[OSL | CCIE_Voice] Voice rack for rent
Hi, I have a voice rack at home with the following devices. It is the exact setup as the current CCIE lab. 4X2811 routers all having PVDM Chips 1X3750 Two 2811 having four port switch module with POE. AIM-CUE 3X7960 3X7961 A server with i5 Processor, 1 TB harddisk and 16gb ram With the UC servers installed and setup. Have the practice WB's as well. Let me know if any one wants to rent it, i can share access remotly. ___ 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] creating vm box in connection
Hi, If in the end user page the first name is not given..do we need to add first name, import user to unity then remove first name..?? as i beleive we are not supposed to change or modify any setting which were configured prior..? 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
[OSL | CCIE_Voice] access to cli in lab
Hi, Do we have access to the server cli in the lab. Its vital for troubleshooting dbissues. 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
[OSL | CCIE_Voice] protctors lab vouchers
Hi, I have some 23x8 hours procotor lab vouchers, i got my own rack so its of no use for me. If any one interested let me know. 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
[OSL | CCIE_Voice] Does cisco repeat labs
I know the mood is not good in the forum due to the colloboration and step dad treatment by metting out voice altogether. Have got my lab coming so cant afford to get distracted. Is there any chances that we get the same lab again on the second attempt..? or do we get a different lab the next time for sure.. 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
[OSL | CCIE_Voice] use of the ^ at the beginning of POTS dial-peer ^9.......$
Hi, Pots dial peer strips only explicitly matched digits. Since you use a ^ the explicit digit match it stopped after that, so it will send what ever number comes after the ^. Thanks --- On Sat, 11/5/13, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com wrote: From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 87, Issue 22 To: ccie_voice@onlinestudylist.com Date: Saturday, 11 May, 2013, 10:28 AM 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. CUCM upgrade (Dharambir kumar varma) 2. Re: CUCM upgrade (William Bell) 3. Cisco SRST Phone (Dharambir kumar varma) 4. Use of the ^ at the beginning of of a POTS dial peer = no digit strip ? (VanBenschoten, Brian) -- Message: 1 Date: Sat, 11 May 2013 02:32:44 +0530 From: Dharambir kumar varma dharambi...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUCM upgrade Message-ID: CA+iWkJQZoh=vmpqgcxw0q72nbywov9l7zdud_jj6eyu1ph-...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hello sir, I want to upgrade CUCM 7.0.1.. to CUCM 8.0.. What is required...for this upgradation. is our existing licences supported ? please help -- * Thanks Regards,* *Dharambir Kumar* -- Message: 2 Date: Fri, 10 May 2013 18:01:55 -0400 From: William Bell b...@ucguerrilla.com To: Dharambir kumar varma dharambi...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUCM upgrade Message-ID: eb67f90c-b0e6-42ab-be94-7297b7d2f...@ucguerrilla.com Content-Type: text/plain; charset=us-ascii That is an extremely open question. There are a lot of moving parts to doing any upgrade. What is required? Planning. Where do you start? Look at the CUCM software and hardware compatibility guides. That will help you navigate software dependencies, hardware dependencies, and map out your upgrade path. You will be able to determine if you can do a direct upgrade or if you need to multi-hop. Your hardware will dictate some of this as well. I would then read the release notes and I would also read the appropriate Install/Upgrade guides for your target version. As you get to the 8.6 and later releases, you have other considerations. Upgrading to 8.6 or greater from any release prior to 8.6 is a refresh install. You should research refresh install so that you understand the inner workings and plan accordingly. As far as licensing, you will need to ensure you have a valid support contract with Cisco. Go to the PUT tool and you should be able to figure it out. If you don't see upgrade options for your CUCM then there is something wrong. Either you are out of support or something is out of whack with your account. Contact your Cisco account team or your integrator's account team. You do require a license. Basically, it is a node license allowing you to use major release 8.x. There is much more but I am not going to get into it via email. You basically need to do some research. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 10, 2013, at 5:02 PM, Dharambir kumar varma wrote: Hello sir, I want to upgrade CUCM 7.0.1.. to CUCM 8.0.. What is required...for this upgradation. is our existing licences supported ? please help -- * Thanks Regards,* *Dharambir Kumar* ___ 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130510/ffa4d495/attachment-0001.html -- Message: 3 Date: Sat, 11 May 2013 05:28:15 +0530 From: Dharambir kumar varma dharambi...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco SRST Phone Message-ID: ca+iwkjqtoxpayiwtgnqokhxdgu8mxdzt8+_2sltanoqpzub...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hi we are using centralized cluster at HQ.when wan link goes down, the remote site router comes in srst mode then in that case ip phone registered to SRST.will these phones will use the license of CUCM DLU in case of wan
[OSL | CCIE_Voice] Issue with RSVP based CAC
Are you getting any message on phone related to call like out of bandwidth etc, can u check if it is using AAR to route calls through the PRI. Run a trace on cucm for the calls and check if the call is looking for a MTP for RSVP. --- On Mon, 1/4/13, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com wrote: From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 86, Issue 5 To: ccie_voice@onlinestudylist.com Date: Monday, 1 April, 2013, 8:55 PM 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: Issue with RSVP based CAC (Suresh Bhandari) 2. Re: Issue with RSVP based CAC (Saeed IDris) 3. Re: Issue with RSVP based CAC (Suresh Bhandari) -- Message: 1 Date: Mon, 1 Apr 2013 20:52:45 +0545 From: Suresh Bhandari bring...@gmail.com To: Saeed IDris saee...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Issue with RSVP based CAC Message-ID: CAExcHf=z3LU7JbVW89cDpm56p3y=3dukvep2ola_cyr00uk...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Can you make sure that your calling and called gateways and phones are in different Locations? And further, have you set the bandwidth reservation requirement between your intended locations is set to Mandatory? If your calling and called phones are not is expected Locations, and no Mandatory reservation is there, then no rsvp is involved. Check and make tests. HTH On Mon, Apr 1, 2013 at 7:26 PM, Saeed IDris saee...@gmail.com wrote: Greeting Experts, I need your assistance for RSVP LAB 10 Vol-1 I have configured MTP ,location and added MTP on CUCM ...and its register working fine plus i have added IP RSVP bandwidth commend under sub-interface of frame relay link on both Gateway HQ and BR1 (RSVP over Frame Relay) the issue that RSVP is not trigger even if i call (VOIP) between the two different location after i changed the IP RSVP bandwidth to less than 24k but still i can initiate more than three calls even i run the debug for both gateway I can't see any info related to RSVP packet from both Gateway. what is the missing on my configuration ? Regards, Saeed ___ 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 -- Suresh Bhandari -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130401/56ad729e/attachment-0001.html -- Message: 2 Date: Mon, 1 Apr 2013 18:12:13 +0300 From: Saeed IDris saee...@gmail.com To: Suresh Bhandari bring...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Issue with RSVP based CAC Message-ID: caohbsxpchlvxwwpprverj-4gmov69q-r9khe4b6yny7nk0y...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Thanks Suresh, i think you didn't see my last update which sent by Email: check it pls: Thanks for your quick reply kindly check the below info: for Location its already configured as following: for HQ: relation to HQ= no Reservation and for HQ: relation to Br1= Mandatory for Br1: relation to Br1= no Reservation and for BR1: relation to HQ = Mandatory MRGL HQ: (applied for HQ_DP) MTP_G729 (VGW-HQ) HW_Codec (VGW-HQ) HW_Conf (VGW-HQ) MRGL BR1: (applied for BR1_DP) MTP_G729 (VGW-BR1) HW_Codec (VGW-BR1) HW_Conf (VGW-BR1) HQ_Phone1 (5001) location HQ_Loc (applied via HQ_DP) BR1_Phone1 (1001) location BR1_Loc (applied via BR1_DP) HQ_VGW configured with HQ Device pool BR1_VGW configured with BR1 Device pool hopefully it clear. Regards, Saeed On Mon, Apr 1, 2013 at 6:07 PM, Suresh Bhandari bring...@gmail.com wrote: Can you make sure that your calling and called gateways and phones are in different Locations? And further, have you set the bandwidth reservation requirement between your intended locations is set to Mandatory? If your calling and called phones are not is expected Locations, and no Mandatory reservation is there, then no rsvp is involved. Check and make tests. HTH On Mon, Apr 1, 2013 at 7:26 PM, Saeed IDris saee...@gmail.com wrote: Greeting Experts, I need your assistance for RSVP LAB 10 Vol-1 I have configured MTP ,location and
Re: [OSL | CCIE_Voice] Issue with RSVP based CAC
can you check if the region setting between HQ and BR1 is configued to use g729. ABout the gateway.. remove all reservation on location and checkand again enable it then check. If still issue happens..increase the rsvp bw on the interface to bw require for one call and check if the call is going out through the gateway. By this way we can check if the calls to gateway are trying to see rsvp or not as i see bw u configured is 22kbps. --- On Mon, 1/4/13, Suresh Bhandari bring...@gmail.com wrote: From: Suresh Bhandari bring...@gmail.com Subject: Re: Issue with RSVP based CAC To: Saeed IDris saee...@gmail.com Cc: Ajay Viswanath ajayviswan...@yahoo.co.in, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, 1 April, 2013, 11:19 PM The above IOS config seems enough for RSVP to invoke if you have proper MTP DP, MRG and MRGL set to corresponding Device pools. So, this point onwards, if I were you, I would definitely checked through the CUCM GUI pages for errors. Further, you've written that the calls from HQ to PSTN are not completing what about calls from BR1 side to PSTN? And what about intra-site (HQ to HQ and BR1 to BR1) calls? On Mon, Apr 1, 2013 at 11:15 PM, Saeed IDris saee...@gmail.com wrote: OK, Suresh here is the Voice Gateway for both sites: HQ-VGW: interface Serial0/0/0:0 no ip address encapsulation frame-relay fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/0:0.1 point-to-point -- to BR1 bandwidth 1536 ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ip rsvp bandwidth 22 ! SCCP -HQ ! sccp local GigabitEthernet0/0.10 sccp ccm 10.10.210.10 identifier 2 version 7.0 sccp ccm 10.10.210.11 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/0.10 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register HQCONF associate profile 1 register HQXCODE associate profile 3 register MTP_G729 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! dspfarm profile 3 mtp codec g729r8 codec pass-through rsvp maximum sessions software 15 associate application SCCP ! ! VGW-BR1 interface Serial0/3/0:0 no ip address encapsulation frame-relay IETF fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/3/0:0.1 point-to-point bandwidth 1536 ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ip rsvp bandwidth 22 ! SCCP-BR1 ! sccp local Loopback0 sccp ccm 10.10.210.10 identifier 2 priority 1 version 7.0 sccp ccm 10.10.210.11 identifier 1 priority 2 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register BR1_HWCODEC associate profile 2 register BR1_HQCONF associate profile 3 register BR1_MTP_G729 switchback method graceful ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729br8 codec g729r8 codec g722-64 maximum sessions 4 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 codec g722-64 maximum sessions 2 associate application SCCP ! dspfarm profile 3 mtp codec pass-through codec g729r8 rsvp maximum sessions software 15 associate application SCCP ! Ajay also i need to clarify another issue since Phone1 HQ is already assign to Location HQ if i dial any PSNT number i got busy tone with Not Enough Bandwidth error (911,...etc) while the calling is going to locate HQ Voice gateway same location , already its configured between HQ and HQ no reservation any idea? i will send you the CUCM logs. Regards, Saeed IDris OK, Suresh here is the Voice Gateway for both sites: HQ-VGW: interface Serial0/0/0:0 no ip address encapsulation frame-relay fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/0:0.1 point-to-point -- to BR1 bandwidth 1536 ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ip rsvp bandwidth 22 ! SCCP -HQ ! sccp local GigabitEthernet0/0.10 sccp ccm 10.10.210.10 identifier 2 version 7.0 sccp ccm 10.10.210.11 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/0.10 associate ccm 1 priority 1 associate ccm 2 priority 2 associate
[OSL | CCIE_Voice] List of bugs to know for the voice lab
Can any one provide the list of bugs we might encounter in the lab. Would be helpful to know the workaround for it. 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
[OSL | CCIE_Voice] B-ACD drop through to loop ephone hunt a second time
Suresh, Make sure you enable the music on hold under the cme. If the moh is not specified the BACD will drop the second time it attempts again. Thanks --- On Thu, 28/3/13, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com wrote: From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 85, Issue 104 To: ccie_voice@onlinestudylist.com Date: Thursday, 28 March, 2013, 4:08 AM 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. B-ACD drop through to loop ephone hunt a second time (Suresh Bhandari) 2. Re: QOS big question (ikizoo4 kwon) 3. MVA functionality (CCIEing) 4. Re: B-ACD drop through to loop ephone hunt a second time (William Bell) 5. Re: MVA partial match issue (William Bell) -- Message: 1 Date: Thu, 28 Mar 2013 00:56:11 +0545 From: Suresh Bhandari bring...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-ACD drop through to loop ephone hunt a second time Message-ID: CAExcHf=6g3nivqohbvpnjchxrzccqjd2oyoq2mokoyu+k2f...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Experts! I configured the embedded drop-through script to match the requirement that if, for the first time, both the agents do not pickup the call, it should once more attempt to send the call to the agents. Succeeded for one time only. On the calling phone, I hear the all of our agents ... or so, and goes on hook, never attempts a second time. Somewhere I read to tweak the second-greeting-timer to 35secs or less. Did it, but no avail. can anyone shed light on what should i do to achieve the results? TIA -- Suresh Bhandari -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130328/d3a0ab13/attachment-0001.html -- Message: 2 Date: Wed, 27 Mar 2013 12:19:30 -0700 From: ikizoo4 kwon ikiz...@hotmail.com To: Suresh Bhandari bring...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, singh singh8...@in.com, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS big question Message-ID: bay172-w45ba04dce7f1bb3edccf0cea...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 i am not talking about theory, as you know there is lot of theory around. as you can see i enabled FRF.12 and cRTP , then make 1 g729 call, the bandwidth priority queue has 25K ( not even close to 12) sh policy-map int Serial0/3/0.2 Serial0/3/0.2: DLCI 103 - Service-policy output: AutoQoS-Policy-Trust queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 15075 packets, 964800 bytes 5 minute offered rate 25000 bps, drop rate 0 bps = 25K Match: ip dscp ef (46) 15075 packets, 964800 bytes 5 minute rate 25000 bps Priority: 70% (537 kbps), burst bytes 13400, b/w exceed drops: 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent: 15075 total, 15074 compressed, == cRTP working 572780 bytes saved, 331720 bytes sent 2.72 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps Class-map: AutoQoS-VoIP-Control-Trust (match-any) 343 packets, 20788 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 343 packets, 20788 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 0 packets, 0 bytes 5 minute rate 0 bps Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 bandwidth 5% (38 kbps) Class-map: class-default (match-any) 430 packets, 47701 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0 (pkts output/bytes output) 5/6516 Fair-queue: per-flow queue limit 16 Date: Thu, 28 Mar 2013 00:42:00 +0545 Subject: Re: [OSL |
