Re: [OSL | CCIE_Voice] AXL servers and CUC
In this instance, best practice is a relative concept. Many applications leverage AXL to retrieve information and you can leverage that API to retrieve information on any node in the cluster. Applications that push configurations should be leveraging the publisher to do so. That said, I would think that you could send a SQL update, addphone, updateuser, etc. AXL command to a subscriber node. Of course, it would only succeed if the publisher node is on line. Anyway, my policy is to enable a secondary AXL server in the cluster if I have applications that are leveraging AXL to pull information. Like CUxAC, CUC, UCM IM/P, etc. CCX actually writes using the AXL API. Not that having a redundant AXL server would hurt but if I just have CCX and UCM, I typically go with a single AXL instance. IMO, you will not harm your cluster if you enable AXL on subsequent nodes in the cluster. You must have it on the first node (pub) if you are going to have it at all. In the IE lab, at least the current blue print, you will want to enable the AXL service on both Pub and Sub. If for no other reason than to avoid goofy issues during initialization of CUE in a scenario where CUE registers directly to UCM. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 24, 2013, at 2:38 PM, probert...@gmail.com wrote: Hi, I have few questions related to AXL servers and CUC. Should CUC be configured to use both sub and pub as AXL servers? According to doc: Cisco AXL Web Service Activate on the first node only. Failing to activate this service causes the inability to update Cisco Unified Communications Manager from client-based applications that use AXL. But as far as I know CUC will not be updating anything on CUCM using AXL it is just used to read data during the user import. So since AXL can be activated on SUB can we use it with CUC? I know it works I just want best practice, if we don't than we have no redundancy. I guess this also applies to UCCX. Should we activate AXL on sub in the lab and should we configure CUC to use both AXL servers in the lab? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Testing SRST
I agree with Brian. I started with using the static routes but then went to shutting down the interface. Though, I would shut down the serial interface. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2013, at 9:19 PM, VanBenschoten, Brian wrote: An easier way to test for THEO and such is to just shut down the voice-port (not the controller or serial). Quick and easy and perhaps not as easy to overlook when troubleshooting. I've left my null routes in a couple of times without realizing it. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher Sent: Thursday, October 17, 2013 11:53 AM To: ccievoice Subject: Re: [OSL | CCIE_Voice] Testing SRST If you are talking about testing redundancy I'll do the same thing on the gateway I want to simulate as being down. For example when doing TEHO where if the remote gateway is down we want to fail to the local gateway, I'll go to the remote gateway and put in the static routes. On Thu, Oct 17, 2013 at 11:46 AM, Alex Mendoza aa.mend...@icloud.com wrote: Hi Bill I use your way to test SRST, I'm wondering what are you using for test Route List when they have 2 route groups. best regards. Alex On Oct 17, 2013, at 09:23 AM, Bill Hatcher wchatc...@gmail.com wrote: I have been looking for quick and easy ways to test SRST, and I've found many different waqys of doing this. With the exception of pulling the WAN interface, they all seem to take a lot of time and effort to accomplish. Anything from creating access-lists to block the traffic to creating new call manager groups and shutting down one of the CallManager services. I have found that a couple of simple static routes to the null 0 interface works very well. ip route 10.10.210.10 255.255.255.255 null 0 ip route 10.10.210.11 255.255.255.255 null 0 no ip route 10.10.210.10 255.255.255.255 null 0 no ip route 10.10.210.11 255.255.255.255 null 0 Add them to a notepad and the no statements as well and you can quickly send your devices into srst mode. Now if you have any VoIP dial-peers that point to other addresses across your WAN you may have to add those as well. What do you guys use? HTH Bill. ___ 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 Important Notice: This email message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of Core BTS. Core BTS specifically disclaims liability for any damage caused by any virus transmitted by 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 ___ 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] Base configs and Dial Plan mapping
I documented my strategy in my blog if interested. Part 2 in the series focuses on building various tables and the read-through portion of the exam: http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html Looking back at my notes, I have the following Ent / Service params that I updated by default: Enterprise Parameters: Auto Registration Protocol: SCCP BLF for Call Lists: Enabled Advertise G722 Codec: Disabled URL Authentication: set IP instead of name URL Directories: set IP instead of name URL information: set IP instead of name URL Services: set IP instead of name Connection Monitor Duration: 60 (or do this at a device pool level) Service Parameters BRQ Enabled: True T302 timer: 5000 H225 T302 timer: 5000 G722 codec enabled: Disabled iLBC codec enabled: Disabled Intraregion Audio codec default: G729 Inter-region Audio codec default: G729 Automated Alternate Routing: True Enable Mobile Voice Access: True Inbound Calling Search Space for Remote Destination: Remote Destination Profile + Line Calling Search Space System Remote Access Block Numbers: update as needed Transfer on-hook enabled: True Display Original Calling Number on Transfer from Unity: True Max Forward unregistered hops to DN: 1 Allow peer to preserve h323 calls: True/*need to add appropriate configuration on h323*/ Another service parameter I have seen people modify is the stop routing on unallocated number parameter. People mod this to allow calls to hunt around a H323 gateway that has a PRI which is down. I didn't use this method because I think it is the wrong approach to fixing that problem. I leveraged the IOS config command: no dial-peer out status pots HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ 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] Base configs and Dial Plan mapping
Bill, You can read about the command here: http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_d1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1459139 The important bit is: When the dial-peer outbound status-check pots command is configured, if the voice-port configured under an outbound POTS dial-peer is down, that dial-peer is excluded while matching the corresponding destination-pattern. Therefore, if there are no other matching outbound POTS dial-peers for the specified destination-pattern, the gateway will disconnect the call with a cause code of 1 (Unallocated/unassigned number), So, when you have this command enabled (default) AND you have a single PRI AND that PRI is down, call set up request from UCM to the VG will result in a response of unallocated/unassigned. Why? Because we have told the router to monitor the status of the PRI and intelligently detect when it is down. When it is down, the dial-peer is no longer evaluated during call setup. By turning this off, we are basically telling the VG to go ahead and try to use the busted PRI. Which then results in a different kind of setup error that will let the CUCM know it should continue hunting through its RG/RL configuration. Lots of people leverage the service parameter I mentioned below to route around PRIs that are off line. That is probably fine for the purposes of the IE lab. I prefer to disable status checking at the GW level. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 18, 2013, at 10:06 AM, Bill Hatcher wrote: Bill, One other question, I'm not familiar with the command no dial-peer out status pots What's it do? On Fri, Oct 18, 2013 at 7:44 AM, William Bell b...@ucguerrilla.com wrote: I documented my strategy in my blog if interested. Part 2 in the series focuses on building various tables and the read-through portion of the exam: http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html Looking back at my notes, I have the following Ent / Service params that I updated by default: Enterprise Parameters: Auto Registration Protocol: SCCP BLF for Call Lists: Enabled Advertise G722 Codec: Disabled URL Authentication: set IP instead of name URL Directories: set IP instead of name URL information: set IP instead of name URL Services: set IP instead of name Connection Monitor Duration: 60 (or do this at a device pool level) Service Parameters BRQ Enabled: True T302 timer: 5000 H225 T302 timer: 5000 G722 codec enabled: Disabled iLBC codec enabled: Disabled Intraregion Audio codec default: G729 Inter-region Audio codec default: G729 Automated Alternate Routing: True Enable Mobile Voice Access: True Inbound Calling Search Space for Remote Destination: Remote Destination Profile + Line Calling Search Space System Remote Access Block Numbers: update as needed Transfer on-hook enabled: True Display Original Calling Number on Transfer from Unity: True Max Forward unregistered hops to DN: 1 Allow peer to preserve h323 calls: True/*need to add appropriate configuration on h323*/ Another service parameter I have seen people modify is the stop routing on unallocated number parameter. People mod this to allow calls to hunt around a H323 gateway that has a PRI which is down. I didn't use this method because I think it is the wrong approach to fixing that problem. I leveraged the IOS config command: no dial-peer out status pots HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production
Re: [OSL | CCIE_Voice] which route pattern discard digits includes even # dialing
Actually, you could use the pattern 9.011![0-9#] to cover both dialing scenarios with one pattern. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 11, 2013, at 9:42 PM, Justin Carney wrote: You need both patterns. The first step is matching a route, then digit manipulation is applied. The two patterns are used to match international calls with variable length digits both with and without dialing #. You need the one with # when the question states something like give users the ability to avoid interdigit timeout. This pattern will only match when user dials the # and you could use predot trailing # for ddi. The pattern without # will only match if a user does not dial # and t302 timer expires. The only time you can get away with only one pattern is if the question says you do NOT need to give users a way to avoid interdigit timeout. My strategy is to always use both patterns unless the question says prevent users from avoiding interdigit timeout in which case this extra config with the # pattern would cause you to lose those points. On Oct 11, 2013 12:59 PM, virajith vir...@rediffmail.com wrote: Hello, I wanted to know which discard digits option in route pattern includes both 9011.! and 9011!# dialing . So that only 1 route pattern is created instead of 2 for dialing without and with #. -Vir Get your own FREE website, FREE domain FREE mobile app with Company email. Know More ___ 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] End User to Device Association
Bill, I put a bunch of these queries on my blog site (see sig. block). I used SQL queries to verify lab configs a fair amount. They are handy and quick. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 12, 2013, at 5:13 PM, Bill Lake wrote: Does anyone know a good source for these type of searches or a good way to create them? Bill Sent from my iPhone On Oct 12, 2013, at 11:40 AM, Martin Sloan martinsloa...@gmail.com wrote: Hey Ryan, So the query isn't super simple but it's definitely something you could memorize for a quick look at user/device associations. The table that holds the relationship between the user and device is the enduserdevicemap table but all the records for user and device are references to the pkid's of the primary table so you have to join those in, the enduser and device table, to get the friendly names. Here's the query: run sql select enduser.userid,device.name from enduserdevicemap inner join enduser on enduser.pkid = enduserdevicemap.fkenduser inner join device on device.pkid = enduserdevicemap.fkdevice The results would give you something like this: userid name == === SBPH2 SEP1234567891236 HQPH2 SEP123456789125 SBPH1 SEP123456789124 HQPH1 SEP123456789123 HTH Marty On Sat, Oct 12, 2013 at 11:27 AM, Martin Sloan martinsloa...@gmail.com wrote: You could do a quick SQL query from the pub cli. I can't recall the table off hand but I will check it out when I get back to my computer. On Oct 12, 2013, at 10:47 AM, Ryan Maxam ryan.ma...@gmail.com wrote: Is there a quick and easy way to see which device an End User is associated with? Without having to run a report or going into the individual End User configuration. It is not an offered search under Find and List End User's. Thanks. Ryan Maxam Sent from my iPad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Route Patterns
Not sure what you would like explained. 9011.! and 9011.!# are for international dialing from North America. Both patterns include a dot (.) which will allow you to apply a digit transform action of pre-dot, if you want. The exclamation (!) is a wild card and instructs the digit analysis process to continue accepting dialed digits (0-9,*). The hash is treated as a termination character. Since ! says I'll keep taking digits until inter-digit timeout expires you sometimes want to provide a way for users to expedite the digit analysis. The # gets you there and you need a route pattern to handle that digit. The 9.1(2-9)xx[2-9]xx is an invalid pattern. Parens are not acceptable characters for route patterns. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 10, 2013, at 9:30 PM, Anthony Nwachukwu wrote: Hi, Can someone explain the Route Patterns below. 9.1(2-9)XX[2-9]XX 9011.! 9011.!# 9.1(2-9)XX[2-9]XX Cheers ___ 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] ringlist
To confirm that the file is available via TFTP do the following: Open web browser URL http://PubIP:6970/Ringlist.xml Do you see the file? Is it accurate? If yes and the phones are still complaining then check TFTP config on phones and see which TFTP host is primary. Adjust as needed. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Jul 22, 2013, at 1:00 AM, Karen Johnson wrote: i have added Ringlist to PUB tftp and restart the TFTP service, even PUB callmanager and when i show file view tftp Ringlist.xml file is there but when i press thru phone, it said can't find it . any missing ? K ___ 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] Transcoding Meeting Place
My take: The MP IP gateway needs the MRGL assigned to it. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Jul 23, 2013, at 3:11 AM, Schmitz, Daniel wrote: Hi all, a customer has the following setup. - Across the MPLS, G.729 should be used - Meeting Place is just configured for High Capacity (G.711) only - CUCM has an IOS transcoder with the following configuration dspfarm profile 3 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 6 associate application SCCP For some reason it is not possible to call from China to the Meeting Place, as soon as I allow G.711 via the SIP trunk everything works just fine, but with G.729 the call cannot be established. Which component needs the correct MRGL for the transcoding? image002.png Do I miss anything? Regards Daniel Senior IT-Specialist Team leader Network Communication Services Managed Cloud Services DIDAS Business Services GmbH | Bernerstr. 38 | 60437 Frankfurt Tel.: +49 69-95022-327 | Fax: +49 69-95022-77327 | Mobil: +49 172-525 2383 Mail: daniel.schm...@didas.de | Web: www.didas.de AG Düsseldorf HRB 63231 | USt-ID-Nr.: DE811548338 | Geschäftsführer: Dirk Kiefer Der Inhalt dieser E-Mail ist vertraulich und ausschließlich für den bezeichneten Adressaten bestimmt. Wenn Sie nicht der vorgesehene Adressat dieser E-Mail oder dessen Vertreter sein sollten, so beachten Sie bitte, dass jede Form der Kenntnisnahme, Veröffentlichung, Vervielfältigung oder Weitergabe des Inhalts dieser E-Mail unzulässig ist. Wir bitten Sie, sich in diesem Fall mit dem Absender der E-Mail in Verbindung zu setzen. Wir möchten Sie außerdem darauf hinweisen, dass die Kommunikation per E-Mail über das Internet unsicher ist, da für unberechtigte Dritte grundsätzlich die Möglichkeit der Kenntnisnahme und Manipulation besteht. The information contained in this e-mail is confidential. It is intended solely for the addressee. Access to this e-mail by anyone else is unauthorized. If you are not the intended recipient, any form of disclosure, reproduction, distribution or any action taken or refrained from in reliance on it, is prohibited and may be unlawful. Please notify the sender immediately. We also like to inform you that communication via e-mail over the internet is insecure because third parties may have the possibility to access and manipulate e-mails.Hi ___ 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] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration
Daniel, First, let me touch on the most important point you have brought up: Honestly, it's really great to see how everyone came together, spoke loudly, and made a difference. Even if the decision isn't reversed, the past few days really demonstrated how dedicated and passionate everyone is to their field and hard-earned certifications. Absolutely outstanding! Everyone has done a great job of keeping this topic on Cisco's radar. Keep it up! Second, I am at the PBT in SJC. Rowan mentioned the CCIE but it was just an acknowledgement that our message has been received. The right thing comment was actually made by another person (name escapes me). Rowan may have echoed the sentiment but no guarantees/promises were made. This isn't bad news. All feedback we have received thus far has been a positive trend in the right direction. Even if it has been incremental. So, there is a light at the end of the tunnel. Cisco will want to close this down as fast as possible. Whatever the answer is, we can all be satisfied that we refused to roll over on this. Again, the volume of support within the community is quite impressive. Don't back off the throttle now. Keep pushing, stay on message, and (as always) be polite. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On Jun 5, 2013, at 2:14 PM, Daniel Pagan dpa...@fidelus.com wrote: Folks: I'm sure many of you are already aware of this news, but one of the heads of Collaboration @ Cisco (I believe Rowan) has made a brief statement today saying, we will do the right thing for everyone. There's also some rumors floating around the Cisco Partner collaboration forums that a manager of the Cisco UC Master program confirmed CCIE Voice will be grandfathered into the CCIE Collab track. I don't believe this is confirmed so please don't hold me to it, but feel free to visit the threads for updated information. Lastly, a community moderator confirmed that Cisco is reviewing all the concerns being posted. https://learningnetwork.cisco.com/thread/56611?start=45tstart=0 https://communities.cisco.com/thread/35337?tstart=0 Honestly, it's really great to see how everyone came together, spoke loudly, and made a difference. Even if the decision isn't reversed, the past few days really demonstrated how dedicated and passionate everyone is to their field and hard-earned certifications. I'm beginning to see a light at the end of the tunnel. Daniel Pagan, CCIE # 25689 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of michael.se...@compucom.com Sent: Monday, June 03, 2013 1:03 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration Importance: High If you haven't already done so please sign this petition: http://chn.ge/17A0zXE Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems ___ 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] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration
FYI. The CCIE WINS its voice back! http://bit.ly/11ise6g Great job to everyone on this list for getting together and pushing for change. Daniel said it best! -Bil -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On Jun 5, 2013, at 2:14 PM, Daniel Pagan dpa...@fidelus.com wrote: Folks: I'm sure many of you are already aware of this news, but one of the heads of Collaboration @ Cisco (I believe Rowan) has made a brief statement today saying, we will do the right thing for everyone. There's also some rumors floating around the Cisco Partner collaboration forums that a manager of the Cisco UC Master program confirmed CCIE Voice will be grandfathered into the CCIE Collab track. I don't believe this is confirmed so please don't hold me to it, but feel free to visit the threads for updated information. Lastly, a community moderator confirmed that Cisco is reviewing all the concerns being posted. https://learningnetwork.cisco.com/thread/56611?start=45tstart=0 https://communities.cisco.com/thread/35337?tstart=0 Honestly, it's really great to see how everyone came together, spoke loudly, and made a difference. Even if the decision isn't reversed, the past few days really demonstrated how dedicated and passionate everyone is to their field and hard-earned certifications. I'm beginning to see a light at the end of the tunnel. Daniel Pagan, CCIE # 25689 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of michael.se...@compucom.com Sent: Monday, June 03, 2013 1:03 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration Importance: High If you haven't already done so please sign this petition: http://chn.ge/17A0zXE Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems ___ 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] CCIE Collaberation
It just means that the CCIE Voice is considered to be a separate IE from Collaboration. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On Jun 3, 2013, at 2:09 PM, Leslie Meade leslie.me...@lvs1.com wrote: Wonder what this means…. http://ciscocert.force.com/english/articles/Article/CCIE-Collaboration-Certification-CCIE-Voice-can-be-recertified-EN?retURL=%2Fenglish%2Fapex%2Finstantanswers%3FproductCategory%3DCCIE_collaborationpopup=false The CCIE Collaboration certification does not directly affect current CCIE Voice certification holders. Current CCIE Voice holders will be able to recertify by passing any CCIE exam including the new CCIE Collaboration written or lab exams. The CCIE Collaboration certification provides new career opportunities for CCIE Voice certification holders Leslie Meade image001.jpg Bach Information Technology CCNA CCVP CCIE Voice 38727 Network Consultant .. Mobile:778.228.4339 | Main: 604.676.5239 Email: leslie.me...@lvs1.com image002.jpg image003.jpg image004.jpg image005.jpg image006.jpgimage007.gif www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original 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 ___ 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] CCIE Collaboration officially announced
My response to Cisco via my blog. Tweeted to @LearningatCisco. Just more line
noise for them to review and/or toss.
CCIE Needs its Voice Back http://bit.ly/138dBGS #FixCCIEVoice
--
William Bell, CCIE #38914
blog: http://ucguerrilla.com
Follow me on twitter @ucguerrilla
On May 31, 2013, at 4:45 AM, Ken Wyan kew...@gmail.com wrote:
There's a serious fault with this new announcement.
We can't say this new blueprint as CCIE Collaboration it should be CCIE
Voice version 4. CCIE Voice track should be allowed to continue with this new
v4 blueprint.
If Cisco want to add another CCIE Collaboration Track they have to add
additional products such as WebEx Server , VCS , Telepresence MSE ,
Telepresence Multipoint Switch , Telepresence Server , Telepresence Manager ,
TMS suitable endpoints.
Then current CCIE Voice guys will be more than happy to complete their
Collaboration certification become dual CCIEs. It will be definitely a
career advancement path for them.
If Cisco can call this new blueprint as CCIE Collaboration ; why can't
they call all current voice CCIE's as Collaboration CCIEs.?
Ken
On Fri, May 31, 2013 at 2:33 PM, Mohammed Al-Assadi m_alass...@hotmail.com
wrote:
Ben e-mail
be...@cisco.com
From: brian.sch...@vitalsite.com
To: rrcr...@yahoo.com; leslie.me...@lvs1.com
Date: Thu, 30 May 2013 13:29:14 +
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
It is Ben Ng. Found his linked in profile below which describes his
position in Cisco.
www.linkedin.com/pub/ben-ng/3/509/940
Profile on the Cisco site.
https://www.cisco.com/web/learning/le31/communities/netpro/bios/benng.html
Anyone have better contact info to send him respectful and thoughtful
arguments on this?
Brian
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm
Sent: Wednesday, May 29, 2013 11:11 AM
To: Leslie Meade
Cc: ccie_voice@onlinestudylist.com; vma...@ipexpert.com
Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Ben Ng comes to mind
On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote:
The question is... what if anything can we do ?
Where would we start..
Original message
From: Mark Holloway m...@markholloway.com
Date:
To: Bill Lake whl...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi
vma...@ipexpert.com
Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially
announced
Granted we all know that taking any CCIE Written will allow us to remain
CCIE's even if Voice is retired, but I think the frustration is Voice and
Collaboration are not THAT far apart and no matter how you look at it, it
all falls under Cisco Unified Communications, which is what the name of
the new CCIE really should be anyway. The core of the Voice blueprint is
still there. The Collaboration equipment list looks like a refresh of
current products, not a forklift of one technology replacing another.
In my opinion this was too harsh of a move to retire Voice and start over
again with Collaboration. There are too many similarities between the two.
On May 29, 2013, at 7:10 AM, Bill Lake
whl...@gmail.commailto:whl...@gmail.com wrote:
Ranting about it won't change anything. I read on line that when they
retire your CCIE, you can still renew by passing a CCIE level written or
lab. If this is true then you do not loose your CCIE just the voice tag.
That seems to be a difficult pill to swallow but it would not be the
first from my reading. Storage had this happen earlier this year as have
several others. See here with the snippet. Now the second is a wiki so we
would want official confirmation.
https://learningnetwork.cisco.com/docs/DOC-17226
http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_
tracks
Retired CCIE tracks
Some previously awarded CCIE specializations have been retired by Cisco.
These are:
* WAN Switching CCIE (Essentially a specialisation focusing on the
IGX/BPX switch products, which had been acquired as part of the
StrataComhttp://en.wikipedia.org/wiki/StrataCom acquisition)
* ISP Dial CCIE
* SNA/IP Integration CCIE (aka CCIE Blue)
* Design CCIE (NOTE: The CCIE Design and CCDE are completely different
design tests in format and subjects examined)
People who hold these now-retired certifications can remain CCIEs,
provided they continue to take recertification exams. They now hold the
title CCIE, rather than CCIE Security, or some other specialization.
So if we can get official confirmation that we won't be stripped of CCIE
if you pass the voice lab, it might
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
That is a fair comment. On re-read that paragraph is unclear and implies
insider knowledge. I will edit accordingly. FYI, I was basis the content on
information exchanged with people involved in the IE program but not in Cisco.
I should have had my legal team review it. Just kidding. I don't have a legal
team.
Thanks,
Bill
--
William Bell, CCIE #38914
blog: http://ucguerrilla.com
twitter: @ucguerrilla
On May 31, 2013, at 2:27 PM, Somphol Boonjing wrote:
Hi Bill,
Great post as always.
From you post, Oh, just in case you missed it. This is a marketing
decision. The decision to kick out the voice IEs was made by someone (or some
group) that has no freakin' idea how much time, energy, money, and effort
people put into getting an IE, any IE. Well, maybe they know about the money
bit because they have to be aware that the Voice IE had a higher average
attempt rate
I am wondering about this bit concerning marketing. I trust you have a
credible source for this. I don't doubt that the name changing stuff is for
brand and marketing reason. I am not so sure that the real reason to
consciously not providing the smooth transition for the current CCIE Voice
holder is because of the marketing reason. (**Note: I must stress that I did
not use the term Marketing team)I don't think it is in the Marketing
team's interest, one way or another, whether a grandfather status is given.
I think it is **much** easier to adopt the traditional method of
grandfathering the cert, than to go against that tradition.
It could be as simple as an oversight, but if it is deliberate, then I am
curious to see what factor contribute the most to the final decision.I
don't think **money** from test taker is a serious part of it. ($6 - $10
millions is not significant enough for the company of that size)
Any insider viewpoint would be much welcome!
Regards,
--Somphol.
--Somphol
On Sat, Jun 1, 2013 at 2:07 AM, William Bell b...@ucguerrilla.com wrote:
My response to Cisco via my blog. Tweeted to @LearningatCisco. Just more line
noise for them to review and/or toss.
CCIE Needs its Voice Back http://bit.ly/138dBGS #FixCCIEVoice
--
William Bell, CCIE #38914
blog: http://ucguerrilla.com
Follow me on twitter @ucguerrilla
On May 31, 2013, at 4:45 AM, Ken Wyan kew...@gmail.com wrote:
There's a serious fault with this new announcement.
We can't say this new blueprint as CCIE Collaboration it should be CCIE
Voice version 4. CCIE Voice track should be allowed to continue with this
new v4 blueprint.
If Cisco want to add another CCIE Collaboration Track they have to add
additional products such as WebEx Server , VCS , Telepresence MSE ,
Telepresence Multipoint Switch , Telepresence Server , Telepresence Manager
, TMS suitable endpoints.
Then current CCIE Voice guys will be more than happy to complete their
Collaboration certification become dual CCIEs. It will be definitely a
career advancement path for them.
If Cisco can call this new blueprint as CCIE Collaboration ; why can't
they call all current voice CCIE's as Collaboration CCIEs.?
Ken
On Fri, May 31, 2013 at 2:33 PM, Mohammed Al-Assadi m_alass...@hotmail.com
wrote:
Ben e-mail
be...@cisco.com
From: brian.sch...@vitalsite.com
To: rrcr...@yahoo.com; leslie.me...@lvs1.com
Date: Thu, 30 May 2013 13:29:14 +
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
It is Ben Ng. Found his linked in profile below which describes his
position in Cisco.
www.linkedin.com/pub/ben-ng/3/509/940
Profile on the Cisco site.
https://www.cisco.com/web/learning/le31/communities/netpro/bios/benng.html
Anyone have better contact info to send him respectful and thoughtful
arguments on this?
Brian
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm
Sent: Wednesday, May 29, 2013 11:11 AM
To: Leslie Meade
Cc: ccie_voice@onlinestudylist.com; vma...@ipexpert.com
Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Ben Ng comes to mind
On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote:
The question is... what if anything can we do ?
Where would we start..
Original message
From: Mark Holloway m...@markholloway.com
Date:
To: Bill Lake whl...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi
vma...@ipexpert.com
Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially
announced
Granted we all know that taking any CCIE Written will allow us to remain
CCIE's even if Voice is retired, but I think the frustration is Voice
and Collaboration are not THAT far apart and no matter how you look
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
When you have a group of people that share an opinion, you need to organize that group of people so that they can speak as one voice. It is called Unified Communications for a reason! The key is to have this group opinion communicated across multiple mediums in a consistent and persistent manner. Basically, you have to market your message. Twitter, FB, and the Cisco Communities are good target mediums if you want to get Cisco's attention. Finding out who is in charge of the IE Voice/Collaboration program and getting their email is another medium. Though, the recipient of said email bomb won't look on that with favorable eyes and it may be counterproductive. Bitching for the sake of bitching won't work. You also have to make sure your argument is one that has a chance of appealing to the other party's willingness or ability to make a compromise. For instance, bitching at Cisco and saying they should rethink retiring the IE voice and grandfather us in may not work. However, launching a campaign to convince them that there should be an alternate path for the IE voice to upgrade their IE may provide a more workable compromise. Thus far I have spoken about organizing our complaints to get attention and putting out a message that provides a reasonable and workable compromise. Cisco has and will listen to that messaging. It has a chance if you say it loud and often. The whole squeaky wheel thing. If you had a way to show that this move costs Cisco money then you would have an even more effective weapon. This is a little harder to conceptualize and even harder to convince everyone to do what would need to be done. -Bil -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On May 29, 2013, at 10:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto:whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataComhttp://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.commailto:m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Agreed. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On May 29, 2013, at 11:29 AM, Bill Lake whl...@gmail.com wrote: I agree that in retiring the exam and requiring that you retake the lab portion again is incomprehensible. They can't tell me that the RS hasn't changed as much or more over its lifetime. It is still the same but they did not retire it (well maybe that is the plan, retire them all and make you earn new) so if you got your RS 10 years or 10 days ago you are CCIE RS. You can easily say the same for others but you get the idea. I think that this is marketing and even so they could have easily done exactly what they did with CCVP to CCNP Voice. When you renew, you do so by passing the CCIE Collaboration written exam (which they make more like the others with some interactive tasks) and you then renew as a CCIE Collaboration. I just think we should stop complaining, organize the CCIE voice community and ask nicely, demand persuasively and argue smartly to get them to change their minds about having to take the lab again to move to CCIE Collaboration. What they have done is weaken in my mind what I strove so hard to earn Bill On 5/29/13, Mark Holloway m...@markholloway.com wrote: Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom acquisition) ISP Dial CCIE SNA/IP Integration CCIE (aka CCIE Blue) Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense whatsoever whether be it from marketing point of view or any other. Why can't big buck makers at Cisco just rename a Cert rather than do something completely rubbish. With just one announcement, they have made many people lose faith in Certification process. I am sure Voice labs will be the most deserted labs until Feb 2014. At the end of day, we can only request Cisco to re-consider this decision. I hope folks concerned collaborate put their suggestions forward on Cisco Support Community direct to Cisco Certification teams so they realize what they are doing is NOT right. I will take some months for us to digest this news. Thanks On Wed, May 29, 2013 at 12:27 PM, Vik Malhi vma
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
They did disable commenting. That's interesting. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 2:24 PM, Martin Sloan wrote: Is it just me or did they disable commenting on the v4 lab topics post? https://learningnetwork.cisco.com/docs/DOC-20804 I wanted to commend William Bell on hitting the nail on the head and put my own 2 cents in. I'm able to place a comment in the equipment list post here: https://learningnetwork.cisco.com/docs/DOC-20804 but not the other. Does anyone have comment options on the exam topics page? On Wed, May 29, 2013 at 12:10 PM, Rrcrumm rrcr...@yahoo.com wrote: Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto:whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataComhttp://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.commailto:m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense whatsoever whether be it from marketing point of view or any other. Why can't big buck makers at Cisco just rename a Cert rather than do something completely rubbish. With just one announcement, they have made many people lose faith in Certification process. I am sure Voice labs will be the most deserted labs until Feb 2014. At the end of day, we can only request Cisco to re-consider this decision. I hope folks concerned collaborate put their suggestions forward on Cisco Support Community direct to Cisco Certification teams so they realize what they are doing is NOT right. I will take some months for us to digest this news. Thanks
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Yeah, it is kind of ironic that the collaboration feature is disabled in a collaborative community article on the IE Collaboration cert. BTW, the twitter handle you can use to get the message to a broader audience is @LearningAtCisco. I think it is a good idea to direct messages to these folks. Of course I recommend being polite. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 4:20 PM, Martin Sloan wrote: I thought so too. I could see if it was getting obnoxious but all of the comments were pretty professional. On Wed, May 29, 2013 at 4:12 PM, William Bell b...@ucguerrilla.com wrote: They did disable commenting. That's interesting. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 2:24 PM, Martin Sloan wrote: Is it just me or did they disable commenting on the v4 lab topics post? https://learningnetwork.cisco.com/docs/DOC-20804 I wanted to commend William Bell on hitting the nail on the head and put my own 2 cents in. I'm able to place a comment in the equipment list post here: https://learningnetwork.cisco.com/docs/DOC-20804 but not the other. Does anyone have comment options on the exam topics page? On Wed, May 29, 2013 at 12:10 PM, Rrcrumm rrcr...@yahoo.com wrote: Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto:whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataComhttp://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.commailto:m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense
Re: [OSL | CCIE_Voice] Does cisco repeat labs
Yup. The proctor picks your lab. They look at all of the labs you have taken. I am sure there are some rules like they can't give you the same lab back to back but I got the sense it was at the proctor's discretion. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 5:20 PM, Josh Petro wrote: I've always been told your lab selection is up to the proctor. Moreover, they generally pick labs you have not done before. Best of luck and God bless! On May 29, 2013 5:07 PM, Ajay Viswanath ajayviswan...@yahoo.co.in wrote: 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 ___ 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] CCIE Collaboration officially announced
You can also post to Facebook (https://www.facebook.com/Cisco.Learning) or this thread on the Cisco learning community (https://learningnetwork.cisco.com/thread/56590?tstart=0). The learning community also has Google+ and other social media accounts. There is one team that manages most of the social media and another team that manages the learning community. I am sure they will listen and bubble up the input they receive. Volume counts. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 8:30 PM, Somphol Boonjing wrote: Bill, Thanks for the twitter account, I just send out my feedback too. I think another point that needs clarification I think is what guidelines is used to EOL the certification? Without transparency, the risk to have sudden and unexplained EOL announcement of any CCIE track can be very real. Without the guideline, no one would know in advance whether CCIE Collaboration will be retired and the new track is created as CCIE Synergy when CUCM is upgraded to version 14.1 in 3-5 years?What makes us think this is a one off? What if the guidelines clearly stated that if the material changes for 30%, the track will be retired and a new track will be created? (What if this is an implicit guideline on their port?)Will people still think it is worth the effort? The lack of transparency and guidelines on how the decision is reached make the CCIE cert easily one of the most risky investment of time and commitment. It is hard, required high level of commitment and can be short-lived without proper communication upfront. By the way, I don't think Cisco will lose money over this. Imaging a few year from now, CCIE Voice will be faded away and will totally be useless. All the job ads will be for CCIE Collaboration. Do you think more and more former CCIE Voice will reset the CCIE Collaboration lab knowing that the new material is not that much anyway. The exam however is tricky and picky, and on average it take some thing like 3.xx times to pass it. (OK, I will discount it to 2 attempts for former CCIE Voice) So assuming 50% of exiting CCIE Voice holders -- appx 1500 - 2000 of them takes this path, Cisco wouldn't be losing revenue do they? 2000 (50% of current CCIE Voice Holder) x $1500 (Lab cost) x 2 (On average, two attempts) = $6,000,000.- Mind you that $6 million is nothing for the company of its size, but the point is they won't be losing money. --Somphol On Thu, May 30, 2013 at 6:31 AM, William Bell b...@ucguerrilla.com wrote: Yeah, it is kind of ironic that the collaboration feature is disabled in a collaborative community article on the IE Collaboration cert. BTW, the twitter handle you can use to get the message to a broader audience is @LearningAtCisco. I think it is a good idea to direct messages to these folks. Of course I recommend being polite. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 4:20 PM, Martin Sloan wrote: I thought so too. I could see if it was getting obnoxious but all of the comments were pretty professional. On Wed, May 29, 2013 at 4:12 PM, William Bell b...@ucguerrilla.com wrote: They did disable commenting. That's interesting. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 2:24 PM, Martin Sloan wrote: Is it just me or did they disable commenting on the v4 lab topics post? https://learningnetwork.cisco.com/docs/DOC-20804 I wanted to commend William Bell on hitting the nail on the head and put my own 2 cents in. I'm able to place a comment in the equipment list post here: https://learningnetwork.cisco.com/docs/DOC-20804 but not the other. Does anyone have comment options on the exam topics page? On Wed, May 29, 2013 at 12:10 PM, Rrcrumm rrcr...@yahoo.com wrote: Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology
Re: [OSL | CCIE_Voice] cups best practice
Well, I can't tell you why you got 0 points. When I provision CUPS, I do the following: CUCM Check to ensure Application Server exists (it should, but check) Add SIP Security Profile for CUPS, with appropriate options Add SIP trunk for CUPS License capabilities for end users Associate end users to DNs (including all shared lines) Associate devices to end users Add end users to End Users and Standard CTI user groups If doing IPPM Add service URL associate with phone(s) Add IPPM user in CUCM Add CUPC if required CUPS Update Cluster Topology so CUPS node uses IP address and not name Update service parameters (UC Proxy, domain setting) Check service parameters (look for anything that isn't default) Check CUCM Publisher status (ensure all green) Add CUCM gateway (technically, not required since we are using Publish mode but I did it anyway) Add ACLs for incoming (ALL,ALL) Activate/Start services IPPM - customize service account user ID and password to match CUCM CTI - associate users General settings: TFTP servers (make sure they match what you are using for phones) VM Profile VM Server - CUC Mailstore is NOT necessary with this version and CUC VM Pilot number Unity Connection: Make sure you mod the CoS to allow UPC to do its thing Log into CUPS as user and add contacts manually When testing: Ensure contact list is built correctly Ensure you have presence status updates for onhook/offhook events Ensure you receive presence status changes when client toggles status If soft phone, place and receive calls If RPC, place and receive calls If VM, then leave a message, ensure notification is received, playback the message (though you won't have audio) IM between clients, both ways If using IPPM, log in, test, and remain logged in If CUPC, log in, test, and remain logged in If you have a requirement to send messages then do so and leave the messages on-screen (all clients) If lab guide gives you screen shots, pay attention to every little detail of the screen shot and make sure your screen looks identical I never had issues getting points on CUPS and I always followed the same procedure. I have spoken to several people who have said that they configured everything, everything worked, but they still got 0 points. I trust that they are right, everything was configured and working. So, that leave two things. One, if they give you a screen shot of what they want, check everything and make sure you mimic the screenshot. Two, pay attention to any clues as to how the clients should be left when you are done with the exam. For instance, shutting down the clients when not told to do so is probably a bad idea. If they say send a broadcast then I would do so and leave it on the screen. Things like that. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On May 23, 2013, at 11:39 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: tks Ccieing, i have included steps u mention below and also some service parameter. Also soft and desk phone was login and send message each other, but i still got 0 mark. any advice from people who got full mark in this section please ? From: CCIEing aboaz...@gmail.com To: Karen Johnson karen.johnson...@yahoo.ca Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Thursday, May 23, 2013 5:15:54 PM Subject: Re: [OSL | CCIE_Voice] cups best practice I do not set to the exam yet.. but here are below the steps I use to configure my CUPS: CUPC Configuration (Softphone) Form CUPS Side : Application -- Settings Application -- CUCP -- add Voice Mail Server Application -- CUCP -- add Mail Store (default 143) Application -- CUPC -- add Voicemail Profile , then add users to this profile Application -- CUCP -- add CTI Gateway (should be created automatic when start services ) Application -- CUCP -- add CTI Gateway Profile Form CUCM Side : Add CUPC soft phone: Device -- add Cisco Unified Personal Communicator with the name UPCUSERNAME Desk phone Configuration CUCM -- Associate device to end user (End user configuration page-- add the device in the Device association list ) CUCM -- Specify Primary extension for the end user (the same Ext on his device) CUCM -- Add the user to the groups (Standard CTI enables CUCM end user) CUPS -- Assign user to the CTI Gateway profile (the one that relative to the user's phone DP) Login/logout from the CUPC and test. On Thu, May 23, 2013 at 4:54 AM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi experts, can anyone share how to config desk mode and soft mode best practice for CUPS in exam? I can't figure out why did not got point for cups even it is working fine. tks ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http
Re: [OSL | CCIE_Voice] proctorlabs replication
When you say ...and CLI also look good... can you be more specific? What command did you use in the CLI to check replication? -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 23, 2013, at 1:21 PM, Karen Johnson wrote: hi all, I used auto phone register in proctorlabs. When UCM group start with SUB, phone never registered. But when I move PUB to 1st Server in Group, phone registered fine. And I also checked in Cisco CallManager Reporting for DB summary: replication look good 2 and CLI also look good any idea what wrong and what is command to check in this case? tks ___ 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] tranfer to Voicemail button for CME - CUE
I haven't tried Kirill's method. My method is similar to Bill's approach. Sample: ephone-dn 5 number *4...$ call-forward all 4600 ! voice translation-rule 1 r 1 /.+\(\)$/ /\1/ voice translation-profile strip-rdnis tr redirect-called 1 ! dial-p v 81010 voip destination-p 4600 session proto sipv2 session target ipv4:x.x.x.xcue dtmf-r sip-not no vad codec g711u translation-p out strip-rdnis ! -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 22, 2013, at 11:08 AM, Kirill Groshev wrote: Hi, ephone-dn 5 number *4001 call-forward all 4001 Since the busy trigger the line is 1, the call will be forwarded to VM of 4001, otherwise it's forwarded to VM of extension *4001, which is not configured. Hope that helps. K. On Wednesday, May 22, 2013, sanity insanity wrote: hi All, I have the following config ephone-dn 3 number 4001 label 4001 description +85224044001 name +85224044001 call-forward busy 4220 call-forward noan 4220 timeout 20 ! ephone-dn 5 number *4001 call-forward all 4220 ! ephone 2 mac-address 1089.CF01.7C99 ephone-template 1 speed-dial 4 *4001 label Xfer-to-VM button 1:3 2:4 When I press *4001 it forwards to voicemail but unity express says that a voicemail box is not configured for this extension although I have a voicemail box configured for extension 4001. What am I missing? Regards, MJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] [DHCP Static Binding Origin FIle]
My take on it: http://www.ucguerrilla.com/2013/05/ccie-voice-tactical-dealing-with-ios.html -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 16, 2013, at 4:56 AM, ie ravindra wrote: Dear All Experts, Where we can obatin DHCP static origin file template officially. Thanks, Ravi. ___ 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] shortcut command
I use sh run | s ephone 1 for example. The trick is to have two spaces after 'ephone'. This is because the config line literally has two spaces. Same for ephone-dn. On Tuesday, May 14, 2013, Dharambir kumar varma wrote: Hi some time i need to see only particular ephone setting/or ephone-dn setting on CME.is there any shortcut command like show ephone | like that ...on CME please share.. -- 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 ___ 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] ssh client
That is a good question actually. In my home lab I was using SecureCRT 6 (which is not the version in the real lab). In CRT 6, you can edit the session settings by: 1. OptionsGlobal Options 2. General...Default Session, select Edit Default Settings 3. Go under TerminalEmulation and modify the Scrollback buffer setting I don't know if it is the same for previous versions. Most likely it isn't the same. I never bothered to check because I just modified the buffer on the IOS device and logged to buffer. I opted to go that path because I didn't want to spend time dorking around with exploration in the real lab. But that's just me. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 14, 2013, at 9:16 PM, CCIEing wrote: So how to increase the buffer :) Sent from my iPhone On May 15, 2013, at 1:47 AM, Barrera, Hugo hugo.barr...@nexusis.com wrote: This dang client cost me about 20 min in the Lab because I didn't know how to increase the buffer. Regards, Hugo On May 14, 2013, at 3:32 PM, Bill whl...@gmail.com wrote: I think it is an old version of secure CRT and not one easily found on the web. I think something like version 3 or 4 but I really did not worry about that, I use the current version and it works similar but don't expect much more that very basic interface Sent from my iPad On May 14, 2013, at 5:17 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I think it should be v2 however I am not quite sure On 14 May 2013 15:07, Barrera, Hugo hugo.barr...@nexusis.com wrote: Anybody know what version of ssh client that is in the real lab on the CUPC Test PC? - Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] Isdn channel order
You have two config options to keep in mind: 1. MGCP PRI in CUCM. Channel selection order controlled by CCM Manager backhaul and, therefore, configured in CUCM 2. PRI controlled by IOS (i.e. the channel isn't back hauled). The IOS config controls channel selection. With the CUCM back-hauled channel the options are: Top Down or Bottom Up. The lowest numbered channel is the TOP and the highest numbered channel is the BOTTOM. Think of them visually as: 1 (top) 2 3 . n (bottom) So, going from 1 to 2 to 3 your are going DOWN or TOP DOWN. Obviously, BOTTOM UP is the opposite. When the D-channel is NOT back-hauled to a remote call agent (like CUCM) then you are using IOS configs and because Cisco likes to toy with your head the terminology is different. You have the option of specifying whether B-channels are chosen in ascending order or descending order. For this think of the actually b-channel numbers (e.g. 1,2,3) as you go from 1 to 2 to 3 you are ascending (or incrementing). So, ascend in IOS is the same as TOP DOWN in CUCM. Descend is the opposite, obviously. In regards to defaults, when CUCM is the back-haul agent the default is Bottom Up . When using IOS, the default is descending. So, the default behavior in both options is the same. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 14, 2013, at 8:15 PM, Dharambir kumar varma wrote: Hi top to botton isdn channel =? 1 to 31 or 31 to 1 which one is by default. confused -- 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 ___ 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] CUCM upgrade
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 ___ 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] How to configure ringlist for a specific phone in CUCM?
There is a specific config file for each phone, this is true. However, that config file does not contain the ring list. That is a separate config file, as I am sure you are aware. As far as I know the ringlist file is universal. The only way you could specify a custom ringlist for one phone would be to point that phone to a different TFTP server and then have a different Ringlist.xml on that TFTP server (along with all of the other files you would need). -BIll -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 7, 2013, at 2:28 AM, Hesham Abdelkereem wrote: Dear Experts, I would like to know how can i edit the ringlist for a specific phone only and not for all? I believe there is a specific configuration file sepxx.cnf is available somewhere in the CUCM but I don't know how to get hold of it. Please share your ideas. Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] How to configure ringlist for a specific phone in CUCM?
Ok , now for that thing you mentioned below to point to a different TFTP server and then have a different Ringlist.xml do you mean by that for example I make the universal on Publisher and let all phones register to Publisher? No. Phone registration and TFTP are completely separate aspects of the phone integration to CUCM. That said, you could have a different Ringlist.xml on the publisher than you have on the subscriber and you can have DHCP scopes set different Option 150 addresses. Assuming that meets your requirements and doesn't conflict with other requirements. It may also be possible to leverage the TFTP service on an IOS device. But this approach is convoluted. Especially if we are talking about a lab scenario. Now , the question where is the parameter where can I apply an external link for the ringlist.xml? No such parameter exists as far as I know. All phones look for the same basic path and file name for Ringlist.xml. The only difference is introduced by the IP address of the TFTP server. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On May 7, 2013, at 11:58 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi William, Thanks a lot for your great input. Yes I am aware of the universal ringlist.xml which is located at http://cucmip:6970/ringlist.xml. I know how to change and edit that very well for all the phones. Ok , now for that thing you mentioned below to point to a different TFTP server and then have a different Ringlist.xml do you mean by that for example I make the universal on Publisher and let all phones register to Publisher? and make the other ringlist on the subscriber and let that specific phone register with the subscriber likewise I should configure the first option 150 ip for the phone to subscriber and publisher is the second. I think I can let the UCCX publish the ringlist.xml as it has an IIS as webserver but I don't know how to apply this file on that specific phone on which tab or parameter I am able to do that. In Directories menu , I can create a custom Directories.xml and publish it via UCCX server then I apply the link on the service provisioning enterprise parameters. Then I make service provisioning both inernal and external. Now , the question where is the parameter where can I apply an external link for the ringlist.xml? I am sure that it has something to do with the original phone file configuration which can be tweaked for that. Thanks, Hesham On 7 May 2013 05:18, William Bell b...@ucguerrilla.com wrote: There is a specific config file for each phone, this is true. However, that config file does not contain the ring list. That is a separate config file, as I am sure you are aware. As far as I know the ringlist file is universal. The only way you could specify a custom ringlist for one phone would be to point that phone to a different TFTP server and then have a different Ringlist.xml on that TFTP server (along with all of the other files you would need). -BIll -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 7, 2013, at 2:28 AM, Hesham Abdelkereem wrote: Dear Experts, I would like to know how can i edit the ringlist for a specific phone only and not for all? I believe there is a specific configuration file sepxx.cnf is available somewhere in the CUCM but I don't know how to get hold of it. Please share your ideas. Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE CLI/GUI ?
Nic, IMO this is akin to the question of whether one uses ccm-manager config to provision MGCP or not. You can get to the end game along multiple paths. For CUE, I primarily practiced using the GUI. That said, I did practice using CLI as well. People generally say that using the CLI is faster. I agree that when it comes to CME-CUE using the CLI is quicker. However, I don't really see much of a difference when doing CUCM-CUE. I actually think the GUI is much easier with CUCM-CUE (vs. CLI for the same integration). I have heard people argue that they may block your access to the GUI and that you may want to get comfy with the CLI. I'm cool with that and I did familiarize myself with the CLI method but I focused most of my practice labs on using the GUI method. There wasn't enough value (measured in terms of time) in the CLI method for me to focus on it. But we all have to make our own call on that one. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On Apr 29, 2013, at 11:13 AM, Nicolas MICHEL mcl.nico...@gmail.com wrote: Hey Guys. I have currently struggling with CUE integration / installation and configuration. What would you use in the Lab ? CLI or GUi ? Because in the Workbooks, the GUI is used approx all the times ... Just wanted to have your thoughts Thanks for the help Nicolas ___ 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] Blocking 91900 pattern
That is not accurate. The 7965 is a Type-B phone: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/srnd/8x/dialplan.html#wp1135386 Type-B phones support KPML: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/srnd/8x/dialplan.html#wp1043976 -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On Apr 27, 2013, at 12:29 PM, ramyoth...@hotmail.com wrote: Hi Ryan, This is because SCCP phones process dialed numbers in real time digit by digit as you dial whereas Type-A SIP phones send digits when you press # or Dial softkey. There are 2 methods to simulate SIP phones; either using phone models that support KPML or configure SIP dial plans on UCM/UCME. 7965 doesn't support KPML. So, you need to configure SIP dial plans. Thanks, Ramy --- Original Message --- From: Ryan Maxam ryan.ma...@gmail.com Sent: April 27, 2013 6:58 PM To: Online Study (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Blocking 91900 pattern Hello All, I am working on WB1 lab 5.6. I have a 91900 RP that is blocked and annunciator set to precedence level exceeded Problem is when I call 919004522138 from br1 ph 2 (SCCP) I can hear the annunciatot after I dial the 5th digit, but when I call it from hq ph 2 (SIP) my call gets dropped after i dial the 5th digit without hearing the annunciator. (all other calls are working fom hq ph2) I am using all hardware phones (7965) Any thoughts? Thanks for your help Ryan Mail Attachment.txt___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] The best way to restore routers to base configs HOMELAB
This is how I do it: http://www.ucguerrilla.com/2012/08/ccie-v-tip-using-config-replace.html -Bill On Apr 26, 2013, at 8:21 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts? I wonder whats the best and most efficient way to restore all the routers/switches of the homelab to the base configs? Should I just write erase on all devices and then paste the base configs? Please give me some advice Thanks in Advance, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla ___ 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] one button login -- url
You could always memorize it. That is what I did. I memorized a handful of URLs for phone services. It made phone provisioning quicker. I'd at least memorize IPPA and IPPM URLs. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 23, 2013, at 1:01 AM, donny f wrote: hi, what other resource to get the One Button Login url if we can't open SRND /Cisco Doc in lab? 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 ___ 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] transcoding SRST
The answer is: you don't. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 19, 2013, at 11:47 PM, ikizoo hello wrote: Hi All, in Site C SRST mode, why need trancoding?, anyway all the traffic through PRI. thanks -ikizoo ___ 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] file tail activelog
Donny, Like many things, you have to memorize. If you can't do that then do the following: admin:file list activelog * dir car_db dir ccm_db dir cm dir core dir mgetty dir mohprep dir patches dir platform dir sa dir syslog dir tomcat dir count = 11, file count = 0 admin:file list activelog cm/* dir bin dir cdr dir cdr_repository dir log dir report dir tftpdata dir trace dir count = 7, file count = 0 admin:file list activelog cm/trace/* dir ac dir amc dir bps dir capf dir carsch dir ccm dir ccmmib dir ccmservice dir cdp dir cdpmib dir cdragent dir cdrrep dir cef dir cfrt dir cmi dir cms dir ctftp dir cti dir ctlprovider dir dbl dir dhcpmon dir dirsync dir licensing dir lpm dir ris dir rtmtreporter dir syslogmib dir taps dir tct dir count = 29, file count = 0 admin:file list activelog cm/trace/ccm/* dir Proglogs dir sdi dir sdl dir count = 3, file count = 0 admin:file tail activelog cm/trace/ccm/sdi/ recent Don't forget that recent option. If you just want to look at one file then use file list to look at the contents of the SDI (or SDL) folder. Then pick your file. I'd recommend using the detail option for a more informative list. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 18, 2013, at 2:29 PM, donny f wrote: hi all, how to remember this directory in exam ? admin:file tail activelog cm/trace/ccm/sdi what CLI command to show what directory for activelog trace 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 ___ 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] OWLE Lab 1 Task 2.4 No Service Configured
Ramcharan, The only alternate solution is that you create two XML files that you host from the UCCX web server. One XML file is a complete replica of the Directory menu so that your normal phones have a normal experience. The other would display a status message of services disabled. Look at OWLE Lab 4, question 2.2 for a question that is similar. Tinker with it a bit and you can map it into your alternate solution. -Bill On Apr 16, 2013, at 11:16 PM, Ramcharan Arya wrote: Hi Vik, Can you please suggest about alternate solutions because current solution break voicemail service on the phone. Thanks Regards, Ramcharan Arya On Tue, Apr 16, 2013 at 12:10 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Thanks Bill I will check archive for this discussion. On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote: Look for a detailed discussion on this very issue around November 19th 2012 Sent from my iPhone On Apr 16, 2013, at 11:25 AM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Hi Bill I breaks voicemail service on that phone. So what is the alternative solutions for this task.? Thanks, Ramcharan Arya CCIE # 28926 (RS) On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.com wrote: Correct. If you convert a phone to using External URLs only then you will be breaking Messages button. You can apply a URL to the messages button but it doesn't restore the behavior you are looking for because the VM services URL that I have seen used/reference is a XML phone menu object. Which means you get a menu. So, what you are seeing is expected per the requirements of the OWLE lab. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote: Hi, When I use proposed solutions according to solution guide for disabling directory service it works but voicemail service also stop working on the same phone. Can someone please test this and let me know if anyone had similar problem. Thanks Regards, Ramcharan Arya CCIE # 28926 (RS) ___ 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 -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla ___ 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] OWLE Lab 1 Task 2.4 No Service Configured
Right. The order you place the MenuItem child nodes in the XML file will be the order they are presented on the phone. What I do when practicing lab 4 was as follows: CCX 1. RDP to CCX 2. From CCX browse to xml directory URL (copied from Ent Params in CUCM) 3. Copy XML from CCX browser to notepad CUCM / CLI 1. Go to CLI on CUCM and run query admin:run sql select name,urltemplate from telecasterservice nameurltemplate === Missed CallsApplication:Cisco/MissedCalls Received Calls Application:Cisco/ReceivedCalls Placed CallsApplication:Cisco/PlacedCalls Intercom Calls Application:Cisco/IntercomCalls Personal Directory Application:Cisco/PersonalDirectory Corporate Directory Application:Cisco/CorporateDirectory Voicemail Application:Cisco/Voicemail Back to CCX In CCX you will edit the XML you have in notepad. You will create MenuItem child nodes using the above output. The SQL name maps to the XML Name field and the urltemplate maps to the URL field. Build your menu accordingly. You will likely need to create more than one XML file to service from CCX. One for normal phones and one for the phone that is being customized. The nature of the customization will vary based on lab requirement. Save files to c:\inetpub\wwwroot\ Back to CUCM (cli) Now you want to disable the various directories. I wouldn't delete them and I sure as heck wouldn't muddle through the web interface to toggle the services as enabled/disable. So, I use SQL to disable the services: run sql update telecasterservice set enabled = 'f' where name like '%d Calls' or name like '%Directory' The above query will disable: Missed Calls, Received Calls, Placed Calls, Personal Directory, and Corporate Directory. Back to CUCM (web) You want to set the services provisioning from Internal to Both. You can do this from Ent Params, via common phone profile, or directly on each phone (via BAT). I use Ent Params because you will also need to remove the default Directories URL. Restart the phones. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 17, 2013, at 9:25 AM, Ramcharan Arya wrote: Your Bill you are correct OWLE Lab4 solution seems to be more appropriate for this task. So when I create XML file I can keep the same priority of services as show in telecasterservices table to keep directory structure identical. Thanks Regards, Ramcharan Arya CCIE # 28926 (RS) On Wed, Apr 17, 2013 at 6:56 AM, William Bell b...@ucguerrilla.com wrote: Ramcharan, The only alternate solution is that you create two XML files that you host from the UCCX web server. One XML file is a complete replica of the Directory menu so that your normal phones have a normal experience. The other would display a status message of services disabled. Look at OWLE Lab 4, question 2.2 for a question that is similar. Tinker with it a bit and you can map it into your alternate solution. -Bill On Apr 16, 2013, at 11:16 PM, Ramcharan Arya wrote: Hi Vik, Can you please suggest about alternate solutions because current solution break voicemail service on the phone. Thanks Regards, Ramcharan Arya On Tue, Apr 16, 2013 at 12:10 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Thanks Bill I will check archive for this discussion. On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote: Look for a detailed discussion on this very issue around November 19th 2012 Sent from my iPhone On Apr 16, 2013, at 11:25 AM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Hi Bill I breaks voicemail service on that phone. So what is the alternative solutions for this task.? Thanks, Ramcharan Arya CCIE # 28926 (RS) On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.com wrote: Correct. If you convert a phone to using External URLs only then you will be breaking Messages button. You can apply a URL to the messages button but it doesn't restore the behavior you are looking for because the VM services URL that I have seen used/reference is a XML phone menu object. Which means you get a menu. So, what you are seeing is expected per the requirements of the OWLE lab. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote: Hi, When I use proposed solutions according to solution guide for disabling directory service it works but voicemail service also stop working on the same phone. Can someone please test this and let me know if anyone had similar problem. Thanks Regards, Ramcharan Arya CCIE # 28926 (RS) ___ For more information regarding industry leading CCIE Lab training, please visit
Re: [OSL | CCIE_Voice] [CORRECTED] CUE QOS on the Switch
Yeah, on the catalyst switch you won't see the counters increment. That's normal. I would tweak the policy-map: policy-map CUEMAP class CUE-SIGNAL set ip dscp cs3 police 32 8000 exceed-action policed-dscp-transmit ! You also need to modify your policed-dscp map: mls qos map policed-dscp 24 to 0 -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 17, 2013, at 5:59 PM, Jack Kamina wrote: on one of the practice lab the need is to police the signaling packets to and from CUE inbound into the HQ switch to 32 kbps and then remark the DSCP to 0. I built up the config below but dont see any packets matched on the show policy-map interface command. CUE IP is 10.1.6.253 . CUCM IP is 10.10.210.10 (pub) and 10.10.210.11 (sub) .is the access list built correctly? access-list 110 permit tcp host 10.1.6.253 any eq 2748 ! class-map match-all CUE-SIGNAL match access-group 110 ! policy-map CUEMAP class CUE-SIGNAL set dscp af31 bandwidth 20 ! interface Fa0/1/0 description HQ-ROUTER-INTERFACE service-policy input CUEMAP mls qos mls qos map cos 0 8 16 24 32 46 48 56 mls qos map policed-dscp 24 26 to 8 ___ 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] Q1T1 - question
Dominik, You can absolutely verify the the queue and threshold assignment using the command: show mls qos dscp-output-q. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 16, 2013, at 3:51 AM, Dominik Łoniewski wrote: Thanks William, Yes. Cisco documentation says that by default DSCP 46 is in Q1T1. But when you look at the mls configuration (after auto qos done on some interface) it shows that DSCP 46 is in Q1T3 :( Then, when you try to move DSCP 46 from Q1T3 - Q1T1 I can't verify it is really in T1. Dominik 2013/4/16 William Bell b...@ucguerrilla.com Dominik, I think that what you are observing is normal on the 3560 platform. To verify your configurations meet the objective use the show mls qos dscp-output-q command. You probably already knew that. Now, to answer your direct question. I don't have a 3750 but based on some quick doc checks, you have the following defaults to consider. For the 3560 (ref: http://www.cisco.com/en/US/docs/switches/lan/catalyst3560/software/release/12.2_55_se/configuration/guide/swqos.html#wp1163863) By default, when mls qos is enabled, dscp 46 is mapped to Queue 1 / Threshold 1. So, that explains why it wouldn't show up in the config when you do a show run. For the 3750 (ref: http://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_note09186a0080883f9e.shtml#def-config) It appears that the same is true. DSCP 46 is mapped to Queue 1 / Threshold 1 So, IOW I would expect that your experience on your test gear to be the same as your experience on a 3750. At least as far as this particular question is concerned. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 15, 2013, at 6:32 PM, Dominik Łoniewski wrote: Hi, I'm trying to assign DSCP EF traffic to egress Q1 and set the T1 to start dropping when 40% of the buffers are full. What is strange - after putting in the cmd: mls qos srr-queue output dscp-map queue 1 threshold 1 46 mls qos queue-set output 1 threshold 1 40 100 100 100 this first of those lines does not appear int the runn config. DSCP EF is removed from Q1T3 as it should, but I can't find that is really assigned to Q1T1. I've checked this behavior on 3560 12.2.50 SE1 and 12.2.53r. Can someone who has access to 3750 can check how it looks like on 3750 platform. Regards, Dominik ___ 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 -- Pozdrawiam, Dominik Łoniewski ___ 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] MGCP gateway events to pstn.!!!
In regards to Q1, you are accurate in that: - When the GW fails over to the backup call agent, it will send a RSIP (not shown in your trace, perhaps not pertinent to your question) - The call agent will send AUEP messages to the MGCP GW for each endpoint - The MGCP GW responds to AUEP with a 200 OK with the local call identifier (I:) and if I: is blank, there is no call on that particular endpoint - For each AUEP response that had a non-blank I: value, the call agent will send a AUCX message requesting the global call ID and connection mode - The MGCP GW responds accordingly with a 200 message and the appropriate C: and M: values - At this point, the backhaul connection is taking care of sending/receiving a status enquiry and appropriate response from the PSTN In regards to Q2, if you are seeing that call preservation fails on your MGCP device then you will want to check your configuration. Specifically, ensure that you have no mgcp timer receive-rtcp configured. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 16, 2013, at 11:41 AM, sanity insanity wrote: hi Guys, Have not heard back on this... On Thu, Apr 11, 2013 at 9:29 PM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hi Guys, I have a MGCP gateway setup when subscriber fail publisher will work Call 911 while the call is active shut down the cucm sub services. Make sure when Complete to bring up the subscriber services so you can continue with the tasks on the lab. Capture the following -- Backup call agent sends the message to the gateway to check the status of the active call OR -- Backup call agent sends the message to the gateway to check the status of the First call -- Gateway sends the status of active calls to the secondary call agent. -- Back up call agent send a message back to the GW requesting additional information about the call My configuration snip looks like this :- ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host ip address of pub ccm-manager mgcp ccm-manager config server ip address of sub ip address of pub here are the events noted from debug mgcp packets : == **Event1: Backup call-agent sends the message to gw to check the status of active call for port 12 Apr 11 05:26:26.911: MGCP Packet received from ip address of pub:2427--- AUEP 178 S0/SU1/DS1-0/1...@r1.ccievoice.com MGCP 0.1 F: X, A, I --- **Event2:Gateway sends the status message of active calls to the backup call-agent for port 12 Note: Nothing mentioned in I: so it refers no call present in this port Apr 11 05:26:26.915: MGCP Packet sent to ip address of sub:2427--- 200 181 I: X: 0 L: p:10-20, a:PCMU;PCMA;G.nX64, b:64, e:on, es-cci, gc:1, s:on, t:10, r:g, nt:IN ;ATM;LOCAL, v:T;G;D;L;H;ATM;FXR L: p:10-220, a:G.729;G.729a;G.729b, b:8, e:on, es-cci, gc:1, s:on, t:10, r:g, nt **Event3: Backup call-agent requesting additional information about the active call on port 23 Apr 11 05:28:28.936: MGCP Packet received from ip address of pub:2427--- AUCX 213 S0/SU1/DS1-0/2...@r1.ccievoice.com MGCP 0.1 I: 3 F: C, M --- Questions : = 1) Are the above events I have elaborated correct? Please correct me If I am wrong. 2) I tried to test call preservation on this setup . On an active call when I break the connection between MGCP gateway and subsriber, the gateway then registers with the publisher . However instead of the call staying up ...I see the ip phone showing Temp fail and the call to pstn drops . Why? Thanks Regards, MJ ___ 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] OWLE Lab 1 Task 2.4 No Service Configured
Correct. If you convert a phone to using External URLs only then you will be breaking Messages button. You can apply a URL to the messages button but it doesn't restore the behavior you are looking for because the VM services URL that I have seen used/reference is a XML phone menu object. Which means you get a menu. So, what you are seeing is expected per the requirements of the OWLE lab. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote: Hi, When I use proposed solutions according to solution guide for disabling directory service it works but voicemail service also stop working on the same phone. Can someone please test this and let me know if anyone had similar problem. Thanks Regards, Ramcharan Arya CCIE # 28926 (RS) ___ 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] Q1T1 - question
Dominik, I think that what you are observing is normal on the 3560 platform. To verify your configurations meet the objective use the show mls qos dscp-output-q command. You probably already knew that. Now, to answer your direct question. I don't have a 3750 but based on some quick doc checks, you have the following defaults to consider. For the 3560 (ref: http://www.cisco.com/en/US/docs/switches/lan/catalyst3560/software/release/12.2_55_se/configuration/guide/swqos.html#wp1163863) By default, when mls qos is enabled, dscp 46 is mapped to Queue 1 / Threshold 1. So, that explains why it wouldn't show up in the config when you do a show run. For the 3750 (ref: http://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_note09186a0080883f9e.shtml#def-config) It appears that the same is true. DSCP 46 is mapped to Queue 1 / Threshold 1 So, IOW I would expect that your experience on your test gear to be the same as your experience on a 3750. At least as far as this particular question is concerned. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 15, 2013, at 6:32 PM, Dominik Łoniewski wrote: Hi, I'm trying to assign DSCP EF traffic to egress Q1 and set the T1 to start dropping when 40% of the buffers are full. What is strange - after putting in the cmd: mls qos srr-queue output dscp-map queue 1 threshold 1 46 mls qos queue-set output 1 threshold 1 40 100 100 100 this first of those lines does not appear int the runn config. DSCP EF is removed from Q1T3 as it should, but I can't find that is really assigned to Q1T1. I've checked this behavior on 3560 12.2.50 SE1 and 12.2.53r. Can someone who has access to 3750 can check how it looks like on 3750 platform. Regards, Dominik ___ 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] SiteB Phones are not talking IP Address
I'd check: 1. DHCP snooping on the switch (sh ip dhcp snoop) 2. Dot1q trunk on HQ-RTR interface (on the switch). Make sure that all appropriate vlans are allowed on the trunk link and that native VLAN lines up (1 is default). 3. Ensure VLANs are provisioned correctly, assigned to the right interfaces, and active (sh vlan b) 4. Double check scope config on CUCM Pub. Check each parameter. If the above check out then I'd restart the DHCP service on the Pub. If that didn't work, I would do the following on the phone: 1. Settings key 2. **# to unlock 3. Press more softkey when it pops up 4. Press Erase softkey -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 15, 2013, at 8:16 PM, CCIEing wrote: Dear all, I am using proctorlabs racks, My siteB phone are not talking IP addresses from CUCM-PUB DHCP. I have configured the switch port connected to the IP phone with the correct access/voice vlan informaation. I also apply the IP hdcp helper-address command on the voice-vlan interface on the router, and pointed to the IP address of CUCM-PUB. When debuging IP dhcp server events/packets the router show the following messages : Apr 16 00:09:53.946: DHCPD: Finding a relay for client 0100.12d9.78ef.01 on interface Vlan240. Apr 16 00:09:53.946: DHCPD: Seeing if there is an internally specified pool class: Apr 16 00:09:53.946: DHCPD: htype 1 chaddr 0012.d978.ef01 Apr 16 00:09:53.946: DHCPD: remote id 020a0a0ac90110f0 Apr 16 00:09:53.950: DHCPD: circuit id Apr 16 00:09:53.950: DHCPD: there is no pool for 10.10.201.1. Apr 16 00:09:53.950: DHCPD: setting giaddr to 10.10.201.1. Apr 16 00:09:53.950: DHCPD: BOOTREQUEST from 0100.12d9.78ef.01 forwarded to 10.10.210.10. Any Idea Please ! Thanks in advance ___ 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] GKCUBEBackbone - Call Question
The CUCM is waiting for the far end to initiate TCS because that is how it is configured by default. As the document you reference noted, you can uncheck a box to modify this behavior. When you insert a CUBE in the call path then you either need to: 1. Disable the wait for far end TCS option OR 2. Leverage Fast Start by forcing MTP (and adding a MRGL and choosing your fast start codec option) The CUBE doesn't initiate TCS. So, you have to leverage the CM config to accommodate the call scenario. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 14, 2013, at 8:03 PM, Barrera, Hugo wrote: Hi Guy’s, Regarding the attached link, why does CUCM have to wait for the far end TCS before it can send it’s own TCS what is the reason for this? ALSO what are the best debugs to run when troubleshooting this?? https://supportforums.cisco.com/docs/DOC-2529 Hugo ___ 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] MGCP - outbound call - channel order question IPExpert Lab 4A in Prep Workbook
Brian, Bottom Up means that the call agent will instruct the gateway to signal the call on the last B-channel first and then move in sequential order to the first B-channel. So, in your case that would be channel 3, 2, and finally 1. Top Down means that the call agent will instruct the gateway to signal the call on the first B--channel and then move in sequential order to the last B-channel. So, 1 then 2 and finally 3. So, what you are observing is expected behavior. Change the Channel Selection order to Top Down. Then do no mgcp / mgcp on the gateway. You should be good to go. HTH -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 13, 2013, at 7:03 PM, VanBenschoten, Brian wrote: I'm working on IPexpert Lab Prep Workbook - Lab 4A Section 4.3. Asks us to configure router at Branch 1 as MGCP using channels 1-3 and ensure that the first outbound call to the PSTN uses channel 1 I've got the gateway all configured and working. But I wasn’t able to force it to use channel 1 bottom up as the field is referenced in CUCM There are 2 commands in CUCM for this Channel Selection Order Channel IE Type image001.png The channel selection order seems pretty straight forward , our choices are Bottom up, and Top Down I set it to Bottom Up , leaving the Channel IE Type to the default setting of Use number when 1B I'm not using CCM Config and I reset the gateway from CUCM and no mgcp... mcgp on the router Outbound calls kept going out channel 3 The only way I could get this to work properly was to change the Channel IE Type = Slotmap Then outbound calls went out the first channel, which on an ISDN Q931 debug showed as Channel 0. I'm not familiar with the Channel IE Type settings. I've never had to mess with them before. Does anyone have any thoughts or comments? ISDN debugs and config is below I'm using my home lab so the router in question is a 2811 running c2800nm-ipvoicek9-mz.151-4.M4.bin. Not sure if that makes a difference or not. !!! WITH DEFAULT CHANNEL IE SETTING AND BOTTOM UP *Apr 13 22:35:59.915: ISDN Se0/1/0:23 Q931: TX - SETUP pd = 8 callref = 0x0003 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Facility i = 0x9F8B0100A112020101020100800A4252312D50686F6E6532 Protocol Profile = Networking Extensions 0xA112020101020100800A4252312D50686F6E6532 Component = Invoke component Invoke Id = 1 Operation = CallingName Name Presentation Allowed Extended Name = BR1-Phone2 Display i = 'BR1-Phone2' Calling Party Number i = 0x0081, '6178631002' BR1(config)# Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '6178632683' Plan:ISDN, Type:National !!! CHANGING THE IE SETTING TO SLOTMAP, keep Channel selection order at bottom up as before *Apr 13 22:39:36.506: ISDN Se0/1/0:23 Q931: TX - SETUP pd = 8 callref = 0x0001 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA99304 Exclusive, Channel 0 Facility i = 0x9F8B0100A112020101020100800A4252312D50686F6E6532 Protocol Profile = Networking Extensions 0xA112020101020100800A4252312D50686F6E6532 Component = Invoke component Invoke Id = 1 Operation = CallingName Name Presentation Allowed Extended Name = BR1-Phone2 Display i = 'BR1-Phone2' Calling Party Number i = 0x0081, '6178631002' BR1(config)# Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '6178632683' Plan:ISDN, Type:National !! CONFIG service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR1 ! boot-start-marker boot-end-marker ! ! no aaa new-model network-clock-participate wic 1 network-clock-select 1 T1 0/1/0 ! no ip domain lookup ip domain name proctorlabs.com no ipv6 cef multilink bundle-name authenticated ! ! isdn switch-type primary-ni ! voice-card 0 ! license udi pid CISCO2811 sn FTX1011A423 ! ! controller T1 0/1/0 pri-group timeslots 1-3,24 service mgcp ! ip tcp synwait-time 5 ! ! interface Loopback0 ip address 10.10.110.2 255.255.255.255 ip ospf network point-to-point
Re: [OSL | CCIE_Voice] Unity Live Record
Suresh, I am not aware of any such solution for CUCM/CUE integration. Live Record works as follows in CUCM/CUE: CUE Config: same as for CME/CUE CUCM Config: 1. Make sure you have a conference bridge (CFB) (HW or SW, depending on need) 2. Make sure your CFB is in a MRG and that the MRG is in a MRGL for the phone you want to initiate LiveRecord 3. You need a DN configured with your Live Record pattern that is provisioned to Call Forward All to your CUE VM Pilot. There are a few ways you can do this. I prefer the method of creating a dummy CTI Route Point (it won't register to UCM) and put the Live Record pattern on the CTI RP. Configured to CFA to VM (CUE) To use LiveRecord from the phone. Have an active call up and then use the Conference Softkey to conference in the Live Record device. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 10, 2013, at 3:58 AM, Suresh Bhandari wrote: Hello Experts! I was trying to record a conference, but I can do that only by conferencing the Live-record DN itself into the conference. (Vol 2 Lab 8 Task 4.2 +) Tried to configure softkey as in CME, but there is no Live Record option available for UCM when connected! Is there any way so that I can record the conference just by pressing a key or so? TIA -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unity Live Record
Suresh. I understand. There is no nifty softkey for live record and CUCM/CUC. From the CUCM perspective, setting up Live Record in CUC is basically the same as with CUE. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 10, 2013, at 11:41 AM, Suresh Bhandari wrote: PS. William, is this the only way to live record. If yes, then I already did that with CUCM-CUC. I was driven by a thought of similar live record execution (using softkey in case of) as in CME-CUE. Thanks On Wed, Apr 10, 2013 at 9:17 PM, Suresh Bhandari bring...@gmail.com wrote: Thanks for the valuable replies. My fault. I didn't mention earlier that I was trying to work on CUCM-CUC. Not CUCM-CUE or CME-CUE. I know the CME-CUE live record configuration. I am trying to get the similar results from CUCM-CUC. For Live record in UCM, I can't find a softkey/phone button. Keeping in mind my question and the situation, is there any suggestions that I should follow. Thanks. On Wed, Apr 10, 2013 at 8:52 PM, William Bell b...@ucguerrilla.com wrote: Suresh, I am not aware of any such solution for CUCM/CUE integration. Live Record works as follows in CUCM/CUE: CUE Config: same as for CME/CUE CUCM Config: 1. Make sure you have a conference bridge (CFB) (HW or SW, depending on need) 2. Make sure your CFB is in a MRG and that the MRG is in a MRGL for the phone you want to initiate LiveRecord 3. You need a DN configured with your Live Record pattern that is provisioned to Call Forward All to your CUE VM Pilot. There are a few ways you can do this. I prefer the method of creating a dummy CTI Route Point (it won't register to UCM) and put the Live Record pattern on the CTI RP. Configured to CFA to VM (CUE) To use LiveRecord from the phone. Have an active call up and then use the Conference Softkey to conference in the Live Record device. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 10, 2013, at 3:58 AM, Suresh Bhandari wrote: Hello Experts! I was trying to record a conference, but I can do that only by conferencing the Live-record DN itself into the conference. (Vol 2 Lab 8 Task 4.2 +) Tried to configure softkey as in CME, but there is no Live Record option available for UCM when connected! Is there any way so that I can record the conference just by pressing a key or so? TIA -- Suresh Bhandari ___ 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 -- Suresh Bhandari ___ 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] incoming call from PSTN
I agree with Ikizoo on the fact that it shouldn't matter what the PSTN is handing off to your gateway. You have multiple ways to handle digit manipulation on ingress: 1. MGCP, chop down to 4d at the gateway 2. Using a translation pattern that the gateway CSS can see 3. H323, use voice translation-profile/-rule That said, I don't agree with the statement the PSTN can't send a +. 1. If the PSTN is outpulsing the call on one of the PRIs then it can most certainly send + 2. If the PSTN is integrated via SIP trunk (pseudo ITSP) then it can send a + 3. If the PSTN is integrated via H323 trunk (direct or via GK) then it can't send a + The PSTN phone is registered to CME and I assume it is registered as SCCP. So, it won't be H323, it will be POTS. If the phone is running the right firmware (I think 9.1(1) or later?) then it can dial a + by pressing the * key for a second or two. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 8, 2013, at 11:57 PM, Suresh Bhandari wrote: When you have the strip digits at gateway level set to 4, and as long as 2220 is a DID then from wherever it is called, it will go to the VM. Regarding plus dialing of VM DID from PSTN, it will be/is registered to CME (h.323), so I don't expect it will send a plus. My two cents. On Tue, Apr 9, 2013 at 7:51 AM, ikizoo hello ikiz...@hotmail.com wrote: Hi All, when they asked to make sure VM pilot (+1408200) can be called directly from PSTN. do i have to make a call from PSTN and dial +1408200? , but pstn phone not support to dial '+'. is that mean dial 1408200 from pstn phone or some tricks behind here? thanks advance -ikizoo ___ 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 ___ 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] CBarge question
Singh, I am not sure how to answer question #1. cBarge requires a hardware conference bridge so I don't know how one would compare/contrast the two. If you are asking about cBarge vs. ad-hoc or MML conferencing then the main difference is who initiates the conference. With cBarge a party with a shared line appearance initiates the conference by barging in on an active call. With adhoc a person already involved in the call is pulling another party in. With MML, a person initiates the bridge that others can dial into. Once configured, you would stand up an active call on the shared line and then barge in from the phone not involved in the call. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 2:37 AM, singh wrote: hi Guys, I am configuring cbarge on CME. 1) Could anyone tell me what are Cbarge's advantages or a normal hardware conference bridge? 2) Also how do I test cbarge once configured? -singh Get Yourself a cool, short @in.com Email ID now! ___ 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] MGCP IOS vs GUI settings
Not sure about item i. I always set the switch type in the GUI. For dtmf relay, I always set in IOS because I believe the versions in the lab fail to set the dtmf correctly. It also fails to set ccm-manager switchback correctly. So, I do that in IOS as well. That said, I do set the appropriate values in the GUI because I am not sure how they grade and, well, I am already in the screen dorking around. It would be interesting to here what others do for the ISDN switchtype. I do know that the CCM-Manager Config process will correctly set that information for you. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 8:51 AM, Suresh Bhandari wrote: I know that MGCP is a client/server protocol, and even that said I have a very basic (?) question: i. When I have to select the isdn switch-type to primary-ni (NI2) in the IOS before configuring the controller, and ii. if I can select the dtmf relay method to out-of-band for MGCP in IOS itself is it necessary/mandatory, in lab exam point of view, to select the said configs (ISDN Switch Type to NI2 and DTMF Relay Method to OOB) while configuring MGCP in the CUCM? Just a curiosity. Thanks, -- Suresh Bhandari ___ 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] MGCP IOS vs GUI settings
Well, sure. If you try to provision a pri-group manually and you haven't defined a switchtype then the CLI will bark at you. I was making an assumption that you were using ccm-manager config. My apologies. If you don't use ccm-manager config then you will need to specify switchtype. If you do use ccm-manager config then you can specify the switchtype in IOS or the config process will do it for you. The ccm-manager config process will not set dtmf nor will it set ccm-manager switchb (if that matters to you). -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 10:12 AM, Suresh Bhandari wrote: William, Are you sure we don't need isdn switch-type in IOS before configuring Controllers? As when I try to configure I always get the error that I should configure the said item first. pri-group time 1-3 ser mgcp %ISDN switch-type must be set first. I am following Device based approach, so until I have my ios ready I won't come to the GUI. Is it due to the approach, where I will configure everything in IOS before going to the GUI? On Tue, Apr 9, 2013 at 7:25 PM, William Bell b...@ucguerrilla.com wrote: Not sure about item i. I always set the switch type in the GUI. For dtmf relay, I always set in IOS because I believe the versions in the lab fail to set the dtmf correctly. It also fails to set ccm-manager switchback correctly. So, I do that in IOS as well. That said, I do set the appropriate values in the GUI because I am not sure how they grade and, well, I am already in the screen dorking around. It would be interesting to here what others do for the ISDN switchtype. I do know that the CCM-Manager Config process will correctly set that information for you. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 8:51 AM, Suresh Bhandari wrote: I know that MGCP is a client/server protocol, and even that said I have a very basic (?) question: i. When I have to select the isdn switch-type to primary-ni (NI2) in the IOS before configuring the controller, and ii. if I can select the dtmf relay method to out-of-band for MGCP in IOS itself is it necessary/mandatory, in lab exam point of view, to select the said configs (ISDN Switch Type to NI2 and DTMF Relay Method to OOB) while configuring MGCP in the CUCM? Just a curiosity. Thanks, -- Suresh Bhandari ___ 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 ___ 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] CBarge question
Ensure that the template is assigned to the ephone. Also, make sure that you have privacy turned off on the ephone. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 8:44 AM, singh wrote: I have a shared line 4003 on 2 of my site C phones ( registered to CME) . I call from the HQ phone to the shared line. I answer the call on SC phone 2 . The shared line is active which softkey do I press from SC phone 1 to barge in? The current tempate for the ephones is the following ephone-template 1 softkeys remote-in-use Newcall CBarge softkeys idle Cfwdall ConfList Dnd Gpickup Join Newcall Pickup Redial softkeys connected Acct ConfList Confrn Endcall Hold Join LiveRcd Mobility Par But I don't see the CBarge softkey on SC phone 1 . Why? -- Original message -- From:William Bell b...@ucguerrilla.com Date: 9 Apr 13 16:53:53 Subject: Re: [OSL | CCIE_Voice] CBarge question To: Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Singh, I am not sure how to answer qu estion #1. cBarge requires a hardware conference bridge so I don't know how one would compare/contrast the two. If you are asking about cBarge vs. ad-hoc or MML conferencing then the main difference is who initiates the conference. With cBarge a party with a shared line appearance initiates the conference by barging in on an active call. With adhoc a person already involved in the call is pulling another party in. With MML, a person initiates the bridge that others can dial into. Once configured, you would stand up an active call on the shared line and then barge in from the phone not involved in the call. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 2:37 AM, singh wrote: hi Guys, I am configuring cbarge on CME. 1) Could anyone tell me what are Cbarge's advantages or a normal hardware conference bridge? 2) Also how do I test cbarge once configured? -singh Get Yourself a cool, short @in.com Email ID now! ___ 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 Get Yourself a cool, short @in.com Email ID now! ___ 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] WAN QoS Calculations
I believe that a 768kbps link falls within the recommendation to leverage a fragmentation mechanism. So, I believe that the map-classes are accurate. Hugo, I know you said you don't want to review a SRND but I definitely recommend you take the time to a look at the WAN Edge Link-Specific QoS Design in the QoS SRND. It is an informative section and not as much of a yawn fest as you may think. Also, if you are ever asked to do class-based traffic shaping, you will be comfortable where to find some good examples. Remember that the QoS SRND is made available to you on the candidate machine. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 9, 2013, at 6:19 PM, Leslie Meade wrote: Hmmm I cannot remember but I am 95% sure J that the fragmentation is not for links over 768… Hence the map-class for the link to Site C is incorrect… remove the frame-relay fragment 960 From: Barrera, Hugo [mailto:hugo.barr...@nexusis.com] Sent: Tuesday, April 09, 2013 3:17 PM To: Abel ...; Leslie Meade Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] WAN QoS Calculations So my first set of commands below this is NOT using 95% of the BW and the second set of commands, in blue, are using 95% correct? Does this look right? map-class frame-relay AutoQoS-FR-Se0/1/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! map-class frame-relay AutoQoS-FR-Se0/1/0-202 frame-relay cir 768000 frame-relay bc 7680 frame-relay be 0 frame-relay mincir 768000 frame-relay fragment 960 service-policy output AutoQoS-Policy-Trust ! ! USING ONLY 95% OF BANDWIDTH: map-class frame-relay AutoQoS-FR-Se0/1/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! map-class frame-relay AutoQoS-FR-Se0/1/0-202 frame-relay cir 729600 frame-relay bc 7296 frame-relay be 0 frame-relay mincir 729600 frame-relay fragment 960 service-policy output AutoQoS-Policy-Trust ! ! Regards, Hugo From: Abel ... [mailto:midga...@gmail.com] Sent: Tuesday, April 09, 2013 2:47 PM To: Leslie Meade Cc: ccie_voice@onlinestudylist.com; Barrera, Hugo Subject: Re: [OSL | CCIE_Voice] WAN QoS Calculations I used the 95% rule and I passed it too. So, read the requirements word by word. On Apr 9, 2013 5:43 PM, Leslie Meade leslie.me...@lvs1.com wrote: Here is my take, and take it at face value.. When I took my lab I asked the proctor am I to follow best practices and use the 95% rule for my QOS. Their response was it is not stated to use the 95% and it was my choice… I did not follow the 95% rule and kept it as default…. And I passed…. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Barrera, Hugo Sent: Tuesday, April 09, 2013 2:09 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] WAN QoS Calculations QoS Guru’s, In the real lab I know I have to do some calculations utilizing 95% of the bandwidth…so if there is a link between SA and SB of 384k and SA and SC of 768k is the 95% from these numbers or what the actual interface can do? Also what is a simple straight to the point read on this, I really don’t want to review an srnd? Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA partial match issue
Actually, one point of clarification here. The service parameter Sergey is referring to ( Inbound Calling Search Space for Remote Destination) does not play a role in digit analysis for MVA calls. That service parameter is a little misleading and confusing. There are two call flows you have to keep in mind when dealing with ingress calls from Remote Destinations. 1. Direct inward dial from the RD to your internal dial plan (e.g. directory numbers, MML, hunt pilots, etc.) Basically, anything that is not the MVA steering digit pattern. The service parameter Sergey mentioned can affect the digit analysis applied to these calls. 2. Calls handed off by the MVA VXML application to CUCM. As Sergey noted, the IOS is terminating the call. Most of the heavy lifting is done at the gateway. The prompts, etc. are all coming by way of VXML but the call is terminated on the gateway. At least up to the point where you choose the option to place a call. At this point in time, the IOS device is going to send the call to the MVA number and the CUCM digit analysis process is going to make a routing decision based on the CSS assigned to the Remote Destination Profile (RDP), regardless of what you have set for that service parameter. So, the service parameter affects ingress calls to DIDs that are not pre-processed by the MVA VXML first. Any call that comes by way of MVA, will use the RDP CSS for digit analysis. You can test this by doing the following. Using the IPExpert topology samples. 1. Create a new partition: block-hqph1_pt 2. Put a translation in this partition 2001/block-hqph1_pt and set that translation to block the number 3. Put this partition in your RDP CSS. Ensure it is at a higher priority partition than the PT that currently holds the 2001 extension programmed on HQ Phone 1. If you use the none partition then it doesn't matter where you add the block PT. 4. Ensure that the service parameter Inbound Calling Search Space for Remote Destination is set to the default value (which is to use the GW CSS). Now, use the PSTN phone line that is associated with your RD and call 2025552001 directly. It should ring (unless you have something else messed up). Using the same PSTN phone line, call into MVA. Log into the service, press 1 to place a call and dial 2001. You should get ANN telling you that the number is unassigned. Another test that may be of interest. 1. Create a new partition: block-hqph1_pt 2. Put a translation in this partition 2001/block-hqph1_pt and set that translation to block the number 3. Put this partition in the CSS you assign to your HQ gateway. Make sure it is sitting at a higher priority than your internal phone PT (if you use one). At this point in time, any calls from the PSTN to 2025552001 will fail. 4. Set the service parameter Inbound Calling Search Space for Remote Destination to use RDP Device/Line CSS 5. Ensure that the RDP CSS does NOT have the block PT 6. Call 2025552001 from the PSTN line associated with your RD. It should work. So, now you can't call 2025552001 from any PSTN line EXCEPT for the line associated with the RD. That would be an interesting IE lab question. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 7, 2013, at 10:59 AM, Sergey Heyphets wrote: Hi Donny, When you dial 4 digit extension from the MVA, the IOS sends call to the MVA number defined under Media Resources, so you must have a dial-peer that matches that number and sends the call to the CUCM. The extension you've dialed is transfered in the Redirected number IE inside the SETUP message sent to the MVA number defined under media resources. Once the call gets to CUCM, it extracts the extension you've dialed from the Redirected Number IE and uses either Gateway CSS or RDP+Line CSS (depending on Service Parameters) to place the call to extension. So, if your call to extension doesn't work, you need to check that you have dial-peer that matches MVA number defined in Media Resources, the Service Params to see which CSS you use for MVA calls and then make sure that whatever CSS you use can reach that extension. I know there are some bugs with partial match in early versions of CUCM 7.X, the workaround is to use complete match. Sergey ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA partial match issue
Hugo may be right. There is a colleague of mine who has had MVA issues in the past and has had to either restart the service, restart Call Manager service, or reload the VXML service on the router. I don't recall exactly which step fixed the issue. Actually, he may have done all of them. Which, while it resolved the issue, was a bit heavy handed. If you are positive you have everything setup correctly then I would try the following: 1. Reprovision MVA on the IOS application no service MVA http://10.3.120.11:8080/ccmivr/pages/IVRMainpage.vxml service MVA http://10.3.120.11:8080/ccmivr/pages/IVRMainpage.vxml /*you should see the IOS read the VXML if you are on the console and logging to console*/ 2. Restart the MVA service on Pub 3. Restart CallManager service I'd try one at a time and test between. Hugo, out of curiosity, you mentioned service parameters. Were you referring to the parameter Mobile Voice Access Number . This is another parameter that confuses me. I never set it and MVA always works. I am wondering if it is legacy or is used for some other method to access mobility? Maybe part of CUMA? Do you know? -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 7, 2013, at 2:40 PM, Barrera, Hugo wrote: It uses the RDP's css while snr uses the re-routing css. Did you also specify the MVA number in the service parameters? A peer had mentioned to me that the service may need to get restarted as well haven't tested it yet though. Regards, Hugo On Apr 6, 2013, at 8:38 PM, donny f f.faraday...@gmail.com wrote: hi Bill and others, I had put the MVA under Media Resources, however when i dial 4 digit ext, it said: the number you dial can't be reached. Questions: - when we use MVA to call 4 digit, are they use IOS dial-peer or RD css to call this 4 digit local ext - my partial match never work , i use 7 digit as match. any idea what missed? tks On Wed, Mar 27, 2013 at 5:46 AM, William Bell b...@ucguerrilla.com wrote: I have ran into a similar problem. In my case I would get a fast busy after entering the extension number followed by #. The issue was I neglected to provision Mobile Voice Access under Media Resources. On Tuesday, March 26, 2013, Barrera, Hugo wrote: Regarding MVA during my first attempt (real lab) I had it working except for when I dialed in and tried to call another 4-digit ext like SAPH1 or SBPH2 any ideas why that didn’t work? Regards, Hugo From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin Carney Sent: Monday, March 25, 2013 1:51 AM To: donny f Cc: ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com); michael.se...@compucom.com; networksanitytoinsan...@gmail.com Subject: Re: [OSL | CCIE_Voice] MVA partial match issue You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match
Re: [OSL | CCIE_Voice] MVA partial match issue
Not quite. The RDP CSS is used by the MVA process in CUCM to make the final call routing decision. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 7, 2013, at 3:46 PM, donny f wrote: yes i had specified it under service param, so far i only restart the MVA service in UCM/. I think this no need RDP css, as i only test MVA. When i press 4 ext , debug voip dialpeer show it hits the MVA number 5999. Here is how I understand , pls correct if this is not right. - when press 1 to call 4 digit, dial-peer voip in IOS router will match 5999 to CallManager VMA 5999 (under Media Resources). - after successfully in UCM MVA, it is up to CallManager VMA process to dial 4 digit (and no need CSS here) Tks d On Sun, Apr 7, 2013 at 12:40 PM, Barrera, Hugo hugo.barr...@nexusis.com wrote: It uses the RDP's css while snr uses the re-routing css. Did you also specify the MVA number in the service parameters? A peer had mentioned to me that the service may need to get restarted as well haven't tested it yet though. Regards, Hugo On Apr 6, 2013, at 8:38 PM, donny f f.faraday...@gmail.com wrote: hi Bill and others, I had put the MVA under Media Resources, however when i dial 4 digit ext, it said: the number you dial can't be reached. Questions: - when we use MVA to call 4 digit, are they use IOS dial-peer or RD css to call this 4 digit local ext - my partial match never work , i use 7 digit as match. any idea what missed? tks On Wed, Mar 27, 2013 at 5:46 AM, William Bell b...@ucguerrilla.com wrote: I have ran into a similar problem. In my case I would get a fast busy after entering the extension number followed by #. The issue was I neglected to provision Mobile Voice Access under Media Resources. On Tuesday, March 26, 2013, Barrera, Hugo wrote: Regarding MVA during my first attempt (real lab) I had it working except for when I dialed in and tried to call another 4-digit ext like SAPH1 or SBPH2 any ideas why that didn’t work? Regards, Hugo From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin Carney Sent: Monday, March 25, 2013 1:51 AM To: donny f Cc: ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com); michael.se...@compucom.com; networksanitytoinsan...@gmail.com Subject: Re: [OSL | CCIE_Voice] MVA partial match issue You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information make sure you are putting a tick in Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox. Otherwise your configuration looks good. Hope you find this helpful. Michael Sears
Re: [OSL | CCIE_Voice] unity user-template no effect on existing users
You can't. The template is only used when creating users. There are bulk edit tools on unity connect. Look under the Tools section in the navigation pane. These tools may or may not expose the attribute(s) you wish to edit. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 6, 2013, at 3:58 PM, Vikky Kumar wrote: Hi, I have noticed that when i make changes to user templates (voicemailusertemplate), there is no change to already existing users in unity. how can i make global changes to users by making change in the template. thanks, Vikky ___ 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] Facility IE
Expected behavior. See CUCM's online help for that page. Send Calling Name in Facility IE Check the check box to send the calling name in the Facility IE field. By default, the Cisco Unified Communications Manager leaves the check box unchecked. Set this feature for a private network that has a PRI interface that is enabled for ISDN calling name delivery. When this check box is checked, the calling party name gets sent in the Facility IE of the SETUP or FACILITY message, so the name can display on the called party device. Set this feature for PRI trunks in a private network only. Do not set this feature for PRI trunks that are connected to the PSTN. Note: This field applies to the NI2 protocol only. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 5, 2013, at 1:24 AM, ikizoo hello wrote: Hello All, i just trying to send out facility ie in MGCP E1 PRI gw which use primary-net5, but in the CUCM gw menu, it it is greyed out. is this expected behavior or some sort of configuration error? thanks -ikizoo ___ 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 Exam Speed Strategy
To deal with the difference in IP addresses between my home lab and the real lab I got in the habit of building what I called the basic.txt file. In this file I put the following (sample) !creds os/web admin userid password !hosts PUB ip SUB ip CUC ip CCX ip UPS ip WS ip BB/NTP ip CUE ip ! ! HQ fa 0/1 R1 fa 0/2 Ph1 fa 0/3 Ph2 fa 0/5 Ph3 !vlan 100 Server subnetip 101 Voice subnetip 102 Datasubnetip lo0 ip etc. The idea is that you type out the IP addresses once. Then copy/paste into your other config notepads / web interfaces as needed. I keep the basic.txt file up all of the time. I actually put it in the top right of the screen so that I can right click/copy quickly. Just my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 3, 2013, at 2:14 AM, ie ravindra wrote: Is there any ways to adapt with IP address scheme. ? Thanksm Ravi. On Wed, Apr 3, 2013 at 6:46 AM, Josh Petro josh.pe...@gmail.com wrote: Thanks much to all who replied! Strategy seems to be key from what I'm hearing here and elsewhere. On Tue, Apr 2, 2013 at 4:45 PM, michael.se...@compucom.com wrote: All testing after you finish the lab. --ms Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist Compucom Systems Western Region Infrastructure Solutions Consulting Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax:+1.978.863.0740 image001.jpg “Designing and Implementing Cisco Unified Communications on Unified Computing Systems” From: Ramcharan Arya [mailto:ramcharan.a...@gmail.com] Sent: Tuesday, April 02, 2013 2:44 PM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy Hi Mike, Thank you for sharing great information. Can you share some detail about approach and sequence to follow like Infrastructure, gateway configuration, QoS and SRST, Presence . Unity, UCCX etc. When did you do SRST testing in the middle or at the end of the lab.? Please share your experience. Thanks Regards, Ramcharan Arya CCIE # 28926 ( Routing Switching) On Tue, Apr 2, 2013 at 12:26 PM, michael.se...@compucom.com wrote: It took me 4 attempts to pass the lab. Actually the first three attempts helped to develop a strategy for passing. The proctor in RTP, David, thought me something, don't look at a no pass as a failure but a as learning experience. After my third attempt I couldn't stand to see another fail on the score report. I took 45 days, doing two labs a day following the same strategy. If your typing skills are below 70 words/minute or less or you are hunt and peck typist take a typing class won't hurt have to type fast. Briefly read the entire lab and absorb as much as possible 5 to 10 minutes maximum regarding CUCM and gateway, QoS, etc. Perform all your switch and gateway configurations first including everything so you don't have to revisit them. Write all configuration for SW and Gateways in notepad prior to putting into devices and same to desktop, leave them there when leaving the lab. Copy all the customization's you'll need and put in notepad and put on desktop, i.e., media resources, dial-peer, other customizations. Don't type and memorize things you can obtain from links copy from links and edit 1.) Configure the SW first and take what configure you can from there and move onto R1. 2.) Configure R1 and take configuration from there to R2 and edit and add additional configuration. 3.) Configure R2 including SRST/GK/Dial-peers/MVA/everything. Move configure from R2 and R1 to R3 and edit. 4.) Configure R3/CUE/Presence/SRST using configuration from R1 and R2 that's reusable. 5.) Don't type the same thing twice. 5.) Now move to CUCM. You should have a pretty good idea of what you will need from reading lab. 6.) Open browser to CUCM Pub, Sub, Unity. Add ntp and any required customizations 7.) Configure CUCM moving from left to right, save phones for last. 8.) Configure UNITY and all voicemail customization 9.) Configure UCCX script and record prompts unless they are pre-recorded for you. 10.)Configure Presence if you have it on your lab. 11.)Need at least three hours to test and validate. 12.)Make every attempt to complete lab before lunch. 13.)Feel good at lunch relax forget the lab 14.)Get your score report that says PASS. 15.)Preform Troubleshooting as you are most comfortable with I saved it for last. There was a guy walking down the street in NYC and he recognized a famous pianist. He stopped him and ask him How do you get to Carnegie Hall. The pianist replied
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
. It doesn't get dedicated face time with me until I have the users configured and need to customize the experience per requirement. I also have a goal to get CUE (and CUC) done before lunch. Approach: I configure IS Engine interface (loopback, routing, etc.) on SC as one of the first things I do on that device. I then reset the module to ensure my IP addressing is applied. I then work on SC configs in notepad and watch CUE in the background When CUE is up, I'll do the software license install (if needed/applicable) and then restore factory defaults I'll usually have SC configs done (except maybe SRST) before CUE comes up the first time. While CUE is reloading a second time I am: working on SC-SRST configs in notepad starting my CUCM work watching CUE labor through it's start up process (I say labor because, it does seem like a lot of effort to boot up a device that doesn't do all that much. It's like watching a 90 year old man climb steps. I digress.) Bottom Line: CUE is one of those items that can mess up your rhythm. Find a way to handle it that works for you. 4. Transitioning from the router configs to the CUCM. You can get a head a steam behind you on the IOS-configs and then BLAM, you are in clickety-click land with a GUI. This transition always threw off my rhythm. I found that figuring out how I needed to deal with CUE helped me with addressing the transition to GUI-land. At the time I am working on Site C, I am working on a couple of things in parallel and am constantly busy. Which helps me carry the rhythm from the CLI-based configs forward. 5. Where to put the dial plan. I was in the mind set of trying to get the dial plan done sooner rather than later. However, since I decided to remove the task of mapping out the dial plan during the read through. I found that picking a place to stick it was pretty key. I decided to wait until after lunch to config the dial plan since lunch forces a natural transition period upon you. Also, I found that your mind could use a moment to collect itself before dorking with the dial plan. Coming back from lunch I start the afternoon the same way I did the morning. I plan. 6. SRST. There are so many freakin' bugs involving SRST in this lab that I found you want to do it sooner rather than later. I actually build the SRST configs during infrastructure phase. I don't apply the configs until I have phones registered and configured. I test SRST once and only once. I do it after I have completed all of the lab config requirements. It is the first thing I validate. 6. Transitioning from doing to checking. The area I am still working on. Over the months of prep I am putting into this thing, I have come to the realization that a validation strategy is just as important as a config strategy. I am not talking about know what commands to use to validate a task. I am talking about how you stack the validation approach. I have found that while it is more efficient to do a dev-based approach for configuration, it is not a good approach for validation. So, I do a tech-based validation. I also do some minimum validation in-line with the configuration. But definitely keep that at a minimum or you bleed minutes. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 26, 2013, at 7:40 PM, Dane Warner wrote: To All, I took my second attempt on Monday, March 25 and did not pass. I was hoping for some insight on concrete suggestions to get faster. I didn’t get hung up on any one task, I seemed to keep moving forward and tried to type as fast as I could, using CLI shortcuts, etc. I used the device-based methodology and I feel pretty confident of my technical knowledge. Yet I didn’t even get to many tasks at all, I would have needed another 2-3 hours to complete all tasks. I hear of candidates completing all tasks in 6-7 hours, which means I would need to become twice as fast as my last attempt. It almost sounds insurmountable. Do I need to take typing classes? Any recommendations that don’t break the NDA would be greatly appreciated. Regards, Dane Warner, CCVP Sr. Network Engineer Epoch Universal, Inc. (909)226-0755 dwar...@epochuniversal.com image001.png ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] Subscriber failure
You definitely have a replication issue. You cannot go by the Unified Reporting or by the perf counters (RTMT or from the CLI) as a valid indication of replication health. I know that sucks but it's true. I recommend using the CLI command: utils dbreplication status all from the Publisher In practice (and in the real lab) I use the following approach since NTP seams to be where replication could fall apart on you: 1. Make sure that if you are using HQ, BR1, BR2 or some other device you control in the lab as the NTP source for CUCM that that NTP server is associated to a valid NTP clock source and is synchronized. Very important step. 2. Set the NTP server on the Pub as you normally would 3. SSH to Pub and use utils ntp status command to ensure that time in synchronized. If it does not sync then look at step 1 and you will need to restart NTP on the pub. 4. SSH to Sub (after Pub is sync'd) and use the utils ntp restart command to restart NTP. You could check status beforehand if you want but you'll need to restart NTP 99.999% of the time. 5. Use utils ntp status on Sub to verify sync status. Once sync'd, proceed to 6. 6. Use utils dbreplication status all from Publisher to check replication status. If you see a db replication issue, I'd recommend fixing it. Various methods exist. The method I use for this lab: 1. Subscriber: utils dbreplication stop 2. Publisher: utils dbreplication stop 3. Once publisher has stopped replicating: utils dbreplication forcedatasyncsub all 4. Reload Subscriber This all takes 10 - 15 minutes, so try to avoid stupid mistakes and also try to catch dbreplication issues early on -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 2, 2013, at 1:14 PM, Hesham Abdelkereem wrote: Dear Experts, I was working yesterday on one of the online Rack Rentals. I have registered all Phones , Gateways and everything to the Subscriber. Something is very odd. I was unable to make any calls from the phone at all and the calls were not reaching the gateway. I have deleted the SLRG and Recreated, Delete all Route Patterns and then Recreated them again. Deleted all Route Groups and recreated them again. Disassociated LRG from Device Pool and Recreated them Again never worked. Restarted all Device Pool , Phones and Gateways never worked. However, When I shut down the subscriber and when it was restarting and everything fails over on Publisher then everything works perfectly and as soon as the Subscriber comes back everything is ruined. However , NTP Server is configured properly , Checked DB replication in Unified Reporting and it's good status. All Endpoints shows registered successfully but I am unable to perform calls. All Devices are configured with the correct Device Pool and Correct CSS. So what's likely other problem that makes the subscriber fail? I restarted it and as soon as it comes back nothing works. Thanks a lot for your great efforts. Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
I am not familiar with Marko's approach for on-screen window placement. I actually don't have a specific strategy in this area. I do create a notepad file for the following: basic.txt : basic infrastructure notes and notes on phone/user configs sw.txt : switch configs hq.txt : HQ gateway/router configs sb.txt : Site B gateway/router configs sc.txt : Site C gateway/router configs rp.txt : Route plan configs (when I get to that point) I have the above .txt files open all of the time. I only keep basic.txt up on the screen. I keep the others minimized. I restore them as needed. During the course of the exam I will create other notepad files temporarily. Most notably: 1. When I create partitions. I have a naming convention that is basically uniform across sites. So, I lay out the HQ versions in notepad. Paste in CUCM. Then do a search/replace for HQ/SB. Repeat for Site C. Kill the notepad 2. When I provision phones. I use a series of SQL commands from the CLI to provision phones. I type them out in notepad and paste from there. Then I kill the notepad. 3. Troubleshooting questions. Because I don't want to deal with VNC's sluggish nature, I'll do my TS work in notepad on the candidate PC and then copy/paste to the VNC desktop. I think that's it. As far as window orientation. I keep basic.txt in the top right corner of the screen. If I need hq.txt/sb.txt/etc. then I restore to bottom right. I'll keep (or try to keep) console sessions in the middle and IE sessions near the left. But I haven't really thought about it that much. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 3, 2013, at 12:32 PM, Ramcharan Arya wrote: Hi Bill, Thank you very much for nice writeup on strategy. This is really helpful for CCIE vocie lab aspirants. Do you have any strategy how many notepad sessions to keep open simultaneously. How to arrange SecureCRT sessions screen, online lab webpage, and notepad on 32 screen. I am still practice same method which I learn during RS bootcamp with Marko.If you have any better approach please share. Regards, Ramcharan Arya CCIE # 28926 (RS) On Wed, Apr 3, 2013 at 10:57 AM, William Bell b...@ucguerrilla.com wrote: I have had this as a draft for a few days. Just too busy to finish it until now. So, some of my thoughts are redundant to what others have said. Hopefully that isn't a bad thing. Timing is definitely a critical aspect of the exam. I know I have areas where I am slower than I should be. I suspect most people do. Most of my comments herein are based on my self-study practice labs. I have taken the lab a couple of times but most of the tinkering I have done with my method is during self-study. When I sit for the real lab, I don't tinker. I go with whatever method I have been practicing. So, that is suggestion #1: Don't tinker on lab day, stick to your guns and don't 2nd guess your method. Going back to the OP, I believe you should look at the bright side. Your statement ...I seemed to keep moving forward... is key. The fact you were able to avoid a stall is important. I believe controlling this exam is about rhythm and finding what config approach helps you establish a sustainable and consistent rhythm. Speed on any individual task is critical but rhythm is king in my opinion. Like others (most?), I follow the device-based approach. It has been around since pre 3.0 blueprint (contrary to popular opinion) and is a proven strategy. However, I have found that you will need to customize that approach to suit your needs. For me, it is about managing the transitions. Again, I believe focusing on establishing and maintaining a rhythm is absolutely key. Smoothing the transitions and/or stacking tasks that help ease transitions is important. Also, you won't maintain the same rhythm throughout the exam. Some tasks you will bang out (or should) very fast. Others, you will need to pay close attention to what you are doing. So, suggestion #2 is find your rhythm. Establishing your rhythm is a product of repetition. Practice, practice, practice as Mr. Sears puts it. You may also need some face time with the real lab to help you come into your rhythm. For example, some of the weak spots I had (or maybe still have) and adjustments I made. 1. Transitioning from read-through to config. The read through is/was the worst for me. Most people I have spoken with (who have passed or come close) are able to get through the read-through in 30 minutes. Some say less. I was taking a whole lot more time than 30 minutes. My budget for this task is 30m today. Adjustments I made: Dial Plan. I was building out my dial plan (on paper) during the read through. My logic was that you have to do it at some point, just do it now. The flaw with that logic is that to establish a good rhythm
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
I separate QoS from standard infrastructure and do it later for two main reasons: 1. I typically use auto qos for LAN QoS. There is just something about the mechanics of that process that is a shift from how I build the CLI commands for other infrastructure bits. That shift is large enough to throw off my rhythm. 2. I like to get my phones, media resources, and GW devices registered to CUCM before dorking with QoS. I then check registrations after QoS is in place. This helps me avoid having too many things to check if my phones or some other device has registration issues. If I do dev reg before QoS then the scope of issue root cause is smaller. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 3, 2013, at 12:58 PM, Suresh Bhandari wrote: A real long mail to read But I read it entirely, not skipping a word. Thanks Bill for sharing the concept. I am also following the device based approach, but I haven't thought of what you call it basic.txt. It will be awesome piece of information in notepad, during the lab day. I saw that everyone is responding QoS as the tail-ender config... what is the speed you can expect at the lab (thinking it might be due to the speed/delay)? Thanks once again. On Wed, Apr 3, 2013 at 9:42 PM, William Bell b...@ucguerrilla.com wrote: I have had this as a draft for a few days. Just too busy to finish it until now. So, some of my thoughts are redundant to what others have said. Hopefully that isn't a bad thing. Timing is definitely a critical aspect of the exam. I know I have areas where I am slower than I should be. I suspect most people do. Most of my comments herein are based on my self-study practice labs. I have taken the lab a couple of times but most of the tinkering I have done with my method is during self-study. When I sit for the real lab, I don't tinker. I go with whatever method I have been practicing. So, that is suggestion #1: Don't tinker on lab day, stick to your guns and don't 2nd guess your method. Going back to the OP, I believe you should look at the bright side. Your statement ...I seemed to keep moving forward... is key. The fact you were able to avoid a stall is important. I believe controlling this exam is about rhythm and finding what config approach helps you establish a sustainable and consistent rhythm. Speed on any individual task is critical but rhythm is king in my opinion. Like others (most?), I follow the device-based approach. It has been around since pre 3.0 blueprint (contrary to popular opinion) and is a proven strategy. However, I have found that you will need to customize that approach to suit your needs. For me, it is about managing the transitions. Again, I believe focusing on establishing and maintaining a rhythm is absolutely key. Smoothing the transitions and/or stacking tasks that help ease transitions is important. Also, you won't maintain the same rhythm throughout the exam. Some tasks you will bang out (or should) very fast. Others, you will need to pay close attention to what you are doing. So, suggestion #2 is find your rhythm. Establishing your rhythm is a product of repetition. Practice, practice, practice as Mr. Sears puts it. You may also need some face time with the real lab to help you come into your rhythm. For example, some of the weak spots I had (or maybe still have) and adjustments I made. 1. Transitioning from read-through to config. The read through is/was the worst for me. Most people I have spoken with (who have passed or come close) are able to get through the read-through in 30 minutes. Some say less. I was taking a whole lot more time than 30 minutes. My budget for this task is 30m today. Adjustments I made: Dial Plan. I was building out my dial plan (on paper) during the read through. My logic was that you have to do it at some point, just do it now. The flaw with that logic is that to establish a good rhythm, you need to avoid lingering on a task for too long. I decided that I would focus on getting the tasks mapped out as quick as possible and the task of mapping out a dial plan could wait. So, I added a section in my table to track the DP-related tasks by task ID (e.g. 4.1) only. Skim. I already know that I am going to do a thorough read on the questions at least once (during config) and likely twice (during validation). No sense in reading the question in detail 3 times. So, I mainly focus on what devices/apps are affected by the question and put the ID in the table. How I use this task: I build a table (like the dev-based approach table) to track tasks I build a table to track what the PSTN wants to see for off net calls (this is key because I can build an entire h323 config just on that info) I track phone/user features/buttons/etc. This goes to speed when customizing phones I build a basic.txt text file
Re: [OSL | CCIE_Voice] how to activate CUPS services
Well, that is an interesting question. Not sure why one would restrict you from using the serviceability tool but I am thinking that you may be able to do this via a SQL update. Though, I haven't tested this. I may test it later tonight after I reset my home lab. Anyway, the service activation status is stored in the SQL table processnodeservice. Something like the following may work: run sql update processnodeservice set enable='t' where tkservice 99 and tkservice 109 Then you can start the service. Again, I haven't tested this so use at your own risk. I will try to test later since I can always rollback the snapshot on home VM if needed. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 3, 2013, at 6:47 PM, ikizoo4 kwon wrote: Hello ALL i am trying to activate CUPS service in SSH CLI, but it looks like not working? FYI, i am not allow to use serviceability/tools menu.. anybody have idea on this? Ikizoo admin:util service admin:utils service utils service auto-restart utils service list utils service restart utils service start utils service stop admin:utils service start ? Syntax: utils service start serv servmandatory name of the service to be started (Note: The serv name may consist of multiple words) admin:utils service start Cisco UP Sync Agent Service Manager is running Service Not Activated Cisco UP Sync Agent[NOTRUNNING] admin: Control-C pressed ___ 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] Subscriber failure
No, I have not had replication issues in the real lab and I check the replication after NTP, every time. I have had replication issues in my home lab but only when I have done something stupid. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 3, 2013, at 9:04 PM, Josh Petro wrote: Have you (or anyone) run into replication issues in the real lab? Im assuming Cisco uses vmware which 7 isn't approved on, so that kind of cheating on cisco's part, no? Your steps are a valid real world troubleshooting technique for sure, but the lab should have better thing to test on I hope. On Apr 3, 2013 12:18 PM, William Bell b...@ucguerrilla.com wrote: You definitely have a replication issue. You cannot go by the Unified Reporting or by the perf counters (RTMT or from the CLI) as a valid indication of replication health. I know that sucks but it's true. I recommend using the CLI command: utils dbreplication status all from the Publisher In practice (and in the real lab) I use the following approach since NTP seams to be where replication could fall apart on you: 1. Make sure that if you are using HQ, BR1, BR2 or some other device you control in the lab as the NTP source for CUCM that that NTP server is associated to a valid NTP clock source and is synchronized. Very important step. 2. Set the NTP server on the Pub as you normally would 3. SSH to Pub and use utils ntp status command to ensure that time in synchronized. If it does not sync then look at step 1 and you will need to restart NTP on the pub. 4. SSH to Sub (after Pub is sync'd) and use the utils ntp restart command to restart NTP. You could check status beforehand if you want but you'll need to restart NTP 99.999% of the time. 5. Use utils ntp status on Sub to verify sync status. Once sync'd, proceed to 6. 6. Use utils dbreplication status all from Publisher to check replication status. If you see a db replication issue, I'd recommend fixing it. Various methods exist. The method I use for this lab: 1. Subscriber: utils dbreplication stop 2. Publisher: utils dbreplication stop 3. Once publisher has stopped replicating: utils dbreplication forcedatasyncsub all 4. Reload Subscriber This all takes 10 - 15 minutes, so try to avoid stupid mistakes and also try to catch dbreplication issues early on -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 2, 2013, at 1:14 PM, Hesham Abdelkereem wrote: Dear Experts, I was working yesterday on one of the online Rack Rentals. I have registered all Phones , Gateways and everything to the Subscriber. Something is very odd. I was unable to make any calls from the phone at all and the calls were not reaching the gateway. I have deleted the SLRG and Recreated, Delete all Route Patterns and then Recreated them again. Deleted all Route Groups and recreated them again. Disassociated LRG from Device Pool and Recreated them Again never worked. Restarted all Device Pool , Phones and Gateways never worked. However, When I shut down the subscriber and when it was restarting and everything fails over on Publisher then everything works perfectly and as soon as the Subscriber comes back everything is ruined. However , NTP Server is configured properly , Checked DB replication in Unified Reporting and it's good status. All Endpoints shows registered successfully but I am unable to perform calls. All Devices are configured with the correct Device Pool and Correct CSS. So what's likely other problem that makes the subscriber fail? I restarted it and as soon as it comes back nothing works. Thanks a lot for your great efforts. Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] [g729 intra region codec - considerations]
Yes there is a bug related to intra-region codecs. I've seen an issue when using CUCM with gatekeeper. Not sure if the issue manifests in other ways or not. I follow the approach of setting G729 as the default intra-region codec as part of my base config. I then ensure I hard code inter/intra-region codec settings according to lab requirements. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 31, 2013, at 3:55 PM, ie ravindra wrote: Dear All, Happy april 01st for all of you.. :-). Is there bug in voice LAB which we need to use intra region codec as forced g729. If so what we need to consider. Thanks, Ravi. ___ 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] B-ACD problem
Mike, I provided a known working config below. This is for a CME-SRST config. So, my telephony-service config stanza will not line up with your CME requirements. The moh line is the key bit here. Out of curiosity, what interface on the local router is the IP address 10.2.2.1 bound to? You will want it to be a loopback interface on the router. !Config Sample R3#sh run | s application application service app-b-acd-aa paramspace english index 1 param max-time-call-retry 700 param voice-mail 4600 param service-name app-b-acd param number-of-hunt-grps 1 param drop-through-option 1 paramspace english language en param handoff-string app-b-acd-aa param max-time-vm-retry 2 paramspace english location flash:/bacdprompts/ param aa-pilot 4000 param drop-through-prompt _connect_prompt.au param second-greeting-time 60 param call-retry-timer 15 ! service app-b-acd param queue-len 1 param aa-hunt1 4999 param queue-manager-debugs 1 param number-of-hunt-grps 1 ! R3# R3#sh run | s ephone-hunt ephone-hunt 1 longest-idle pilot 4999 list 4001, 4002 timeout 12, 12 R3#sh run | s 8102 dial-peer voice 81020 voip service app-b-acd-aa max-conn 2 destination-pattern 4000 session target ipv4:10.3.110.3 incoming called-number 4000 dtmf-relay h245-alphanumeric codec g711ulaw R3#sh run int lo0 ! interface Loopback0 ip address 10.3.110.3 255.255.255.0 ip ospf network point-to-point ! R3#sh run | s telephony-s telephony-service srst mode auto-provision dn srst ephone template 1 max-ephones 10 max-dn 20 ip source-address 10.3.103.1 port 2000 strict-match timeouts interdigit 5 system message Your current options time-zone 42 time-format 24 voicemail 4600 mwi relay max-conferences 8 gain -6 moh music-on-hold.au transfer-system full-consult secondary-dialtone 9 ! -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 31, 2013, at 7:50 PM, Mike wrote: I changed the hunt group and still didn’t work. Does anyone have a working B-ACD config with drop through using a VoIP dial-peer that I can test with? I run script debug and the tcl script isn’t even being invoked. Thanks. Mike From: Suresh Bhandari [mailto:bring...@gmail.com] Sent: Thursday, March 28, 2013 12:13 AM To: Mike Cc: Sergey Heyphets; Online Study Subject: Re: [OSL | CCIE_Voice] B-ACD problem Change the hunt group to ephone-hunt 1 to match what is specified in your queue application. It will work. HTH On Thu, Mar 28, 2013 at 8:16 AM, Mike mik...@msn.com wrote: Sorry its there should have included it in the config. ephone-hunt 2 longest-idle pilot list 3312 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergey Heyphets Sent: Wednesday, March 27, 2013 10:02 PM To: Online Study (ccie_voice@onlinestudylist.com) Subject: Re: [OSL | CCIE_Voice] B-ACD problem I believe you're missing ephone-hunt object with pilot number and list of DNs to try. Sergey On Wed, Mar 27, 2013 at 7:56 PM, Mike mik...@msn.com wrote: Anyone see any issues with the config? When I dial 5000 it just times out. application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 0 param number-of-hunt-grps 1 param drop-through-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 5000 paramspace english location flash: param second-greeting-time 60 param drop-through-prompt _bacd_allagentsbusy.au param call-retry-timer 15 param max-time-call-retry 700 param voice-mail 5000 param service-name queue ! service queue flash:app-b-acd-3.0.0.2.tcl param queue-len 10 param aa-hunt1 param queue-manager-debugs 1 param number-of-hunt-grps 1 dial-peer voice 5000 voip service aa destination-pattern 5000 session target ipv4:10.2.2.1 incoming called-number 5000 dtmf-relay h245-alphanumeric codec g711ulaw no vad telephony-service sdspfarm units 3 sdspfarm transcode sessions 3 sdspfarm tag 1 xcode1 sdspfarm tag 2 hwconf authentication credential admin cisco em logout 0:0 0:0 0:0 max-ephones 5 max-dn 10 ip source-address 10.2.2.1 port 2000 system message CCIE LAB CME load 7921 CP7921G-1.2.1 load 7941 SCCP41.8-3-3S load 7961 SCCP41.8-3-3S voicemail 8000 max-conferences 12 gain -6 moh en_bacd_music_on_hold.au multicast moh 239.1.1.1 port 2000 web admin system name admin password cisco dn-webedit transfer-system full-consult create cnf-files version-stamp 7960 Mar 25 2013 15:20:54 sh telephony-service CONFIG (Version=7.1) = Version 7.1 Cisco Unified Communications Manager Express ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
Re: [OSL | CCIE_Voice] MVA functionality
Not really. MGCP is unable to load the VXML application because the Q931 is back hauled. If the ingress gateway was H323 then it could actually service more than one UCM cluster. Obviously, network (security/QoS) considerations must be taken into account. You could also put a H323 GW local to the target CUCM cluster and use the hairpin approach to launch MVA. Not optimal but certainly feasible. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 29, 2013, at 11:06 AM, Ahmad Taamneh wrote: The gateway on the other cluster is mgcp, does it help Sent from my iPhone On Mar 29, 2013, at 10:16 AM, Pixar Perfect pixarperf...@live.com wrote: And where do you plan to invoke the script and vxml function? Date: Wed, 27 Mar 2013 23:51:10 +0300 From: aboaz...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA functionality Hello Friends... I have the following setup, I am not sure if the will be suitable to enable the MVA feature ! I have CUCM cluster, but his CUCM cluster has no voice GW or DID .. but this CUCM cluster has Inter-cluster trunk to another CUCM cluster which has the DID numbers ? Can I configure the MVA for this setup.. Appreciate your input. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voicemail Ports on SA
I follow the requirements for site A phones but I mask with the number of the VM pilot. So, using ipExpert lab examples: Site A would be +1202555 for the phones Unity Connection VM Pilot number is 2600 I set the external mask on vm ports to +12025552600. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 27, 2013, at 9:50 PM, Barrera, Hugo wrote: Hi Guy’s, For the voicemail ports on SA what do you recommend to put for the external mask? Should it match the phones external mask OR should it be only 10 digits because you’re not supposed to send the 1 out of SA? Thoughts would be appreciated? - Hugo ___ 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] Voicemail Ports on SA
Steve, Not sure if you were presenting that question to me or the OP. I can answer from my perspective: I set the emask on the ports because I feel it is best practice and it is part of my base config. I am already dorking with the VM ports to assign a CSS, AAR CSS, and AAR group and while I am there I mod the emask and save the config. As far as relevance or how it is used. If you didn't toggle the Display Original Calling Number on Transfer from Unity from the default value then the port emask would be used for direct and transfer calls from CUC. If you toggled the aforementioned service parameter to true then the emask on the port would only be used for direct calls. A relevant IE question involving direct calls would be sending notification messages to a telephone number. Whether that question has been / is / will be on an IE exam is any one's guess. Again, I set the emask to the vmpilot number as part of base config template. Now, if I wasn't already going into the VM port for another reason then I'd likely say screw it unless there was a question that made me deal with it. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 28, 2013, at 2:36 PM, Steve Keller wrote: Would this only be used if there is a call outbound from Unity and you do not have the service parameter to use original caller id when call routes through unity? Not sure of the exact parameter name , but i think everyone is familiar with that one by now. Thus caller id would be Voicemail/+12025552002 for a call the came from Unity. Even in this case i would think you would want to change the service parameter to pass the original party caller id through. I cannot think of another place this value would get leveraged. For AAR or SRST you always call the pilot. No devices every really try to call the vm ports themselves. Please let me know if there is some other feature that would make setting the VM port external number mask useful. Very curious as to the motivation to set this. thanks steve On Thu, Mar 28, 2013 at 7:43 AM, William Bell b...@ucguerrilla.com wrote: I follow the requirements for site A phones but I mask with the number of the VM pilot. So, using ipExpert lab examples: Site A would be +1202555 for the phones Unity Connection VM Pilot number is 2600 I set the external mask on vm ports to +12025552600. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 27, 2013, at 9:50 PM, Barrera, Hugo wrote: Hi Guy’s, For the voicemail ports on SA what do you recommend to put for the external mask? Should it match the phones external mask OR should it be only 10 digits because you’re not supposed to send the 1 out of SA? Thoughts would be appreciated? - Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA partial match issue
I have ran into a similar problem. In my case I would get a fast busy after
entering the extension number followed by #.
The issue was I neglected to provision Mobile Voice Access under Media
Resources.
On Tuesday, March 26, 2013, Barrera, Hugo wrote:
Regarding MVA during my first attempt (real lab) I had it working except
for when I dialed in and tried to call another 4-digit ext like SAPH1 or
SBPH2 any ideas why that didn’t work?
** **
*Regards,***
*Hugo*
** **
*From:* ccie_voice-boun...@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice-boun...@onlinestudylist.com'); [mailto:
ccie_voice-boun...@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice-boun...@onlinestudylist.com');] *On Behalf Of *Justin Carney
*Sent:* Monday, March 25, 2013 1:51 AM
*To:* donny f
*Cc:* ccie_voice@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com');,
(ccie_voice@onlinestudylist.comjavascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com'););
michael.se...@compucom.com javascript:_e({}, 'cvml',
'michael.se...@compucom.com');;
networksanitytoinsan...@gmail.comjavascript:_e({}, 'cvml',
'networksanitytoinsan...@gmail.com');
*Subject:* Re: [OSL | CCIE_Voice] MVA partial match issue
** **
You can have the rd with 7 digits only and without the 9 for pstn access -
use either application dial rules (match 7 digits, prefix 9) or a
translation pattern to modify the rd to match your existing local route
pattern.
I'm not sure if there's an MVA bug in this version of cucm, but its pretty
easy to configure it so that you always have a full match since you will
likely have only one rd. This is what I do for the lab.
A real world (for nanp) example of MVA partial match would be using e164
address for all rd (+1 npa-nxx-) and set partial match to 10 or 7
depending on whether all sites receive inbound ani as 10d for local calls
or if any sites receives only 7d. This would also work for lab, but takes
extra steps if you aren't already required to use + dialing
For partial match to work, the rd must be longer than the inbound ani (ani
7d and rd +11d). You cannot use partial match with an ani longer than the
rd (ani 10d and rd 7d), in this case your options would be to apply inbound
transformation on the gateway to make rd ani shorter (ie match the rd) or
make your rd longer and manipulate outbound dnis to make it route.
On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote:
hi,
I config the Service parameter for MVA , using partial match 7 digit .
However when I dial the RD using 7 digit ,it never works.
seem like UCM only take Full match. I heard this is bug,
Any suggestion for the work around if still want to use partial match ?*
***
d
On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote:
Greetings,
I think you are doing everything right just need a few tweaks. Place a
call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway.
What digits do you see for the calling number. 7 or 10? If seeing 7
digits inbound change your Remote Destination Number to 525, without
the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your
remote destination number to XXX525, in other words match what you're
seeing in the isdn debug for calling party and make that you're Remote
Destination Number.
Do NOT require the prefix of 9 on the Remote Destination Number. Also,
under Remote Destination Information make sure you are putting a tick in
Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox.
Otherwise your configuration looks good. Hope you find this helpful.
Michael Sears
CCIE 38404
Date: Sun, 17 Mar 2013 18:23:01 +0530
From: sanity insanity networksanitytoinsan...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Mobile Voice Access not working since many
days!!
Message-ID:
cag4zmyxmd5xj67pwv+_gpabedjoydyg+zbnmugtsfuj3nsx...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Hello All,
I have been trying this config for MVA for close to 2 weeks now and it
does not work . Here are the details
The Issue :
==
I am trying to Intiate a Call from PSTN phone to site B gateway (H323)
3033300 it should ask for
authentication once authenticated press 1 to make any 4 digit calls if it
is from SB phone 1 . Make sure to display 4 digits number for calling
number along with calling name SB Phone 1 they can use local gateway to
make the call.
Also 2nd line on PSTN phone should be used to dial 3033300 and you will
prompted to login.
Details:
=
My config is following
1) The dial-peers are set in the following way
dial-peer voice 102 voip
preference 2
destination-pattern 3300
session target ipv4:ip address
Re: [OSL | CCIE_Voice] B-ACD drop through to loop ephone hunt a second time
Suresh, If I follow your question you are looking for a way to cause the BACD application to: 1. Drop through to a hunt group 2. Ring the ephone-dns in the hunt group 3. Play the all of our agents are busy prompt 4. Attempt to ring out the agents again 5. Hang up If that is accurate then I think you want to tweak the param max-time-call-retry timer. The default is 600. If you copied from Cisco BACD examples then you most likely have this parameter set to 700. Try setting it to 30. After changing the parameter, do the following: R3#show call application session Session ID 2A App: app-b-acd Type: Service Url: builtin:app_b_acd_script.tcl R3#call application session stop id 2A Stopping session R3# .Mar 27 22:27:13.547: %IVR-6-APP_INFO: TCL B-ACD: B-ACD Service Terminated HTH. -Bill On Mar 27, 2013, at 3:11 PM, Suresh Bhandari wrote: 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 ___ 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 -- William Bell blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla ___ 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 partial match issue
I concur with Sergey. On Mar 27, 2013, at 2:14 PM, Sergey Heyphets wrote: The prompts you hear on the when you dial-in are the results of IOS executing the VXML script, which was defined in the application/service definition. When you, however, press 1 to make the call and enter the number, the VXML script instructs the IOS to place the call to the MVA number defined under media resources. So if you don't have the MVA number defined under Media Resources, the initial prompts would work, but placing the call would fail. Sergey On Wed, Mar 27, 2013 at 1:07 PM, Barrera, Hugo hugo.barr...@nexusis.com wrote: But would the MVA number still work on the gateway when you dial in? May it would huh because the MVA AA on the IOS is separate? Regards, Hugo From: William Bell [mailto:b...@ucguerrilla.com] Sent: Wednesday, March 27, 2013 4:47 AM To: Barrera, Hugo Cc: Justin Carney; donny f; ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com); michael.se...@compucom.com; networksanitytoinsan...@gmail.com Subject: Re: [OSL | CCIE_Voice] MVA partial match issue I have ran into a similar problem. In my case I would get a fast busy after entering the extension number followed by #. The issue was I neglected to provision Mobile Voice Access under Media Resources. On Tuesday, March 26, 2013, Barrera, Hugo wrote: Regarding MVA during my first attempt (real lab) I had it working except for when I dialed in and tried to call another 4-digit ext like SAPH1 or SBPH2 any ideas why that didn’t work? Regards, Hugo From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin Carney Sent: Monday, March 25, 2013 1:51 AM To: donny f Cc: ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com); michael.se...@compucom.com; networksanitytoinsan...@gmail.com Subject: Re: [OSL | CCIE_Voice] MVA partial match issue You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information make sure you are putting a tick in Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox. Otherwise your configuration looks good. Hope you find this helpful. Michael Sears CCIE 38404 Date: Sun, 17 Mar 2013 18:23:01 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Mobile Voice Access not working since many days!! Message-ID: cag4zmyxmd5xj67pwv+_gpabedjoydyg+zbnmugtsfuj3nsx...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, I have been trying this config for MVA for close to 2 weeks now and it does not work . Here are the details
Re: [OSL | CCIE_Voice] UCCX agent routing and script verification!!
When you use a media step that is collecting input from the user, you need to determine whether you want to repeatedly prompt the user when (a) they fail to respond or (b) their response doesn't match filter criteria. There is a retries setting in the Menu step. Adjust that to adjust the behavior. The default is 3 retries. Which accounts for the 4 prompts you are hearing. The question should give you guidance here. Given the way you presented the question, I would adjust retries to 0 and then it should meet expected requirements. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 26, 2013, at 6:26 AM, sanity insanity wrote: hi guys, Any update ? I Don't have this working... -MJ On Mon, Mar 25, 2013 at 3:02 PM, sanity insanity networksanitytoinsan...@gmail.com wrote: hi William, Thanks for your reply. I have now added the following for step (b) and added (c) ,(d),(e) -start -accept -play prompt ( welcome prompt) -menu( triggering contact , operator.wav) -option 1:- a) call redirect to 4001 b) If successful terminate c) if busy goto queueloop d) if Invalid goto queueloop e) if Unsuccessful goto queueloop - Under Select Resource ( triggering contact - from CSQ) :- a)queueLoop: 1) Now when I call 4000 it says Thank you for calling this number ...if you dialled this number by mistake please press 1 else someone will be with you shortly Tests done:- i I press 1 it goes to 4001 correctly - This works ii If I don't press any key and wait for timeout the same prompts I hear with are u still there ?4 times and then it goes to the agents 4101 and 4102 - not clear whether this is right ii If I press any other key other than 1 it says please dial again and I need to press the same key ( for example digit 3 on the keypad) atleast 3 times before it goes to the queue - not sure if this is the correct method. Please let me know if this is correct? Thanks once again. -Mj On Fri, Mar 22, 2013 at 12:59 AM, William Bell b...@ucguerrilla.com wrote: 1) when I call 4000 I can hear the greeting saying Press 1 to be transferred to priority agent or stay online for next available agent . The call does not ring on the ipcc agent phones of 4101 and 4102 but when I press 1 it is transferred to 4001 as expected. My question is what is preventing it from ringing 4101 and 4102 even though the agents are in a Ready state? Given the way you presented your script logic this behavior is expected. You are asking the contact to press 1 and handling the transfer action prior to the Select Resource step. 2) The resources set for 4101 and 4102 are in Resource group name S and the CSQ for this is named as CSQ. The resource criteria is Longest Available. Is this correct? Longest Idle == Longest Available 3) Any other parameter that needs to be checked under the Resource group or the CSQ? Can't say. Assuming you have configured your resources and CSQ correctly and you have properly employed either Resource Group or Skills based routing then I think you are OK. If you have failed to configure resources/CSQ/etc. correctly then you are not OK. 4)Is the configuration steps correct ? What steps are missing if any and how do we correct it? Is the script correct? Is something not behaving the way you want or expect it to? If yes, then something is provisioned incorrectly. Your script has a logic flaw. -option 1:- a) call redirect to 4001 b) If successful goto queueLoop Step (b) doesn't make sense to me. If you successfully redirect the contact then the script logic shouldn't go to the queueLoop. You should terminate. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 21, 2013, at 1:03 PM, sanity insanity wrote: Hi All, Need your help. I am configuring DNs 4101 4102 ( both DNs are uccx agent extensions). Calls to 4000 should here a greeting Press 1 to be transferred to priority agent or stay online for next available agent . If the caller presses 1 , calls should be transferred to 4001. Otherwise it should be hunted as per Longest idle time. These are the configuration steps I followed -- 1) recorded a prompt for the greeting called operator.wav 2) Configured one button login for the phone dns ( agent DNs - 4101 4102) 3) Setup the CSQ and resources in UCCX 4) Wrote the following script... -start -accept -play prompt ( welcome prompt) -menu( triggering contact , operator.wav) -option 1:- a) call redirect to 4001 b) If successful goto queueLoop - Under Select Resource ( triggering contact - from CSQ) :- a)queueLoop: -End 5) Configured a trigger for 4000 Questions : 1) when I call 4000 I can hear the greeting saying Press 1 to be transferred to priority agent or stay online
Re: [OSL | CCIE_Voice] CME Presence
There is also a third method. The method you use will depend on the requirements in the lab. They may or may not make a direct statement. More than likely they will give requirements which hint at the correct approach. Methods button 2m1 (button 2 monitors ephone-dn 1) Monitors a single DN only Can monitor a DN shared across 1 ephones blf-speed-dial Similar to monitor line This option lets you monitor SIP lines (2m1 does not) This option also lets you monitor presence subscriptions off-box (e.g. CUCM) Requires allow watch on target DN (if on CME) button 2w1 (button 2 watch ephone-dn 1) Similar to monitor line except that you are watching ALL lines on the ephone associated with the target DN (e.g. ephone-dn 1). So, with watch you are able to see status change regardless of which line is in a connected/offhook/etc. state. Works with DnD (if watched ephone selects DnD softkey then the watcher will see the status on the watch button) The watched ephone-dn must NOT be a shared line The watched ephone-dn must be the primary line on the watched phone Requires allow watch on target DN (if on CME) HTH. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 24, 2013, at 12:01 PM, Barrera, Hugo wrote: Hi Guy’s, Question for the seasoned test takers or CCIE’s…regarding CME Presence there appears to be two ways to get the same thing done, shown below. If required to monitor the status of another phone which way would you do it? Way 1: ephone-dn 1 number 1001 description 4001 allow watch ! ! ephone-dn 2 number 1002 description 1002 ! ! ephone 1 device-security-mode none mac-address .. button 1:1 ! ephone 2 device-security-mode none mac-address .. blf-speed-dial 1 4001 label “MONITOR_PH-01″ button 1:2 ! presence presence call-list ! sip-ua presence enable ! Way 2: ! ephone-dn 1 number 1001 description 4001 allow watch ! ! ephone-dn 2 number 1002 description 1002 ! ephone 2 device-security-mode none mac-address .. button 1:2 2w1 Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
Vikky, Please clarify. You say you have configured Branch 2 as CME and CUE. Then you say CUE is registered to CUCM. That doesn't jive for me. Perhaps I am mis-reading or mis-interpreting that part of the question. You also say you are doing CAC between HQ and Branch 2. Is this locations based CAC or RSVP? Finally, when you say you can't call CUE. What does that mean? Do you get a fast busy? Annunciator? Does it ring and fail? Others have touched on the key points and the natural inclination is to look at CODEC since Branch 2 phones -- CUE work fine. If Branch 2 is a CUCM site then you have to: a. Create transcoder at Branch 2. Looks like you have done this b. Make sure you have that transcoder in a MRG and MRGL that is assigned to CUE CTI devices (use Device Pool) c. If you are using RSVP. Make sure you provision the same codec under the software MTP resource as you expect to have on the WAN and that matches one of the codecs supported by the transcoder. The allow connections under voice service voip shouldn't come into play in a CUE-CUCM integration because the CUE-CUCM integration isn't SIP nor is it H323. It is TAPI. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 2:16 PM, Vikky Kumar wrote: Hi Experts, I configured branch 2 CME/CUE working normal for Voice mails. CUE is registered with CUCM but I call not call CUE(6220) from HQ and Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every where. FYI. I have also configured CAC between on Br2 site - HQ site Please hel. Regards Vikky ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
I advise against using codec pass through on the MTP. I'd recommend something like the following: dspfarm profile 2 mtp no codec g711u codec g729r8 max sess softw some number assoc app sccp no shut dspfarm profile 3 transcod codec g729r8 max sess some number assoc app sccp no shut ! ccm group 1 ..stuff.. assoc prof 2 register sc-rsvp assoc prof 3 register sc-xocder ..stuff.. ! -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 7:20 PM, Vikky Kumar wrote: Willam, BR2 is CUCM site, and there is integration b/w CUE - CUCM I have configured RSVP, rsvp bandwidth = 136 kbps on both sides When i call HQ phone to BR2-CUE it gives fast busy tone and give Ring out display on HQ Phones pt. b. transcoder in a MRG and MRGL that is assigned to CUE CTI devices (use Device Pool) ... Done Already pt. c. i want codec g729 between sites, hence under MTP i selected only codec g729r8 + codec pass thru ?? still prob.. Regards, Vikas On Thu, Mar 21, 2013 at 12:53 AM, William Bell b...@ucguerrilla.com wrote: Vikky, Please clarify. You say you have configured Branch 2 as CME and CUE. Then you say CUE is registered to CUCM. That doesn't jive for me. Perhaps I am mis-reading or mis-interpreting that part of the question. You also say you are doing CAC between HQ and Branch 2. Is this locations based CAC or RSVP? Finally, when you say you can't call CUE. What does that mean? Do you get a fast busy? Annunciator? Does it ring and fail? Others have touched on the key points and the natural inclination is to look at CODEC since Branch 2 phones -- CUE work fine. If Branch 2 is a CUCM site then you have to: a. Create transcoder at Branch 2. Looks like you have done this b. Make sure you have that transcoder in a MRG and MRGL that is assigned to CUE CTI devices (use Device Pool) c. If you are using RSVP. Make sure you provision the same codec under the software MTP resource as you expect to have on the WAN and that matches one of the codecs supported by the transcoder. The allow connections under voice service voip shouldn't come into play in a CUE-CUCM integration because the CUE-CUCM integration isn't SIP nor is it H323. It is TAPI. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 2:16 PM, Vikky Kumar wrote: Hi Experts, I configured branch 2 CME/CUE working normal for Voice mails. CUE is registered with CUCM but I call not call CUE(6220) from HQ and Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every where. FYI. I have also configured CAC between on Br2 site - HQ site Please hel. Regards Vikky ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] H323 One Way Audio Troubleshooting
It depends. If the call path to the PSTN phone is going through a voice gateway (SA, SB, or SC) then you will want to look at: 1. IP routing in your environment 2. Binding signaling protocol on your gateway !SIP voice service voip sip bind all source-interface Interface !h323 int Interface h323-g v bind src ipaddress !mgcp mgcp bind control source-interface Interface If you are using SIP or H323 or H323-GK to get to the PSTN ITSP then you will need to look at IP routing and you will need to look at the address used for the h225 trunk/ h323 gw / or SIP trunk. If GK, then look at dial-peers, do you have CUBE, etc. Now, all of that said, it still doesn't answer your exact question. Which was: how can you tell one-way audio is happening via the CUCM trace. For MGCP, I know that at the conclusion of a call there is a CMR-like message sent from the MGCP endpoint to CUCM. I have seen it in MGCP packet traces and I assume (but have not checked) that this message would show up in the CUCM trace. Someone can verify. I may test this later this evening and pipe back in myself. For H323, I don't think there is a message similar to the MGCP message. If there were, it would come on the RTCP channel. Perhaps if you enabled CMR records that would put something in the CUCM trace. Using just the H323/ISDN traces, I would say that you could look at the IP addresses presented in the H245 messages exchanged between CUCM and the remote call processing. You would need to convert from hex but if you saw a bogus address and your audio stream that had no audio was your IP phone to the PSTN, then that would be a clue. Again, I have to play with this scenario a little. For SIP, I think that you would want to look at the SDP info and check for IP addressing information. Same as for H323 but easier since you don't need to do the hex conversion. Again, not sure if there is a call statistics message involved after disconnect. You have peaked my curiosity. Thanks for that. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 19, 2013, at 4:19 AM, CISCO CCIE VOICE wrote: Hi Experts, Can any one share there knowledge and experience on how to troubleshoot one-way audio when the call is answer from PSTN phone which messages do i need to look at on RTMT and which traces do i need to enable on CUCM to check the One way audio problem .. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Enable Multicast routing on Routers
Jamie, My understanding of the OP's question was that he wanted to know how to handle multicast MOH where the stream actually traverses the WAN and is sourced from the CUCM. The original question said nothing to indicate that the OP wants to stream from the local router flash, which is a valid alternative scenario. Hope that clarifies my response for you. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 12, 2013, at 4:26 AM, Jamie Parr (jamparr) wrote: Some part of this don’t make sense to me a. We should only consider G711 for multicast moh, we want to stream multicast from the local gateways using G711, not traverse the WAN – the MOH region used by the MOH media resources device pool should be set to G711 to all regions b. Agreed multicast-routing must be enabled c. Agreed dense-mode needs to be enabled on all interfaces d. Provision the multicast audio source needs to be done and the hop count should be 1, we do not want the MOH servers to traverse the WAN e. Ccm-manager music-on-hold must be enabled in global config f.Under telephony-service (or call-manager-fallback) “moh filename.au” must be enabled – the file must be in the correct format for G711 g. Under telephony-service (or call-manager-fallback) “multicast moh 239.1.1.1 port 16384 route loopback address” must be enabled – This MUST be a loopback address or it will not work Someone please correct me if I am missing anything here Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIEing Sent: 11 March 2013 21:10 To: William Bell Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Enable Multicast routing on Routers Great Input Bell, appreciated On Mon, Mar 11, 2013 at 11:34 PM, William Bell b...@ucguerrilla.com wrote: If you are asked to do multicast over the WAN then you need to: a. Consider CODEC. Likely, you will need to support G729 across the WAN and you will want to update the IPVMS, Regions/DP, etc. to facilitate that b. Enable ip multicast-routing on HQ and SiteB routers. c. Enable pim dense-mode on all layer 3 interfaces/hops between the CUCM, the Site B phones, and PSTN callers d. Provision mcast audio source, mcast on MOH server(s), mcast on MRGs. On MOH Servers, ensure you have the proper hop count. In the IE lab v3.0 topology, there should be 3 hops from CUCM servers and SiteB phones. e. Make sure you use ccm-manager music-on-hold on SiteB gateway to service the PSTN callers -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 11, 2013, at 3:52 PM, CCIEing wrote: Hi Friends.. I have question on the MOH multicast, In case we have HQ and SiteB are connected to the same CUCM cluster, and we need to enable the the multicasting to be used with MOH. Which interfaces on both router should we enable the Multicast traffic , and based on which criteria ?? Cheers for all ___ 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] Directory folder on Router
Based on my understanding (which may be imperfect) it sounds like your flash is formatted as a Class B file system. If you want to use commands like mkdir you will need a Class C file system (on an ISR). This requires the flash be re-formatted (actually using the format command, not erase). As far as creating directories in the Class B file system, I am not sure. I have guesses but nothing absolute. As soon as I realize that flash is formatted using Class B, I reformat. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 12, 2013, at 10:51 AM, CISCO CCIE VOICE wrote: Hi Experts, can any one help me I want to create directory folder on router without formatting flash when i use mkdir command its saying that invalid input thnks ___ 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] Enable Multicast routing on Routers
If you are asked to do multicast over the WAN then you need to: a. Consider CODEC. Likely, you will need to support G729 across the WAN and you will want to update the IPVMS, Regions/DP, etc. to facilitate that b. Enable ip multicast-routing on HQ and SiteB routers. c. Enable pim dense-mode on all layer 3 interfaces/hops between the CUCM, the Site B phones, and PSTN callers d. Provision mcast audio source, mcast on MOH server(s), mcast on MRGs. On MOH Servers, ensure you have the proper hop count. In the IE lab v3.0 topology, there should be 3 hops from CUCM servers and SiteB phones. e. Make sure you use ccm-manager music-on-hold on SiteB gateway to service the PSTN callers -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 11, 2013, at 3:52 PM, CCIEing wrote: Hi Friends.. I have question on the MOH multicast, In case we have HQ and SiteB are connected to the same CUCM cluster, and we need to enable the the multicasting to be used with MOH. Which interfaces on both router should we enable the Multicast traffic , and based on which criteria ?? Cheers for all ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP and H323 Trunk to PSTN
To emulate this you will need: 1. A voip dial-peer on the PSTN router to accept incoming calls - this will be where you can emulate breaks like codec negotiation failures, etc. 2. Voice translation-rules / profiles to handle any dial plan stuff - So, if you want to pass 01191123456789 to the PSTN but your PSTN phone is configured with DN 91123456789 you will want to translate DNIS 3. GK configs - If you are emulating a gatekeeper 4. (optional) voice class / h323 class. Depends on what you want to emulate 5. voice service voip - you most likely don't need allow connections, but you may - depending on what you want to do - you may want to bind sip to a specific address The specific configurations will depend on the scenario you are trying to emulate. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 8, 2013, at 1:46 PM, CISCO CCIE VOICE wrote: HI Guys, Can any one Share H323 and SIP Trunk configuration that need to be done on PSTN Router in order for it to work properly ... Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
Pixar, Are you certain about the Phone NTP reference and CUPC? I have not heard that before. I was under the impression that CUPC would use the clock of the underlying OS. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 6, 2013, at 12:15 AM, Pixar Perfect wrote: you still need the Phone NTP reference on the labs as CUPC client is a SIP client ..there are no SIP phones on the Version 3 labs but we might see lot on Version 4. Date: Tue, 5 Mar 2013 01:05:22 +0300 From: aboaz...@gmail.com To: corygray22...@hotmail.com; bring...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference Oh thanks a lot for your input. Appreciated .. On Tue, Mar 5, 2013 at 12:48 AM, Cory Gray corygray22...@hotmail.com wrote: Phone ntp reference is for SIP phones only Sent from my iPhone On Mar 4, 2013, at 4:42 PM, CCIEing aboaz...@gmail.com wrote: Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] RSVP a big problem
I am a little confused by the question. The way I read it you have: DEFAULT Intra-site codec should be G711 DEFAULT inter-site codec should be G729 For HQ to Branch_X use G711 codec and RSVP So, in other words the question gives you permission to not use G729 for calls between HQ and Branch_X. At least, that is one way to read it. Another way to read it is that you provisioned RSVP bandwidth based on G711 but still use G729. Why one would do that I am not sure. While I am clearly confused, one thing is for certain, you would need to adjust the ip rsvp bandwidth statement if either of the above interpretations is correct. You are accounting for 4 G729 calls not 4 G711 calls with your BW statement. Sorry, not much help as the wording of the question is a little odd. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 6, 2013, at 11:19 AM, sanity insanity wrote: hi Guys, I have to Configure IP Phones and gateways in such as way that all calls within same site should use G711 Codec. Also, all calls between the sites to remote IP phones and gateways should use G729 Codec. RSVP Call Admission Control (CAC) between HQ and branch site based on bandwidth limitations. There can be 4 concurrent calls. G711 CODEC to be used for multi-directional audio. Steps:- 1) I set the location Bw between my headquater and branch as Mandatory. 2) I also have the MTP registered and added to the correct MRG MRGLs 3) The following is a snip of my config on headquarter... dspfarm profile 1 mtp no codec g711u codec g729r8 codec pass‐through rsvp maximum sessions software 4 associate application SCCP ! interface Serial0/0/0.2 point‐to‐point ip rsvp bandwidth 112 # 4 call similarly on branch site... dspfarm profile 1 mtp no codec g711u codec g729r8 codec pass‐through rsvp maximum sessions software 4 associate application SCCP ! interface Serial0/0/0.2 point‐to‐point ip rsvp bandwidth 112 # 4 call Questions: == 1) With the above config I notice that when I make a call from headquarter site 2XXX to branch site 4XXX . The message on the phone is Not enough Bandwidth and the call disconnects. What is the exact problem? 2) Is my config above correct? -MJ ___ 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] RSVP a big problem (sanity insanity)
I have to ask. Am I the only one that thinks the requirement of There can be 4 concurrent calls. G711 CODEC to be used for multi-directional audio. is odd when an earlier requirement states you should use G729 between sites? -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 6, 2013, at 1:01 PM, Jason Lee wrote: Not sure it makes any difference in this situation, but I never use codec pass-through on my configuration. I've never had any issues. On Wed, Mar 6, 2013 at 12:32 PM, michael.se...@compucom.com wrote: --MJ Your problem is a misconfigured location somewhere in CUCM. Your configuration on the gateways is correct to allow 4 calls using RSVP based CAC. In my experience the issue your running into is not going to be an issue with the configuration on your gateways (use show SCCP on gateways to verify media resource registration), but a misconfigured location in CUCM of an assignment of a location either on phone, gateway or device pool. Not only are your calls not invoking CAC/AAR but they are NOT rerouting which points to your Route Patterns/Route List configuration. You might also verify the mask on your phones regarding AAR kicking in as well as applying the AAR calling search space on the gateways and the Device level of the phone. You also need to apply the AAR group to the gateway and Phone device level. On the live level you must also set the AAR group. Michael Sears CCIE (V) 38404 2. RSVP a big problem (sanity insanity) -- Message: 2 Date: Wed, 6 Mar 2013 21:49:54 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP a big problem Message-ID: cag4zmyw5dpqbxmgrcj3finope+pnur8zbjepv26cywgqyfh...@mail.gmail.com Content-Type: text/plain; charset=utf-8 hi Guys, I have to Configure IP Phones and gateways in such as way that all calls within same site should use G711 Codec. Also, all calls between the sites to remote IP phones and gateways should use G729 Codec. RSVP Call Admission Control (CAC) between HQ and branch site based on bandwidth limitations. There can be 4 concurrent calls. G711 CODEC to be used for multi-directional audio. Steps:- 1) I set the location Bw between my headquater and branch as Mandatory. 2) I also have the MTP registered and added to the correct MRG MRGLs 3) The following is a snip of my config on headquarter... dspfarm profile 1 mtp no codec g711u codec g729r8 codec pass?through rsvp maximum sessions software 4 associate application SCCP ! interface Serial0/0/0.2 point?to?point ip rsvp bandwidth 112 # 4 call similarly on branch site... dspfarm profile 1 mtp no codec g711u codec g729r8 codec pass?through rsvp maximum sessions software 4 associate application SCCP ! interface Serial0/0/0.2 point?to?point ip rsvp bandwidth 112 # 4 call Questions: == 1) With the above config I notice that when I make a call from headquarter site 2XXX to branch site 4XXX . The message on the phone is Not enough Bandwidth and the call disconnects. What is the exact problem? 2) Is my config above correct? -MJ ___ 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] Unity Connection user template time format
First, I believe you do want the users provisioned in CUC to be provisioned with the correct timezone. Second, the method followed is up to you. I do the following: 1. Create hqusers template based on voicemailusers template. - Change timezone - Change tutorial option - Change password options (GUI and TUI) - Change password (GUI and TUI) 2. Create sbusers tmplate based on HQ - Change timezone 3. Create scusers template based on HQ - Change timezone Import users based on the appropriate template. The above is my preference. I see it this way. I have to dork with the templates anyway. I have to at least create one that modifies tutorial, password settings, etc. The other two templates only require one change each. So, that is changing two elements. In contrast, if I import all users using the same template then I have to possibly go to 4 users and make the same change. So, I am potentially changing four elements. Maybe one argues that it could be less than 4 elements (users). I don't care. At that point, it is more efficient for me to have a method that is more flexible and stick to it then ponder over such a small task at exam time. Just shoot and scoot. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 28, 2013, at 9:09 AM, Cory Gray wrote: I would rather do it on the subscriber page vs changing the template multiple times. I think that would be faster but as always, go with whatever you practice. From: Chrysostomos Christofi [mailto:ch.christ...@logicom.net] Sent: Thursday, February 28, 2013 9:07 AM To: Cory Gray; 'Nicolas MICHEL'; 'Jamie Parr (jamparr)' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Unity Connection user template time format Guys Take it logically If HQ site has different time zone with Site B then for sure the users in CUC must have the correct time zone for each branch 1) User template in CUC (modify there anything you want include time zone),Import HQ users 2) Then modify again the user template to the correct time zone for users in site B and then import the users for site B Regards From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray Sent: Πέμπτη, 28 Φεβρουαρίου 2013 2:53 μμ To: 'Nicolas MICHEL'; 'Jamie Parr (jamparr)' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity Connection user template time format I had struggled with whether to match each subscriber with their correct time zone. My GUESS is that it only matters if a Unity Connection question involves any type of time stamp such as when the message was delivered. It probably cannot hurt to do it as a best practice as I seriously doubt it can hurt your scoring but you never know so you have to decide what is best. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nicolas MICHEL Sent: Thursday, February 28, 2013 6:45 AM To: Jamie Parr (jamparr) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unity Connection user template time format Does CUCN has something to do with the display of the phone ? :=) Le 2/28/2013 12:07 PM, Jamie Parr (jamparr) a écrit : Hi all If we are instructed to display the phones time in 24 hour format, should we reflect this in the user templates for Cisco Unity? Thanks image001.jpg Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 Cisco Systems 9-11 New Square Bedfont Lakes Feltham Middlesex TW14 8HA United Kingdom www.cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CSS at gateway level
I am with Bill on this one. I actually pre-built gateway calling search spaces in my standard config template. I apply them to all gateways. I may have no partitions in the CSS but I still apply it. When I got through the lab in detail and I hit a question where I need a special pattern on ingress (just for the gateway at a site) then I stick that pattern (and new partition) into the config and move on. The benefit is that I don't have to go back into the gateway and reset anything. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 27, 2013, at 2:31 PM, Bill wrote: Does it help you, and by this I mean you specifically, reach your goal of making your dial plan work just the way you want. I know lots of people don't use css at the gw but I felt more comfortable using them. Could I pass without it, yeah as long as everything works like it is supposed to but for me it just flows better when I use them and that is how I passed. Sent from my iPad On Feb 27, 2013, at 11:40 AM, Steve Keller skeller...@gmail.com wrote: probably not necessary in this lab but it certainly would not hurt to configure a CSS-gateways and stick it everywhere in case later in the lab you needed one for some reason. The reason i most often see this done in the field is if you are getting DID's from the carrier that do not map into the extensions on the phones cleanly and you need a translation pattern to modify the DNIS to match your extensions. Just a guess, but for this lab I dont think you would have DIDs like 61455512XX but your extensions are like 655XX, maybe there is internal numbering scheme you are trying to adhere to thoughout your enterprise and your DIDs will not always match. On Wed, Feb 27, 2013 at 11:49 AM, Ben John benjoh...@hotmail.com wrote: Hello everyone, Question: it is a good practice not to configure CSS at gateway level during the real lab ? i am doing some IPExpert labs i don't see them configuring any ? My internal DNs are in none Partition. Please advise ? Ben ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] FRF12 frame-relay fragment 480
Jason, This is mostly likely due to the fact that SCCP is getting fragmented on one end and the other end isn't expecting that. When I break fragmentation in this way, I also find phones won't re-register after a restart. Also, TFTP would be broken in this scenario. Oh, it is worth noting that SCCP will try to pack as many events in a packet as possible. Wireshark will help you see this happening. Place a call to or from a phone and I suspect you will see what I mean. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 27, 2013, at 4:39 PM, Jason Aarons wrote: Odd things happen if you have “one-direction” fragmentation. Say for example at Branch1 you leave off fragment. If you call from headquarters to Branch1 the phone will ring but you can’t answer it. But at Branch1 I can call out. Then later the phones at Branch1 lose their Line Text Labels. Why does this 1 way fragmentation cause the bizzard Branch1 phone behavior? What is actually happening to create the bizarre problems at Branch1? Why does fragmentation have to be both directions? I’m going to Wireshark the back of Branch1 phone to dig into this deeper. I know both sides have to match, but why? Example Branch1#note the missing frame-relay fragment on purpose map-class frame-relay AutoQoS-FR-Se0/1/0-202 frame-relay cir 364800 frame-relay bc 3640 frame-relay be 0 frame-relay mincir 364800 service-policy output AutoQoS-Policy-Trust HQ# map-class frame-relay AutoQoS-FR-Se0/1/0-101 frame-relay cir 364800 frame-relay bc 3640 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ___ 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] Fast dialling
Actually, I think this is controlled by a CM service parameter: Strip # from called party number From the context help: This parameter enables the stripping of # sign digits from the called party information element (IE) for the inbound, outbound, Q.931, and H.225 SETUP messages. Valid values specify True (strip # sign) or False (do not # sign). This is a required field. Default: True -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 21, 2013, at 7:53 AM, Cory Gray wrote: Some correct if I am wrong but I believe MGCP drops it automatically. It is transmitted to H323 but the default dial-peer terminator is # so that is how it works there Sent from my iPhone On Feb 21, 2013, at 4:32 AM, Jamie Parr (jamparr) jamp...@cisco.com wrote: Hi all When configuring fast dialling, do we need to configure the patterns to drop the trailing # The calls seem to leave the gateway looking the same either way? Thanks image001.jpg Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 Cisco Systems 9-11 New Square Bedfont Lakes Feltham Middlesex TW14 8HA United Kingdom www.cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Recording AU file
I use UCCX to record prompts. It works fine. You can rename the .wav to .au if you want. Still works fine. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 21, 2013, at 4:08 AM, Chrysostomos Christofi wrote: Hi I have use either IPCCX or CUC to record the BACD prompts Then I have upload it to the flash with wav format and its worked perfect Just in the configuration of BACD parameters I added wav format and not au I am not 100% if this method is acceptable in real lab From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pixar Perfect Sent: Πέμπτη, 21 Φεβρουαρίου 2013 10:00 πμ To: CCIE Voice OSL Subject: [OSL | CCIE_Voice] Recording AU file What is the quickest way to record AU file in the lab for BACD file needs? i tried recording script on the IPCC Express but it dumps WAV file. CUC recording applet never works on my windows box. any better way? THANKS ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fast dialling
I don't have that problem. I have done it with and without the predot-trailing # and all calls work. Assuming we are running the same version of CUCM and more or less the same version of IOS, I'd say there is a difference in how we configure dial plans. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 21, 2013, at 9:22 AM, Bill Lake wrote: When I set up my International dial peer with 9.011 and 9.011#, I do predot for the first one and predot, trailing # for the second one. Without this my MGCP gateways will not process the calls. On Thu, Feb 21, 2013 at 8:01 AM, William Bell b...@ucguerrilla.com wrote: Actually, I think this is controlled by a CM service parameter: Strip # from called party number From the context help: This parameter enables the stripping of # sign digits from the called party information element (IE) for the inbound, outbound, Q.931, and H.225 SETUP messages. Valid values specify True (strip # sign) or False (do not # sign). This is a required field. Default: True -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 21, 2013, at 7:53 AM, Cory Gray wrote: Some correct if I am wrong but I believe MGCP drops it automatically. It is transmitted to H323 but the default dial-peer terminator is # so that is how it works there Sent from my iPhone On Feb 21, 2013, at 4:32 AM, Jamie Parr (jamparr) jamp...@cisco.com wrote: Hi all When configuring fast dialling, do we need to configure the patterns to drop the trailing # The calls seem to leave the gateway looking the same either way? Thanks image001.jpg Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 Cisco Systems 9-11 New Square Bedfont Lakes Feltham Middlesex TW14 8HA United Kingdom www.cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] Questions regarding Unified CME as SRST
Leslie/Steve/Jason, What are your thoughts on pre-configuring ephone-dns when you are permitted to use CME-SRST with autoprovision dn or all? Instead of dorking around with templates (which I hear is flaky) I was thinking about tweaking my approach to pre-configure ephone-dns when I build out SRST. I have done some basic tests and read the docs. It is supported and appears to work. The benefits: I don't have to wait for phones to failover to finish SRST related configs. I can configure BACD, call coverage for VM, mwi sip, name, description, etc. Thoughts? -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 20, 2013, at 10:23 PM, Leslie Meade wrote: Hey Steve, I just ran this via my lab and the light turns on.. If I run debug ccsip messages I see the cue send a mwi notify to the ephone and the light comes on R3(config)# *Feb 21 03:11:56.231: %IPPHONE-6-REG_ALARM: 10: Name=SEP001BD4607B13 Load= SCCP41.8-4-1S Last=TCP-timeout *Feb 21 03:11:56.279: %IPPHONE-6-REGISTER: ephone-2:SEP001BD4607B13 IP:10.69.66.20 Socket:1 DeviceType:Phone has registered. *Feb 21 03:11:58.615: %IPPHONE-6-REG_ALARM: 10: Name=SEP0017E066C2E7 Load= SCCP41.8-4-1S Last=TCP-timeout *Feb 21 03:11:58.679: %IPPHONE-6-REGISTER: ephone-1:SEP0017E066C2E7 IP:10.69.66.21 Socket:2 DeviceType:Phone has registered. *Feb 21 03:12:15.235: //-1//SIP/Msg/ccsipDisplayMsg: Received: NOTIFY sip:4002@10.69.66.254:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.69.66.253:5060;branch=z9hG4bKyHFdoT6xNYZ85fvOD9z4kQ~~1 Max-Forwards: 70 To: sip:4002@10.69.66.254:5060 From: sip:4002@10.69.66.253:5060;tag=ds3be1f82d Call-ID: c234c79-1100@sip:4002@10.69.66.253:5060 CSeq: 1 NOTIFY Content-Length: 115 Contact: sip:4002@10.69.66.253:5060 Content-Type: application/simple-message-summary Event: message-summary Messages-Waiting: yes Message-Account: sip:4002@10.69.66.253 Voice-Message: 1/0 (0/0) Fax-Message: 0/0 (0/0) *Feb 21 03:12:15.243: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.69.66.253:5060;branch=z9hG4bKyHFdoT6xNYZ85fvOD9z4kQ~~1 From: sip:4002@10.69.66.253:5060;tag=ds3be1f82d To: sip:4002@10.69.66.254:5060;tag=3BF3A8-1459 Date: Thu, 21 Feb 2013 03:12:15 GMt Call-ID: c234c79-1100@sip:4002@10.69.66.253:5060 CSeq: 1 NOTIFY Content-Length: 0 sip-ua mwi-server ipv4:10.69.66.253 expires 3600 port 5060 transport udp unsolicited ! ! ! gatekeeper shutdown ! ! telephony-service srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 30 max-dn 30 no-reg both ip source-address 10.69.66.254 port 2000 time-zone 42 voicemail 4220 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files version-stamp Jan 01 2002 00:00:00 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Keller Sent: Wednesday, February 20, 2013 12:23 PM To: Jason Lee Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] Questions regarding Unified CME as SRST Well i confirmed today that if using a CUCM-CUE integration at a branch site, th you will want to setup your MWI to be subscribe/notify when you complete your CUE integratoin with CUCM. MWI works great when registered to CUCM and using CUE for VM. When the site fails over in to srst mode and your phone has an existing MWI on it, this is what you would want to do in order to preserve that MWI lamp. 1) When integrating your CUE to CUCM choose MWI type subscribe/notify. 2) When building your router config for SRST, make sure to build an ephone-dn-template that specifies MWI SIP that will get applied to the phones when they register (under your telephony service). 3) when configuring sip-ua / mwi-server i did not use unsolicited key word this has allowed the current MWI lamp to stay lit when failover to unified CME as SRST. When integrating using unsolicited notificiatoins I was not maintaining the MWI lamp on during failover. i am using 7975 model phones and i bounced the failover a few times and it seemed to work pretty consistently. please try at home and let us know if you have the same results with this if you are interested. On Tue, Feb 19, 2013 at 11:33 PM, Jason Lee jas7...@gmail.com wrote: I typically use unsolicited on my SRST sites for MWI, but you may be on to something. Maybe this method would be preferred. All depends what they are looking for! Thus begins my tangent ;o) I've seen the same behavior with the + as Bill. Sent from my iPhone On Feb 19, 2013, at 9:55 PM, William Bell b...@ucguerrilla.com wrote: Steve, Jason's response is spot on for your first question. Though, I have found the integration to be a little flaky myself. But that was a recent
Re: [OSL | CCIE_Voice] Questions regarding Unified CME as SRST
Steve, Jason's response is spot on for your first question. Though, I have found the integration to be a little flaky myself. But that was a recent observation when I was trying pre-build ephone-dns before swinging a site to CME. In regards to your second question, I don't think the phone is display the + on the call plane. But it should display it in the status line at the bottom of the screen. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 19, 2013, at 4:33 PM, Steve Keller wrote: Recently i have noticed a few things in my lab as i have been preparing for the lab exam. Using CME as SRST specifically in this situation, i have been trying to preserve as much features and appearance as i can when my UCM phones register to the gateway. Two scenarios i have question on because i cannot seem to get them to work. 1) If my branch 2 phone has a voicemail and MWI turned on, when it falls back to Unified CME as SRST the MWI does goes off, however i can retreive the vm because my CUE integratoin does remain in tact. Is it possible to have the phones fail over and maintain the MWI status automatically? If i leave a new vm while in SRST mode then the light does come on. 2)When making a call from the UCM phones to my BR2 phones (CFUR) the call comes in and at the gateway level i see the ANI is full e164 format including the + character. However the phone never shows the plus character in SRST mode. Is this possible? Does Unified CME as SRST support the + character? I am thinking if this is possible it would be nice to include these capabilities as part of my config if asked to preserve features, functionality while in SRST. thanks in advance all. ___ 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] Questions regarding Unified CME as SRST
Jason, I have recently been toying with the idea of pre-building my ephone-dns for scenarios where a site is a CME-SRST site and I am allowed to use autoprovision dn or all. I like to build out my entire config in notepad before pasting it in and I like to fine tune the ephone-dns before they are in SRST mode. I have a done a couple of tests using this method and it seems to work fine. Though, I did see some flaky behavior with Subscribe Notify and MWI. Which is the cornerstone of my tangent ;-) -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 19, 2013, at 8:04 PM, Jason Lee wrote: Steve, I think that if you set up Subscribe Notify for MWI instead of Unsolicited Notify it might preserve the light. In order to get that to work you would have had to load the phones into SRST (auto provision all) at least once so that they populate the running config. You can then configure the mwi sip option under the ephone-dn. That will force it to subsribe to to the CUE for MWI updates. I imagine that subscription happens every time the phone comes online or in this case when they register to the CME-SRST router during failover. It should then be followed by a notify with the MWI status. I did this on a straight CME lab yesterday and pulled the following traces. Given that occurs every time the phone boots up, you should meet your requirement. I'll test tomorrow since I'll be doing a 3 CUCM site lab. r2800-2j-b(config-ephone-dn)#mwi sip r2800-2j-b(config-ephone-dn)# Feb 18 21:11:30.316: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SUBSCRIBE sip:3002@192.168.106.2:5060 SIP/2.0 -- HERE IS THE SUBSCRIBE MESSAGE Via: SIP/2.0/UDP 192.168.106.1:5060;branch=z9hG4bK191EA4 From: sip:3002@192.168.106.1;tag=58ACCE8-1615 To: sip:3002@192.168.106.2 Call-ID: 996B6A71-794611E2-80CEE1A3-4F484EB8@192.168.106.1 CSeq: 101 SUBSCRIBE Max-Forwards: 70 Date: Mon, 18 Feb 2013 21:11:30 GMT User-Agent: Cisco-SIPGateway/IOS-12.x Event: message-summary Expires: 3600 Contact: sip:3002@192.168.106.1:5060 Accept: application/simple-message-summary Content-Length: 0 Feb 18 21:13:11.067: //-1//SIP/Msg/ccsipDisplayMsg: Received: NOTIFY sip:3002@192.168.106.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.106.2:5060;branch=z9hG4bKUNWTUT.iNZVt5tr6uAHS+A~~3 Max-Forwards: 70 To: sip:3002@192.168.106.1;tag=58ACCE8-1615 From: sip:3002@192.168.106.2;tag=dec1fdb9-1100 Call-ID: 996B6A71-794611E2-80CEE1A3-4F484EB8@192.168.106.1 CSeq: 3 NOTIFY Content-Length: 114 Contact: sip:3002@192.168.106.2 Event: message-summary Allow-Events: refer Allow-Events: telephone-event Allow-Events: message-summary Subscription-State: active Content-Type: application/simple-message-summary Messages-Waiting: yes - HERE'S THE NOTIFICATION OF MWI ON Message-Account: sip:3002@192.168.106.2 Voice-Message: 1/0 (0/0) Fax-Message: 0/0 (0/0) On Tue, Feb 19, 2013 at 4:33 PM, Steve Keller skeller...@gmail.com wrote: Recently i have noticed a few things in my lab as i have been preparing for the lab exam. Using CME as SRST specifically in this situation, i have been trying to preserve as much features and appearance as i can when my UCM phones register to the gateway. Two scenarios i have question on because i cannot seem to get them to work. 1) If my branch 2 phone has a voicemail and MWI turned on, when it falls back to Unified CME as SRST the MWI does goes off, however i can retreive the vm because my CUE integratoin does remain in tact. Is it possible to have the phones fail over and maintain the MWI status automatically? If i leave a new vm while in SRST mode then the light does come on. 2)When making a call from the UCM phones to my BR2 phones (CFUR) the call comes in and at the gateway level i see the ANI is full e164 format including the + character. However the phone never shows the plus character in SRST mode. Is this possible? Does Unified CME as SRST support the + character? I am thinking if this is possible it would be nice to include these capabilities as part of my config if asked to preserve features, functionality while in SRST. thanks in advance all. ___ 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
Re: [OSL | CCIE_Voice] MWI Best Practice
Well, I don't know what will get you the points. The rumor I heard in regards to best practice is to use unsolicited. My experience is that unsolicited is reliable in a CME-CUE build. Just as reliable as the MWI On/Off. In a CME-SRST build, I have been using unsolicited without issue. Though, recently I have been testing with SIP notify since it will preserve the MWI status on failover. That said, I have (in my limited testing) seen that the SIP Notiy option is somewhat flaky. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 19, 2013, at 11:05 PM, Pixar Perfect wrote: Experts and wannabe experts friends, what are the best practices for MWI in CME and SRST modes for the CUE site BR2? i was used to using MWI ON and MWI OFF DNs on a CME but i was told by a fellow aspirant that MWI ON/OFF are not preferred (grading wise) and that solicited MWI is that gets you to the needed points. however i have seen solicited and unsolicited to be verify unreliable on 7965 phones .. you have to do no mwi sip and mwi sip to get solicited to work and sometimes reboot CUE or router to get both solicited and unsolicited to work. I am 1 month away from exam date and dont want to waste time exploring instead adopt best common practice that works flawlessly ..and so far it has been ON/OFF DN ___ 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] Custom Tones
Jason, I played with this some today and I think a lightbulb went off for me. The assumed scenario for cBarge + custom-cptone is: 1. PhoneC calls shared line on PhoneA/PhoneB (Phones A and B are registered to CME) 2. Phone A answers on shared line 3. Phone B seizes line (remote in use) and selects the cBarge softkey 4. At this point the custom-cptone for JOIN should be played out 5. Phone B disconnects from call 6. Our assumption is that the custom-cptone for LEAVE should be played out I have always had the same experience you noted. Which is: Step 4 works fine, no problem. Step 6 never works. IOW, I never hear a leave tone. I tested different configs for custom-cptone, even though doing so didn't make much sense. The behavior is the same. You do want to make sure that the frequency is different. The cadence can be the same as far as I can tell, but it can be diff too. Not really all that relevant to the question. I then tested MML using the same cptone setup and I do get JOIN and LEAVE tones. A clue that the voice-class assignment to the dspfarm is healthy. I then tested ad-hoc conference from one of the phones. Only test 3 party conference. I hear a JOIN tone when the 3rd party is added. I DO NOT hear a LEAVE tone when that third party disconnects. At this point it dawns on me what is going on. For giggles, I did another set of tests. I tested ad-hoc with 4 parties. I also tested a barge-in and then an ad-hoc add for a fourth party. If any single party (save the initiator) leaves that ad-hoc conference, a tone is played out to remaining parties (which is now 3). If one of the remaining three parties leaves (except for the conference initiator) then there is NO tone played out to the remaining two parties. Based on observed behavior, I am thinking that things are behaving as designed. The custom-cptones are associated with the dspfarm profile. When you transition from a 2-party call to a 3+ party call, you are involving the dspfarm and getting the tones. When you drop to a 2-party call, you are dropping the need for a dspfarm and the call becomes point-to-point. So, if the dspfarm was attempting to playout tones, it is no longer involved in the media path. So, the absence of the LEAVE tone seams (IMO) to be expected behavior. Assuming that one accepts that the observed behavior is expected then the question requirement to playout a tone when a party leaves is bogus. If I hit this in the real lab and the requirement says a tone must be played when the line is barged AND when the barging party leaves, I would bring it up to the proctor as a bogus requirement. The dspfarm is removed from the call at the point where the barging party leaves and is no longer in the media path. If, on the other hand, it simply says parties on the call should hear a tone when the line is barged then there is no problem. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 9:45 PM, Jason Lee wrote: I'll give it a go tomorrow. I already reverted my pod this evening. I'll be doing another lab tomorrow, so I should be able to test this put by tomorrow afternoon. Sent from my iPad On Feb 17, 2013, at 9:14 PM, Bill whl...@gmail.com wrote: I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.com wrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration
Re: [OSL | CCIE_Voice] Custom Tones
I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1 0/1/1 ! controller T1 1/0 cablelength long 0db ! controller T1 1/1 cablelength long 0db ! ! ! ! ! interface Loopback0 ip address 192.168.96.2 255.255.255.255 h323-gateway voip bind srcaddr 192.168.96.2 ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.105 encapsulation dot1Q 105 native ip address 192.168.105.1 255.255.255.0 ! interface GigabitEthernet0/0.106 encapsulation dot1Q 106 ip address 192.168.106.1 255.255.255.0 ! interface Service-Engine0/0 ip unnumbered GigabitEthernet0/0.106 service-module ip address 192.168.106.2 255.255.255.0
[OSL | CCIE_Voice] OWLE Lab 4 CME-SRST Question
In OWLE Lab 4 there is a requirement to allow 4-digit dialing to Site A and Site B from Site C, while Site C is in SRST mode. I always handle this with the following config: voice translation-rule 91051 rule 1 /^3...$/ /1408387\0/ type any international plan any isdn rule 2 /^2...$/ /1202555\0/ type any international plan any isdn voice translation-profile 91050 translate called 91051 dial-peer voice 91050 pots translation-profile outgoing 91050 destination-pattern [23]...$ port 0/3/0:15 In the solution guide, it is handled in the following manner: voice translation-profile 900 translate called 900 ! voice translation-rule 900 rule 1 // // type any international plan any isdn ! dial-peer voice 900 pots destination-pattern 9001.. port 0/0/0:15 forward-digits 11 translation-profile out 999 num-exp 2...$ 90012025552... num-exp 3...$ 90014083873... Both options achieve the desired result but I am wondering if the latter option is preferred for any technical reason. Thanks in advance, -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla ___ 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
