? Call comes in on BR1 router, tries to ring a phone (on the CCM cluster) at BR1, RNA, gets forwarded to VM... OK. Who is calling VM? the Phone at BR1 that CFNA to VM? Or the gateway? From an IP perspective, the call leg is terminated and initiated by the GW, not the phone at BR1... So, we will then have RDNIS back out the GW to HQ and to Unity... My question is, will this RDNIS get passed to Unity when the call is redirected and will it go to the right mailbox. From Unity's perspective, the original called party is the Unity pilot, as the GW is initiating that call leg... How do we get that call into the right mailbox Jonathan On Thu, Apr 17, 2008 at 8:53 AM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) [EMAIL PROTECTED] wrote: Ah ! Thanks Greg and Jonathan. I didn't consider the case when a PSTN caller called BR1 phone, forcing a hairpin on BR1 GW in case no BW left between the BR1 GW and the HQ's VM. I suppose when BR1 phone redirects to Unity, signaling (ECS) sets up a new call between calling (BR1 GW) party and called party (Unity) right? Just to make sure it's correct what I say above: 'no BW left between the BR1 GW and the HQ's VM' - or: CAC between calling (BR1 GW) and called (Unity pilot) is considered, not CAC BR1phone - Unity pilot Cheers, Juan -Original Message- From: Gregory Jost (grjost) Sent: Thursday, April 17, 2008 3:18 PM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); Jonathan Charles; [EMAIL PROTECTED] Cc: CCIE Voice Subject: RE: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... The gateway is calling device (on the VoIP network). In the case of a redirected call, the called party becomes the redirecting party, so its AAR Group/CSS is not used. When Location bandwidth is exhausted, the calling device needs to know the prefix (AAR Group) and appropriate gateway (AAR CSS) in order to re-route the call over PSTN. In this case, the gateway will hairpin back to PSTN. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Sent: Thursday, April 17, 2008 6:51 AM To: Jonathan Charles; [EMAIL PROTECTED] Cc: CCIE Voice Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... I'm missing the point why we need the AAR CSS and AAR group on the remote GW for redirected calls? Cheers, Juan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Wednesday, April 16, 2008 10:43 PM To: [EMAIL PROTECTED] Cc: CCIE Voice Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail... That makes sense... Now, what about when a GK denies the call to insufficient bandwidth? The only thing I can think of is to put an H.323 gateway second the route-list for the GK-controlled trunk (from CCM) and turn off the stop hunting on unallocated/busy/etc... Jonathan On Wed, Apr 16, 2008 at 3:12 PM, Vik Malhi [EMAIL PROTECTED] wrote: AAR cannot be down and AAR can't deny a call. Your entire CallManager cluster is either unavailable (SRST is the solution) or it is available (AAR being the solution when the WAN is saturated). If CallManager is active and alive then AAR cannot fail unless off course it is misconfigured or you are out of B-channels on your PSTN connection. Assuming the CallManager is operational and the remote sites phones and gateway are still registered then AAR is the solution for when Locations CAC blocks the call. So on the remote phon you hit the Messages button and CallManager determines there is no Locations bandwidth available. The External Number Mask and AAR Group needs to be configured on the Hunt Pilot for Voicemail. The AAR CSS and AAR Group needs to be configured on the calling (remote) phone. Also in the case of a Call Forward
Re: [OSL | CCIE_Voice] QoS marking based on port
Hi Greg, I've been putting this on the agenda too - and was planning to use ethereal to grab all port numbers sent out by the UCM/Unity... - I believe it's only there that you need to classify based on the port numbers. Maybe yes, your idea about netstat may even be a better alternative one... Another thing relating to this, is the CCME, which terminates RTP and recreates it (example: call CCME phone/VTA to UCM phone). It looks like the CCME is setting up this new RTP stream with the TOS you set on the voip dialpeer towards UCM/IPIP, so there you loose the AF41 classification in case you have a video call. cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Jost (grjost) Sent: Thursday, April 17, 2008 4:07 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] QoS marking based on port There's a shroud of mystery around protocol port mappings. It's documented one way, taught another, but no one really knows what the proctor is looking for. To me, the definitive answer would be to look at the open ports on servers (netstat -a from CMD), and the open ports on the routers (sh ip sockets). This will show the exact ports being used by the active devices, including src/dst and udp/tcp (IP 17 and 6 respectively); however, this may not be what the proctor is looking for. For example, if you're using UDP for SIP, there will not be an open TCP port. If the proctor sees that you've only included udp 5060 for SIP, he may deduct points. For something like this, there should be a right way; otherwise, we should be able to just remember the port numbers and use tcp/udp src/dst for all signaling traffic. It doesn't make sense to me that we can be overkill with some, but not with others. Since my lab is next week, I'm going to just memorize it per IPExpert and hope for the best, instead of trying to make sense of it. I think it's worth bringing up to the proctors though. Anyone have any thoughts or suggestions on this? Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME?
My 2 cents/never got this right though: When calling CCMCME, you'll be using the SIP DP: IPIP-CME. When CME phone presses a digit now, DTMF tones are sent using RFC2833 (I checked that), and would need to be stripped when transported over h323 to CCM phone. I'm not 100% sure about this though, a bit confused actually about the 'apply digit stripping on inbound SIP dialpeer' statement - but this makes the most sense for me. Any other thoughts? Juan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Thursday, April 17, 2008 5:36 PM To: Gregory Jost (grjost) Cc: CCIE Voice; Jacob Owen Subject: Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME? Also, silly question on 4.9: We have CCM - IPIPGW - CCME CCM to IPIPGW to CCME is H323 to SIP CCME to IPIPGW to CCM is H323 to SIP If I am reading it correctly Then it says to ensure when calls are coming from SIP to H323 that we strip RFC2833 but we aren't going from SIP to H323, just the other way around... I also thought SIP to H323 wasn't supported... Jonathan On Thu, Apr 17, 2008 at 9:17 AM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: Oh. I guess that would be literally huh... :) I only know what I talking about a fraction of the time. Maybe I should stay off these forums. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Thursday, April 17, 2008 9:13 AM To: Gregory Jost (grjost) Cc: Jacob Owen; CCIE Voice Subject: Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME? They have a screenshot of the non-GK ICT being created Jonathan On Thu, Apr 17, 2008 at 9:11 AM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: I think ICT is being used figuratively here (e.g. a trunk between disparate systems), not literally. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JonathanCharlesSent: Thursday, April 17, 2008 9:03 AM To: Jacob OwenCc: CCIE VoiceSubject: Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME? No, I am looking at the solution thingy, and it shows a non-GK controlled ICT trunk... bizarre... I thought this was exclusively for connecting to another CCM cluster... in fact, the SRND says so... Never tried hooking it up to an IPIPGW... Jonathan On Thu, Apr 17, 2008 at 8:59 AM, Jacob Owen [EMAIL PROTECTED] wrote: Jonathan, The 172.x.10y.1 addresses are the loopbacks for the devices: 172.x.100.1 - HQ Router (x is pod number) 172.x.101.1 - BR1 Router 172.x.102.1 - BR2 Router I think it's probably an ICT trunk to an H323 Gatekeeper but I don't have the book handy. --- Jonathan Charles [EMAIL PROTECTED] wrote: Does this work? Looking at solution for 4.9 in the workbook, and it shows a regular ICT but the IP address, 172.1.100.1 doesn't match up to anything... I am curious what it is... BR2's loopback should be 172.X.102.1 not 100.1... typo or am I lost? Jonathan Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ __ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
Re: [OSL | CCIE_Voice] VATS and Percentage Bandwidth
that's indeed what happens when enabling traffic shaping. For more info, there's a thread on this of some 2 weeks old. cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Narvaez Sent: Friday, April 11, 2008 5:51 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VATS and Percentage Bandwidth Guys, It is normal that when I have configured VATS, once performed the comand show policy-map it displays the percentage bandwidth in relation to the CIR (33%/ 121 Kbps), and it does not apply the percentage in relation to the command bandwidth configured under the serial ??? --- interface Serial0/1/0:0.1 point-to-point bandwidth 768 ip address 162.9.102.2 255.255.255.0 frame-relay interface-dlci 102 class LFI-SHAPE ! policy-map SHAPE class class-default shape average 729600 3648 0 shape adaptive 368400 shape fr-voice-adapt deactivation 30 service-policy LLQ --- P19-BR2-RTR#show policy-map interface Serial0/1/0:0.1 Serial0/1/0:0.1: DLCI 102 - Service-policy output: SHAPE Class-map: class-default (match-any) Adapt Queue Packets Bytes Packets Bytes Shaping Active Depth Delayed Delayed Active BECN 0 22141 620342453868 no Voice Adaptive Shaping inactive Service-policy : LLQ Queueing Strict Priority Output Queue: Conversation 40 Bandwidth 33 (%) Bandwidth 121 (kbps) Burst 3025 (Bytes)
Re: [OSL | CCIE_Voice] Inbound policing as per QOS SRND
Thanks Scott, it surely helped confirm what I started to suspect after sending out this mail - that it rounded up to 128Kbps to take into account overhead too, with some extra on the side cheers mate, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Monasmith Sent: Thursday, April 03, 2008 5:55 PM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Cc: CCIE Maillist Subject: Re: [OSL | CCIE_Voice] Inbound policing as per QOS SRND 64k is only the g711ulaw codec bandwidth. Below is the reason why... The calculation would be as follows: BW = ([L2 overhead + IP_UDP_RTP Overhead + Sample Size] / Sample_Size) * Codec_Speed BW = ([32+40+160]/ 160) * 64000 BW = 92.8k For a better understanding, read page 1-15 of the QoS SRND 3.3 HTH, Scott On Mon, Mar 31, 2008 at 10:16 AM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) [EMAIL PROTECTED] wrote: Small Q: why does the QOS SRND polices inbound voice bearer (p.2-37) to 128Kbps to limit a port to 1 call max - as it's inbound, and thus one-way? I would think of 64K instead. cheers, Juan -- There are only 10 types of people in the world: Those who understand binary, and those who don't
[OSL | CCIE_Voice] Inbound policing as per QOS SRND
Small Q: why does the QOS SRND polices inbound voice bearer (p.2-37) to 128Kbps to limit a port to 1 call max - as it's inbound, and thus one-way? I would think of 64K instead. cheers, Juan
Re: [OSL | CCIE_Voice] Fwd: bandwidth usage
Jason, not sure but: when you do sh policy-map interface XX, you see the call signaling and bearer in their classes with the used bps. This is averaged out over 5 min by default, but I think you can use the command 'load interval' to tune this interval of needed. If this can be of any help.. cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jason sung Sent: Friday, March 28, 2008 10:50 PM To: Mark Snow Cc: CCIE Maillist Subject: [OSL | CCIE_Voice] Fwd: bandwidth usage Mark, can you please shed some light on this question. Either I am asking someting so stupid nobody wants to answer OR I am asking something impossible? Basically I am trying to send few g711 calls and check the bandwidth and than compare it with few g729 calls. -- Forwarded message -- From: jason sung [EMAIL PROTECTED] Date: Thu, Mar 27, 2008 at 9:25 PM Subject: bandwidth usage To: CCIE Maillist ccie_voice@onlinestudylist.com I have been trying different commands, but none of them give me a definative answer on HOW TO CHECK BANDWIDTH USAGE on the router? Does anybody have any ideas? I tried the show policy-map interface command but that does not show me what I want.
[OSL | CCIE_Voice] exchange server and AD cleanup for Unity
Is there a nice way to clean up mailbox leftovers on the exchange system? I used the 'Bulk Subscriber Delete' from the Unity toolbox, but this only deletes Unity subscribers' mailboxes that are defined in the Unity DB. Is there a way to clean up the message store completely, as I get a 'system error' every time I want to leave a message and I think it's related to adding new users on Unity, whose settings conflict with leftover mailbox information. What tool could I use to delete all mailboxes on exchange and all (unity) accounts in AD? cheers, Juan
[OSL | CCIE_Voice] Multicast MOH over trunk supported?
Is it normal that UCM streams MOH in unicast over a trunk, even if the MRG is setup to use multicast? Perfmon shows unicast is being used, and the holdee receives MOH (MOH server is multicast enabled, moh source is multicast enabled and MRG is multicast) In other words, is multicast MOH supported over a h323 or sip trunk? Anybody ? cheers, Juan
[OSL | CCIE_Voice] inconsistent AAR and CAC
I have GK BW control for all calls off the CCM cluster, towards CME. For
this, I use 2 trunks - GK controlled - in a route group and the first
trunk uses g711, the second g729. Within the CCM cluster, locations CAC
is used to control BW for intracluster calls between sites. My setup is
such that only 24k is available to BR1. Calls to the CME are via route
pattern 3xxx in partition 'internal'.
The weirdest thing happens: when placing a call from BR1 to CME (via GK
trunk), initially I get the out of bandwidth message - which seems
normal to me (CAC: 24k for the first route group option ('g711' trunk)
is not enough and AAR should be kincking in). But, if I wait long enough
after having reset the CCM service, it fails over to the second option
in the route group: the trunk using g729 ! Why does locations CAC not do
it's job here (nowhere I configured AAR yet, so that's not active) ?
From perfmon I observe:
when failing over to the second trunk: only 1 out of resource is
indicated
when not failing over to the second trunk, and CAC simply preventing the
call: 2 out of resources per call are shown (?)
This just doesn't make any sense - is this a known issue of has anyone
ever seen this happen?
regards,
Juan
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 25, Issue 102
Allright ! looking forward to work together - I am sure it will help us both! My lab is planned for July 15th - and still looking forward to it ;) - think I started 2nd half of December with the labs Anyone else starting off/about to start/just started the challenge labs? From: Paul and Bobs [mailto:[EMAIL PROTECTED] Sent: Friday, March 21, 2008 10:23 AM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Cc: CCIE Voice Online Study List Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 25, Issue 102 Great idea.. How long have you been studying for and how long before you write your lab?? On Fri, Mar 21, 2008 at 7:50 PM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) [EMAIL PROTECTED] wrote: Hi Paul, nice to hear you are starting end of next week with the challenge labs. That puts us more or less on the same spot - as I am planning to start with them at April 1st. I hope this opens up some extra possibilities to work together. Are there any others starting more or less with their challenge labs? cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul and Bobs Sent: Thursday, March 20, 2008 9:12 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 25, Issue 102 Hi guys I know the NDA on the CCIE prevents everyione from telking to much but the Proctor lab, which one is closest to how the real lab feals. I am busy working my way through the proctor guide and work book and will be attmpting the multi protocol challenges from end of next week. Is it the challenges that simulate what the lab is like and is those that you need to aim to get done in around 6-7 hours. Thanks Paul On Fri, Mar 21, 2008 at 3:00 AM, [EMAIL PROTECTED] wrote: 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: RTP stream addresses used (Jose Linero Welcker) 2. Re: R: tech-prefix on gatekeeper (Jose Linero Welcker) -- Message: 1 Date: Thu, 20 Mar 2008 16:37:30 + From: Jose Linero Welcker [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] RTP stream addresses used To: Edward French [EMAIL PROTECTED], Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) [EMAIL PROTECTED], CCIE Voice Online Study List ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Also, if you are using SIP as one of the call legs, use the bind command under voice service voip, and sip. Date: Thu, 20 Mar 2008 09:02:52 -0700From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] RTP stream addresses used On your gateways your h323-gateway voip bind command under the interface will choose the source of the RTP - Original Message From: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) [EMAIL PROTECTED]To: CCIE Voice Online Study List ccie_voice@onlinestudylist.comSent: Thursday, March 20, 2008 11:55:22 AMSubject: [OSL | CCIE_Voice] RTP stream addresses used Hi, I have a call setup from CCM to CME via IPIP GW. The call is setup, a codec gets involved to convert g711 (CCM) to g729 (CME) - so seemingly no worries. Is there a way to tell the IPIP GW or CME device which IP addresses to use for the RTP? On the IPIP I have for example a dialpeer with session target ipv4 address, for call signaling purposes. But the RTP to CME is setup to another IP address on the CME router. Is there a way so that the chosen src/dst IP addresses for the stream are set
Re: [OSL | CCIE_Voice] Whats Missing
of course, but reading some feedback on people who've tried already might be helpful in my opinion - it's a personal matter of course - each and everyone of us does these exercises on it's own terms - but it's to get a general idea before we even try the lab :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ccievoice1 Sent: Thursday, March 20, 2008 11:36 AM To: Edward French Cc: CCIE Maillist Subject: Re: [OSL | CCIE_Voice] Whats Missing Give the lab a try!! And you will know what is missing. Well, i personally think it might be better to complete within 6 - 7 hrs so you can have time for troubleshooting. On Thu, Mar 20, 2008 at 6:32 PM, Edward French [EMAIL PROTECTED] wrote: I have seen several comments about people who have attempted the test in the past couple of weeks. How well did the Proctor labs ultimate lab guide prepare you for the test? What areas did you find the most difficult? I can quickly and without reference perform all tasks in the books with the exception of: IPMA, Fast/Quick Dial, EM, Fax, BACD, QOS on 6500, QOS on FR,and sometimes Gatekeeper gets me. I can quickly find the IPMA, Fast/Quick Dial, EM, Fax and BACD on the univercd or other available source. and I can usually complete the full lab scenarios in th 7:45 proctor lab session. Based on your experience with the lab and my above statements do you think I am ready to take the lab? Additionally I have been working in voice for 21 years, I have been a CCNA for I think 10 years and I have been Microsoft certified since NT. Thanks for your opinions Ed
Re: [OSL | CCIE_Voice] MTP Requirement for H323 Trunk
My opinion about this: the MTP reflects the requirement *on the other
side of the trunk*: so depending to which device the H225 setup from CCM
finally goes an MTP might or might not be required (independently of
being GK or non-GK controlled)
Cheers,
Juan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Charles
Sent: Wednesday, March 19, 2008 12:02 AM
To: Devildoc
Cc: CCIE Voice Online Study List
Subject: Re: [OSL | CCIE_Voice] MTP Requirement for H323 Trunk
As I was reading it, it made no sense to me either... it appears the
need for the MTP doesn't make sense in this context... I am sure Mark
Snow can enlighten all of us...
Jonathan
On Tue, Mar 18, 2008 at 4:30 PM, Devildoc [EMAIL PROTECTED]
wrote:
Thanks for the good info man. So from the provided info, since my
non-gk-controlled ICT trunk requires MTP, it must be using the older
version of H323. And my gk-controlled trunk must be using the newer
version of H323 because it does not require MTP.
Well, that's strange since the two trunks mentions above originate
from the same CCM server to the same HQ-RTR using the same version of
IOS. That makes no sense to me. Oh well.. i'll chuck it up to one of
those mysteries.
Thanks for the information.
JD
Date: Tue, 18 Mar 2008 15:46:42 -0500
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MTP Requirement for H323 Trunk
Oh... my guess is that it is using H323v1...
From all that I have read, the MTP is only required for older,
crappier
h323...
From the CCM SRND:
H.323v2 implements Open/Close LogicalChannel and the
emptyCapabilitySet features. The use of
H.323v2 by H.323 gateways, beginning in Cisco IOS Release 12.0(7)T
and Cisco Unified CallManager Release 3.0 and later, eliminates the
requirement for an MTP to provide supplementary services.
and
Media termination points (MTPs) are generally not required for
normal operation of the H.323 trunk.
They are, however, required for communication with devices that are
H.323 Version 1 or that do not
support the Empty Capabilities Set (ECS) for supplementary
services.
My guess is that your gatekeeper does not support ECS...
Cool note in the SRND tho...
If the MTP Required box is checked, the default behavior is to
allow
calls
on the H.323 trunks even if MTP resources are unavailable or
exhausted. This default behavior might result in no voice path for
the call, but the behavior can be changed by setting the Cisco
CallManager service parameter Fail Call if MTP allocation fails
under the H.323 section to True.
Jonathan
On Tue, Mar 18, 2008 at 3:31 PM, Devildoc [EMAIL PROTECTED]
wrote:
Dude... you're missing my point! I am NOT asking you which trunk
is better. I am asking WHY does the non-gk-controlled trunk
requires the
use
of MTP and the gk-controlled trunk does not require the use of
MTP.
That's
all.
JD
Date: Tue, 18 Mar 2008 15:26:59 -0500
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] MTP Requirement for H323 Trunk
CC: ccie_voice@onlinestudylist.com
The Gatekeeper...
Seriously.
With the GK, the GK is used for CAC and routing... without it,
you would need to full-mesh your trunks (if you have more than 2
clusters...)
Jonathan
On Tue, Mar 18, 2008 at 3:07 PM, Devildoc
[EMAIL PROTECTED]
wrote:
Ok.. so what's the difference between the non-gk-controlled
ICT
trunk
and
the gk-controlled H323 trunk?
I mean why does one trunk use MTP and the other doesn't.
They're all
H323
trunk. The only difference is the gk control.
JD
Date: Tue, 18 Mar 2008 15:04:47 -0500
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MTP Requirement for H323
Trunk
If you are using H323v1 (no one does) and you want to
provide supplementary services (hold, xfer, etc...)
Jonathan
On Tue, Mar 18, 2008 at 2:56 PM, Devildoc
[EMAIL PROTECTED]
wrote:
Hello,
Would someone tell me why is MTP required for
non-gatekeeper-controlled
H323 ICT trunk, but the gatekeeper-controlled H323 trunk
does
not
need
MTP?
As the matter of fact, if i configured the gk-controlled
H323
trunk
to
use
MTP, my calls would not connect. Thanks for any info.
JD
Helping your favorite
cause is as easy as instant messaging. You
IM,
we
give. Learn more.
Helping your favorite cause
is as easy as
Re: [OSL | CCIE_Voice] Issue with Transcoder on HQ Router POD15
Devildoc, your inbound DP for CCMCME call is not g711. Have you configged the codec on the H323 inbound DP (I think it's DP 3001) ? Make also sure the trunk on CCM will use g711 to speak with IPIP. hope this helps - cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devildoc Sent: Tuesday, March 18, 2008 3:53 PM To: CCIE Voice Online Study List Subject: [OSL | CCIE_Voice] Issue with Transcoder on HQ Router POD15 Hello, Can someone help me out with this issue that i am having with the local transcoder on the HQ-RTR? I configure the HQ-RTR to be an IPIPGW. Calls come from CCM to CME via the IPIPGW. The call leg from CCM is G711/H323 and the call leg to CME is G729/H323. The IPIPGW is supposed to engage the local transcoder when the calls pass through, but it did not. As the result, the calls never connect. It just keeps on ringing. I did the show call active voice brief and found out that the codec for the call legs is g729r8 pre-ietf. I think this is the reason why the transcoder did not get engaged due to the pre-ietf bits. So my question is how do my call legs get the pre-ietf version of the g729r8? Is it in a configuration setting somewhere that I can disable? The call from CME (G729/H323) to CCM (G711/H323) is properly transcoded by the IPIPGW. Below is the output from the show call active brief for both calls. Calls from CCM to CME does not work properly, but call from CME to CCM works properly. Any help to resolve this issue is greatly appreciated as i am currently working on a lab that needs resolution. Thank you. 1. Output from show call active brief for calls from CCM to CME. Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 2 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 261D : 58 7925780ms.1 +-1 pid:3001 Answer 1001 connected dur 00:00:00 tx:0/0 rx:0/0 IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf media inactive detected:n media contrl rcvd:n/a timestamp:n/a 261D : 59 7925790ms.1 +-1 pid:3013 Originate 3001 connecting dur 00:00:00 tx:0/0 rx:0/0 IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf media inactive detected:n media contrl rcvd:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 2 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 Dspfarm Profile Configuration Profile ID = 1, Service = TRANSCODING, Resource ID = 1 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 Codec : gsmfr, Maximum Packetization Period : 20 Codec : g729r8, Maximum Packetization Period : 60 Codec : g729br8, Maximum Packetization Period : 60 SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED 01 4.4.21 UP N/A FREE xcode 1 - - - 01 4.4.21 UP N/A FREE xcode 1 - - - 01 4.4.21 UP N/A FREE xcode 1 - - - 01 4.4.21 UP N/A FREE xcode 1 - - - Total number of DSPFARM DSP channel(s) 4 2. Output from show active call brief for call from CME to CCM. Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 2 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Total call-legs: 4 123A : 60 8430160ms.1 +1440 pid:1203 Answer 3001 active dur 00:00:07 tx:340/6800 rx:0/0 IP 172.5.102.1:19444 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 media inactive detected:n media contrl rcvd:n/a timestamp:n/a 123A : 61 8430170ms.1 +1430 pid:1211 Originate 1001 active dur 00:00:07 tx:340/54400 rx:355/56800 IP 10.5.200.21:24664 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw media inactive detected:n media contrl rcvd:n/a timestamp:n/a 0: 62 8431670ms.1 +0 pid:0 Originate connecting dur 00:00:07 tx:340/6800 rx:0/0 IP 172.5.100.1:2000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 media inactive detected:n media contrl rcvd:n/a timestamp:n/a 0: 64 8431670ms.2 +0 pid:0 Originate connecting dur 00:00:07 tx:340/54400 rx:339/54240 IP 172.5.100.1:2000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw media inactive detected:n media contrl rcvd:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 2 Call agent controlled call-legs: 0 SCCP call-legs: 2 Multicast call-legs: 0 Total call-legs: 4 Dspfarm Profile Configuration Profile ID = 1, Service = TRANSCODING, Resource ID = 1
Re: [OSL | CCIE_Voice] DCOM issue
Hi Anup, went through the doc you sent - no info was corrupt tough. Nevertheless - I rebooted the sub/pub, recreated replication and the problem seems solved. It's a weird thing, because before I did the rebuild of the cluster, I was not able to get rid of this error (also went through the doc you sent) - hence the rebuild in the end. But, thanks a lot for reaching out - my CCM is back up and running without probs - which is what counts. cheers, Juan From: Anand, Anup [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 18, 2008 4:33 PM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); CCIE Maillist Subject: RE: [OSL | CCIE_Voice] DCOM issue Please check this tech note http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0 9186a008073f582.shtml Regards, Anup From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Sent: Tuesday, March 18, 2008 10:28 AM To: CCIE Maillist Subject: [OSL | CCIE_Voice] DCOM issue Hi all, did anyone ever have issues with the following error: I rebuild my whole CCM 2 days ago - to get rid of this - , but somewhere along the way today this reappeared :-S. Last thing I did was restarting the IIS services - causing an error, thus rebooting the whole publisher. I think it might have started there. Symptoms are for example the devices do not failover anymore to the publisher when restarting the subscriber. So it looks like SDL intracluster communication is gone... I recreated the replication between sub and pub and checked that it works. Any help on this is more than welcome... cheers, Juan image001.jpg
Re: [OSL | CCIE_Voice] Catalyst 6500 QoS Marking
burst size for Cat6K uses a Tc of .25ms. Hence Bc = rate x 0.00025 or [1518x8], whichever is the greated value. The 1518*8 comes from the worst case MTU for ethernet, expressed in bits/s. Round it up to 13Kbps. If burst size would be smaller, all ethernet traffic would be policed, which is not the intention. ps: cat6k always uses Kbps as units cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ccievoice1 Sent: Friday, March 14, 2008 4:48 PM To: CCIE Maillist Subject: [OSL | CCIE_Voice] Catalyst 6500 QoS Marking ! ! set qos cos-dscp-map 0 8 16 24 32 46 48 56 set qos policed-dscp-map 0,24,26,46:8 set qos policer aggregate VVLAN-VOICE rate 128 burst 8000 drop set qos policer aggregate VVLAN-CALL-SIGNALING rate 32 burst 8000 policed-dscp set qos acl ip IPPHONE dscp 46 aggregate VVLAN-VOICE udp 177.1.101.0 255.255.255.0 any range 16384 32767 set qos acl ip IPPHONE dscp 24 aggregate VVLAN-CALL-SIGNALING tcp 177.1.101.0 255.255.255.0 any range 2000 2002 commit qos acl IPPHONE set port qos 2/45 trust-device ciscoipphone set qos acl map IPPHONE 2/45 ! ! Question: on policer aggregate, burst is to specifies the burst size; valid values are 1 to 32000 kilobits. So what value do we use? And how to calculate the value? Thanks.
[OSL | CCIE_Voice] QOS headaches: mapping on wrong class in router...
Guys, I going crazy: Regarding QOS: packets arrive from the LAN on the router interface. A test policy map inbound on that interface shows the DSCP values I was expecting to see: control traffic remapped to AF33, RTP to default class - like I configured it on the LAN: BR2-RTR#sh policy-map int gi0/0.250 GigabitEthernet0/0.250 Service-policy input: test Class-map: EF (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps Match: ip dscp ef (46) Class-map: CS3 (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps Match: ip dscp cs3 (24) Class-map: AF31 (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps Match: ip dscp af31 (26) Class-map: AF33 (match-all) 6 packets, 522 bytes 5 minute offered rate 0 bps Match: ip dscp af33 (30) Class-map: class-default (match-any) 540 packets, 42098 bytes 5 minute offered rate 2000 bps, drop rate 0 bps Match: any This is how it arrives at the router. Then I remove the test policy map from the interface and watched the counters on the outbound FR PVC, and still see hits in the EF and AF31 class: BR2-RTR#sh policy-map int s2/0/0.1 Serial2/0/0.1: DLCI 102 - Service-policy output: parent Class-map: class-default (match-any) 45164 packets, 2910082 bytes 5 minute offered rate 25000 bps, drop rate 0 bps Match: any Traffic Shaping Target/Average Byte Sustain ExcessInterval Increment Rate Limit bits/int bits/int (ms) (bytes) 729600/7296004563648 0 5 456 Adapt Queue Packets Bytes Packets Bytes Shaping Active Depth Delayed Delayed Active BECN 0 45163 1205300 0 0 no Voice Adaptive Shaping active, time left 30 secs Service-policy : WAN-EDGE Class-map: Voice (match-all) 44862 packets, 2871168 bytes 5 minute offered rate 25000 bps, drop rate 0 bps Match: ip dscp ef (46) Queueing Strict Priority Output Queue: Conversation 40 Bandwidth 33 (%) Bandwidth 120 (kbps) Burst 3000 (Bytes) (pkts matched/bytes matched) 44861/1166386 (total drops/bytes drops) 0/0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent:44861 total, 44861 compressed, 1704718 bytes saved, 986942 bytes sent 2.72 efficiency improvement factor 100% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps Class-map: Call-Signaling (match-any) 189 packets, 25616 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 0 packets, 0 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 189 packets, 25616 bytes 5 minute rate 0 bps Queueing Output Queue: Conversation 41 Bandwidth 2 (%) Bandwidth 7 (kbps) Max Threshold 64 (packets) (pkts matched/bytes matched) 192/26048 (depth/total drops/no-buffer drops) 0/0/0 Class-map: class-default (match-any) 113 packets, 13298 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing Flow Based Fair Queueing ... As far as I know, traffic should match the default class only, as everything entering the router has AF33 or CS0 - so definitely no matching on AF31 and EF like is happening on the serial link. Anybody an idea - I must be missing something important here - and it drives me
Re: [OSL | CCIE_Voice] QOS headaches: mapping on wrong class inrouter...
man o man, I'm gonna share this, it's quite obvious - as it's always when you've found the solution - but as a quick refresher/reminder/anything it can always help: the br2 router was the CME router - not just a branch router. Hence was terminating sccp and sourcing h323 (h323 voip DP to CCM), with it's default af31 marking :-S. That's why cheers, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Sent: Thursday, March 13, 2008 4:38 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] QOS headaches: mapping on wrong class inrouter... Guys, I going crazy: Regarding QOS: packets arrive from the LAN on the router interface. A test policy map inbound on that interface shows the DSCP values I was expecting to see: control traffic remapped to AF33, RTP to default class - like I configured it on the LAN: BR2-RTR#sh policy-map int gi0/0.250 GigabitEthernet0/0.250 Service-policy input: test Class-map: EF (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps Match: ip dscp ef (46) Class-map: CS3 (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps Match: ip dscp cs3 (24) Class-map: AF31 (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps Match: ip dscp af31 (26) Class-map: AF33 (match-all) 6 packets, 522 bytes 5 minute offered rate 0 bps Match: ip dscp af33 (30) Class-map: class-default (match-any) 540 packets, 42098 bytes 5 minute offered rate 2000 bps, drop rate 0 bps Match: any This is how it arrives at the router. Then I remove the test policy map from the interface and watched the counters on the outbound FR PVC, and still see hits in the EF and AF31 class: BR2-RTR#sh policy-map int s2/0/0.1 Serial2/0/0.1: DLCI 102 - Service-policy output: parent Class-map: class-default (match-any) 45164 packets, 2910082 bytes 5 minute offered rate 25000 bps, drop rate 0 bps Match: any Traffic Shaping Target/Average Byte Sustain ExcessInterval Increment Rate Limit bits/int bits/int (ms) (bytes) 729600/7296004563648 0 5 456 Adapt Queue Packets Bytes Packets Bytes Shaping Active Depth Delayed Delayed Active BECN 0 45163 1205300 0 0 no Voice Adaptive Shaping active, time left 30 secs Service-policy : WAN-EDGE Class-map: Voice (match-all) 44862 packets, 2871168 bytes 5 minute offered rate 25000 bps, drop rate 0 bps Match: ip dscp ef (46) Queueing Strict Priority Output Queue: Conversation 40 Bandwidth 33 (%) Bandwidth 120 (kbps) Burst 3000 (Bytes) (pkts matched/bytes matched) 44861/1166386 (total drops/bytes drops) 0/0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent:44861 total, 44861 compressed, 1704718 bytes saved, 986942 bytes sent 2.72 efficiency improvement factor 100% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps Class-map: Call-Signaling (match-any) 189 packets, 25616 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 0 packets, 0 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 189 packets, 25616 bytes 5 minute rate 0 bps Queueing Output Queue: Conversation 41 Bandwidth 2 (%) Bandwidth 7 (kbps) Max Threshold 64 (packets) (pkts matched/bytes matched) 192/26048 (depth/total drops/no-buffer drops) 0/0/0 Class-map: class-default (match-any) 113 packets, 13298 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing Flow Based Fair Queueing ... As far as I know, traffic should match the default class only, as everything entering the router has AF33 or CS0 - so definitely no matching on AF31 and EF like is happening on the serial link. Anybody an idea - I must be missing something important here - and it drives me
[OSL | CCIE_Voice] FRTS: BW command to limit BW to 95% of CIR
Hi, I'm doing exercise 12.8: FRTS. On the HQ RTR, FRTS is activated on both PVCs, which I understand. On the other side, on BR1, the solution also proposes FRTS. My question is: suppose the interface would be a WIC-2T, but configured with the 'bandwidth 1466' on it (95% of CIR). Would that limit the actual line speed of the serial line to 95% of the CIR, removing the need to config FRTS (as there's also only 1 PVC on that link)? I know on T1/E1 it's the number of timeslots that determines the speed of the link, so there the bandwidth command would not be enough. Kind regards, Juan
Re: [OSL | CCIE_Voice] FRTS: BW command to limit BW to 95% of CIR
You mean no LFI on links faster than 768K. On slower links LFI is needed to cope for the serialization delays. Above 768K the max MTU of ethernet does not generate more than 10ms serialization delay - so LFI is not needed there. Avoiding DE frames with FRTS makes sense to me. The applied BW to the policy map does indeed classify/prioritize the available BW, but FRTS makes it such that this BW is averaged-out - so that makes sense to me Jonathan. Thanks both for the feedback! Mark/Vik, you also agree on this? Thanks for the replies guys ! -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 12, 2008 9:32 PM To: Devildoc Cc: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] FRTS: BW command to limit BW to 95% of CIR The QOS SRND sayd not to use LFI on links slower than 768k, you should ALWAYS use FRTS on ALL Frame Relay circuits to avoid DE frames. Jonathan On Wed, Mar 12, 2008 at 3:28 PM, Devildoc [EMAIL PROTECTED] wrote: Exercise 12.8 does not ask you to configure FRTS. It only asks you to configure LLQ. To answer your question, if you configure an interface with a bandwidth command, then the router would use the configured bandwidth amount to calculate for any bandwidth-related policy that you may have for that interface. QoS SRND does not recommend the use of FRTS for link less than 768Kbps because the high CPU usage does not warrant the benefits of FRTS for faster speed links. JD Date: Wed, 12 Mar 2008 21:00:05 +0100 From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] FRTS: BW command to limit BW to 95% of CIR Hi, I'm doing exercise 12.8: FRTS. On the HQ RTR, FRTS is activated on both PVCs, which I understand. On the other side, on BR1, the solution also proposes FRTS. My question is: suppose the interface would be a WIC-2T, but configured with the 'bandwidth 1466' on it (95% of CIR). Would that limit the actual line speed of the serial line to 95% of the CIR, removing the need to config FRTS (as there's also only 1 PVC on that link)? I know on T1/E1 it's the number of timeslots that determines the speed of the link, so there the bandwidth command would not be enough. Kind regards, Juan Helping your favorite cause is as easy as instant messaging. You IM, we give. Learn more.
[OSL | CCIE_Voice] unassigned DN numbers
hi all, does anyone know if there is a way to automatically delete unassigned DN numbers from CCM? regards, Juan
[OSL | CCIE_Voice] CME caller
Hi, I was trying to configure CME such that 2 users can intercom to each other, as per exercise 14.3. It's specifically asked to configure the intercoms such that both parties can have an immediate 2-way communication - apart from the barge-in functionality. But apparently, I have to choose between configuring 'barge-in' or 'no-mute' - configuring both of these options is not possible. Is there another way to make it such that the mute is not invoked by the intercom? But as a more general question: when you expect a 'bug' causing you a headache, where do you start looking? Is there a way or method to (quickly) find what you're looking for? (for example, under telephony-service, the command 'caller-id name-only' doesn't seem to do very much) regards, Juan
Re: [OSL | CCIE_Voice] VPIM from Unity to CUE
Please disregard the mail below - I found the missing config - it was a user problem, as usual: the addressing options on the primary locations page do the trick of course :-D This makes my day ;) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Sent: Monday, February 25, 2008 9:41 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VPIM from Unity to CUE Hi, I have a problem getting VPIM to work in the Unity = CUE direction. The opposite direction works just fine: leaving messages on the Unity server, by dialing the locationID + HQ/BR1 phone number, leaves a message in the corresponding Unity subscriber's mailbox. But when I try the same in the Unity = CUE direction, I get the Unity response stating 'the extension you entered was not found, try a different extension'. On the CUE box I don't even see any VPIM messages coming in (sh trace networking vpim receive). I verified that the Delivery Location exists on Unity, with DialID set to 331. So when composing a message from within Unity, sending it to 3313001 (3001=CUE primary extension), this should be handed over to the VC on the exchange server. The SMTP connection to CUE also seems to be working: The only thing I can find is that the Voice Connector properties in the Exchange system manager applet is not showing the MTS-IN and MTS-OUT queues, like in the workbook - but I followed each and every step as documented on CCO to setup VPIM - so I am not sure as to whether this is normal for Exchange2003 (and also: VPIM CUE = Unity seems to work) Does anyone have an idea what can be missing - or encountered the same ? As always, thanks for the help regards, Juan Outlook.jpg
[OSL | CCIE_Voice] FW: CUE and the GMD
I got it working in the end! However, I doubt that the method I used is not really what the workbook suggests to use - and that it's way to complicated the way I've done it... Let me briefly explain the way I set up exercise 16.2: using the CUE AA script from lab 15, pressing 2 to get to the support Q redirects the call to the extension 3500. This corresponds in CME to the B-ACD's AA trigger. Via a loopback dialpeer I invoke the AA script, and use it in drop-though mode to redirect the call to a hunt group's queue (used as the supportQ). Via the param voice-mail command I redirect the call back to CUE - the SupportQ's GDM - if no agents are available in the huntgroup. I really doubt that this is what was needed for this exercise, as the proctor guide nor the solutions guide provide any indication to setup any B-ACD (the context I'm talking about in the mail below), exept for the param voice-mail command. Mark or Vik, can you please provide some feedback on this - how you guys want this exercise to be completed? Kind regards, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Sent: Saturday, February 23, 2008 9:46 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE and the GMD Hi all, I was working on the exercise 16.2, and I have been stucked on the proposed solution: I just can't see how the solution works... I suppose it's an alternative to configuring the 'final' parameter in the SupportQ hunt-group. Mark, Vik or anybody, could you please shed a light on how configuring 'param voice-mail' pointing to the GDM primary extension does make things work? What happens exactly with this command in this context? What am I missing? Many thanks for the feedback and the support, Juan
Re: [OSL | CCIE_Voice] FW: CUE and the GMD
Hi Mark, thanks for the feedback, I appreciate it. But there's something I must be missing in this: the whole idea is that users use the CUE's AA to get into the support Q (huntgroup), right? If then there's no agent available, use the param voice-mail command under the b-acd's AA application to provide the alternate destination for a huntgroup: application service aa flash: param voice-mail 3215 Can you give me an idea about how this works please? As per the solution guide the B-ACD's AA application is not invoked on any dial-peer, yet we use it's configuration... That was why I was trying to invoke the AA on an inbound dialpeer. Kind regards, Juan From: Mark Snow [mailto:[EMAIL PROTECTED] Sent: Monday, February 25, 2008 2:27 AM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] FW: CUE and the GMD Juan, Well that particular question states: Create a General Delivery Mailbox for the Support Queue but give it the extension of 3215. Ensure that any phone in the office can access this mailbox by pressing 9 after they sign into their VM box. Finally, modify the Support Queue (not the hunt group) so that if agents are un-available they will go to this GDM. Also ensure that all BR2 phones see if there is a message waiting in the GDM. So we ask you to modify the Support Queue (but specifically not the hunt group nor the CUE based AA option2 accessing the hunt group) - to make sure that anyone not being able to reach the Support HG (if no one is logged in perhaps) so that they go to a GDM. We set the VM parameter - and then instead of pointing directly to the VM Pilot - we send it to a dummy ephone-dn (not assigned anywhere) that forwards always to VM so that the VM Box of that dummy ephone-dn shows as the Redirecting DN (RDNIS). Then we assign that same DN to every ephone - so that when MWI is set to on - we see the message icon next to that line on all the phones. What you are talking about (while not what this question asked for) is a good way of allowing ephone-dns or CUE itself to be able to trigger Applications since they have a limitation - namely that Applications cannot be triggered on an Outbound DP - they must be triggered on an Inbound DP. I will expound on it here for the benefit of others wondering a little bit more about how you accomplished what you are describing. If an ephone-dn (or CUE) goes to call an application Pilot - their POTS DP is the Inbound - and so we have to find another inbound DP. So to accomplish this we send the call Outbound on a DP like this: ! dial-peer voice 3620 voip destination-pattern 3000 session-target ipv4:172.1.102.1 ! And since we sent the call to our own router's Loopback interface - the call comes right back to us and we pick it up and trigger the Application with a DP like this: ! dial-peer voice 3625 voip service aa incoming called-number 3000 ! Hope that helps you and maybe some others here as well, Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Feb 24, 2008, at 7:11 AM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) wrote: I got it working in the end! However, I doubt that the method I used is not really what the workbook suggests to use - and that it's way to complicated the way I've done it... Let me briefly explain the way I set up exercise 16.2: using the CUE AA script from lab 15, pressing 2 to get to the support Q redirects the call to the extension 3500. This corresponds in CME to the B-ACD's AA trigger. Via a loopback dialpeer I invoke the AA script, and use it in drop-though mode to redirect the call to a hunt group's queue (used as the supportQ). Via the param voice-mail command I redirect the call back to CUE - the SupportQ's GDM - if no agents are available in the huntgroup. I really doubt that this is what was needed for this exercise, as the proctor guide nor the solutions guide provide any indication to setup any B-ACD (the context I'm talking about in the mail below), exept for the param voice-mail command. Mark or Vik, can you please provide some feedback on this - how you guys want this exercise to be completed? Kind regards, Juan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Sent: Saturday, February 23, 2008 9:46 PM To: ccie_voice@onlinestudylist.com
Re: [OSL | CCIE_Voice] called number in Unity's CallViewer mimicsForwardingnumber
I am trying out VM integration at the moment, over a ISDN PRI, just to see it working - as I was told it should also be working with PRI. But: the called number is extended: Feb 18 18:46:49.138: //-1/B253876B809E/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=912122251600#2001 (2001 being for example the forwarding station) But in debug isdn q931, the called number is the normal number (without #2001 at the end): Calling Party Number i = 0x2180, '6175252002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '2122251600' Plan:ISDN, Type:National I removed the redirect IE from the PRI, to be sure not to be using this IE for VM access, by way of: no isdn outgoing ie redirect-number. The problem is that the called number IE is not modified to append the DTMF digits - thus the sign-in greeting is heard for subscriber 6175252002. My question is whether this is something that's normal when using VM-integration with PRI lines - as I only see this feature documented for 'analog' lines... Can anybody tell me? regards, Juan From: jason sung [mailto:[EMAIL PROTECTED] Sent: Monday, February 18, 2008 7:53 PM To: [EMAIL PROTECTED] Cc: Matthew Cody; [EMAIL PROTECTED]; Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); ccie_voice@onlinestudylist.com; Matthew Saskin Subject: Re: [OSL | CCIE_Voice] called number in Unity's CallViewer mimicsForwardingnumber Brilliant. On Feb 18, 2008 12:50 PM, Vik Malhi [EMAIL PROTECTED] wrote: I'll post that again with the correction:-) the translation pattern should be 166x in my earlier example (1) Ensure that the HQ site has some spare DID numbers- e.g. in PL we route 21222x1...to the HQ gateway. (2) Use the alias command in call-manager-fallback to route each extension to a unique DID number. E.g. Call-manager-fallback voicemail 912122211600 alias 1 2001 to 2001 cfw 912122211661 timeout 12 alias 2 2002 to 2002 cfw 912122211662 timeout 12 alias 3 2003 to 2003 cfw 912122211663 timeout 12 (3) On the CallManager create a Translation Pattern as shown below: DN = 166X / pt-internal CSS = css-internal Called # Mask = 200x (4) When the CCM receives the call it tries to ring 200X which is not registered. It will then use the call fwd no answer setting (which should be send to VM). (5) Add the Alternate Extension on Unity so that direct calls are routed to subscriber sign-in. Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com http://www.ipexpert.com/ Toll Free: +1.866.225.8064 International: +1.810.326.1444 -Original Message- From: Vik Malhi [mailto:[EMAIL PROTECTED] Sent: Monday, February 18, 2008 10:49 AM To: 'Matthew Cody'; '[EMAIL PROTECTED]'; 'Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)'; 'jason sung' Cc: 'ccie_voice@onlinestudylist.com'; 'Matthew Saskin' Subject: RE: [OSL | CCIE_Voice] called number in Unity's CallViewer mimicsForwardingnumber I'll present a different strategy that works when a router is in SRST mode and you cannot rely on the RDNIS for whatever reason. It allows you to avoid dealing with vm-integration. Let me stress, in the field the workaround is to upgrade the IOS to a release with the RDNIS issue fixed and beg your telco to pass the RDNIS. (1) Ensure that the HQ site has some spare DID numbers- e.g. in PL we route 21222x1...to the HQ gateway. (2) Use the alias command in call-manager-fallback to route each extension to a unique DID number. E.g. Call-manager-fallback voicemail 912122211600 alias 1 2001 to 2001 cfw 912122211661 timeout 12 alias 2 2002 to 2002 cfw 912122211662 timeout 12 alias 3 2003 to 2003 cfw 912122211663 timeout 12 (3) On the CallManager create a Translation Pattern as shown below: DN = 1661 / pt-internal CSS = css-internal Called # Mask = 200x (4) When the CCM receives the call it tries to ring 200X which is not registered. It will then use the call fwd no answer setting (which should be send to VM). (5) Add the Alternate Extension on Unity so that direct calls are routed to subscriber sign-in. Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com http://www.ipexpert.com/ Toll Free
[OSL | CCIE_Voice] Conference bridge issue
Hi Mark et all, I thought I understood the media resource part of CCM, but I encounter something strange and wonder if it's normal. It's related to Q9.8 in the Lab workbook, where you conference in a CTI RP, forwarded to Unity, the be able to record the session. For some reason the conference wasn't set up: Cannot complete conference. Going back a step, and trying to setup a simple conf call between HQ (1001) and BR1 (2001 and 2002), I realised the following: 1001 calls 2001 : takes 24K, perfmon shows 1 call in progress from the BR1 locations counter 1001 presses conference and calls 2002: this puts 2001 onhold, MOH server at HQ streams, perfmon shows 2 calls in progress from the BR1 location (all what I expected) 1001 presses conference again, to setup the conference: a) if BR1 location set to 48K: Cannot complete conference : perfmon shows the HW CFB at HQ gets active (as 1001/HQ is conference initiator), but then gets disconnected and immediately after this event the locations 'out of resource'' for BR1 gets incremented. How is this possible: there should be 2 calls over the WAN once the CFB is in place: 2001 -- CFB and 2002 - CFB ? So 48K should be sufficient b) to verify the previous point, I set the BR1 location to 72K and did the same: now the conference is up, AND: the BR1 location on perfmon indeed shows 2 calls over the WAN, consuming 48K, with still 24K available. So to come back on a): this should have worked with 48K too, but it simply doesn't: you have to set it to a least 72K... :-S Kind regards, Juan
[OSL | CCIE_Voice] Unity Fax server
Hi guys, the FAX ID in the subscriber's field, is that the one used to send a fax to a specific subscriber's inbox? I'm not sure whether it is the Fax Delivery Number that must be set to correspond to the digits Unity monitors on an inbound fax or that Fax ID is used (the Fax delivery Number is only used for outbound faxes?). It's that I don't have a fax to test this at the moment and I'm not sure. Regards, Juan
Re: [OSL | CCIE_Voice] inbound dial-peer matching on MGCP
Hi Mark, the config is working :) - at the time I posted the mail below I was under the impression you would have dial-peer matching inbound on the MGCP PRI - which indeed doesn't happen. I always used the command 'sh call active voice brief' , thinking the PID would indicate the dialpeers being matched on the router. But in MGCP PRI this can indeed be misleading, as this traffic simply is backhauled to the CCM - disregarding the configs (for example service aa) applied on the dial-peer that you see getting matched with the PID/sh call active voice brief. But - I wrote this in another thread, 'TCL scipt for SRST AA , I'm still going to try out what happens with MGCP T1 CAS, where you'd have a MGCPAPP dial-peer, and at the same time a H323 dial-peer on the router: can the h323 dialpeer be matched instead of the MGCP, in which case you mentioned Mark, to use the preference command to make sure the MGCP dialpeer gets matched as long as MGCP is up and running. But I still need to try this setup out though. Up till now, the MGCP PRI has already unveiled a bit more of it's secrets to me :-D - thanks for the clarification on this Mark and Ovais ! regards, Juan From: Mark Snow [mailto:[EMAIL PROTECTED] Sent: Sunday, February 10, 2008 5:51 AM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Cc: ovais Iqbal; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] inbound dial-peer matching on MGCP Did you get this particular configuration working? I never saw a specific response to this thread - so I thought I would add this after re-reading your post - and maybe understanding it a little better: While I mentioned that H323/SIP/MGCP can co-exist next to one another - bear in mind what Ovais said: in the case of MGCP and specifically PRI's - the Q931 is being backhauled to the UCM - and therefore NO matching dial-peer locally is going to change that behaviour - the call will be routed to the UCM regardless. Let us know if it is working now Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Feb 5, 2008, at 9:37 AM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) wrote: Hi Ovais, from the output ''11E2 : 9 4617010ms.1 +0 pid:10 Originate active'' (sh call active voice brie, shows you which call legs/dial peers are matched) : pid:10 indicates that dial-peer 10 is matched. For example, if I add dial-peer with ''incoming called-number. '' , the PSTN call effectively chooses that dial-peer instead, even if it's a H323 dial-peer :-S cheers, Juan From: ovais Iqbal [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 05, 2008 3:26 PM To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] inbound dial-peer matching on MGCP In the case of MGCP you don't match those H323 inbound or ourbound dial-peers at all, how are you seeing dial-peer 10 is being matched? in your command out put I don't see any reference to dial-peer 10. I guess you are looking at the wrong place or missing a point. On 2/5/08, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) [EMAIL PROTECTED] wrote: Hi, on the BR1 router, I have a MGCP PRI endpoint (port 1/0:23). This port is backhauled to the CCM. When a call comes in, the L3 (calling, called number...) is transported to the CCM which does call routing. My question is, how does the router do inbound dial-peer selection on the MGCP GW? I see that for an incoming call from PSTN dial-peer 10 is being used: 11E2 : 9 4617010ms.1 +0 pid:10 Originate active dur 00:00:08 tx:383/61280 rx:423/67680 Tele 1/0:23 (9) [1/0.1] tx:7700/7700/0ms g711ulaw noise:-53 acom:-1 i/0:-49/-50 dBm However, the config is like: dial-peer voice 10 pots incoming called-number 617521280. direct-inward-dial Why is there an inbound match on dial-peer 10 in this case - as there's even not a port 1/0:23 configured on the dial-peer (to be chosen as last resort inbound dial-peer matching for a POTS PRI port) kind regards, Juan
