Re: [OSL | CCIE_Voice] Extenal call not happening.
Dharambir Kumar, What happens when you try and place a call outbound. Do you get a rapid busy or do you get your call cannot be completed as dialed. If you're getting your call cannot be completed as dialed chances are you have a css or pt problem. --Michael ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
[OSL | CCIE_Voice] Accessing SB cluster using single browser (Amdo Ngawa)
Greetings Amdo, Not being aware of your lab topology I can tell what I believe you need. You will require a connection for VMWare Management Client Connection, One for your headquarters site, one for your SB site, and one for the backbone site to effectively simulate the lab topology. I would suggest you buy more NICS (I'm using 6) for your VMWare server if it has enough Cores, Disk Space and Memory to support 4xCUCM servers, 1xCUC Server, 2xIMP Servers and 1 2008 R2 DNS Server and possibly 2-3 Windows 7 desktops. You will need four routers and 4 switches. One router for your PSTN, One for Site B and one for Site C, and One for you headquarters site. One Switch for Headquarters and one switch for your Backbone. In site B and site C you will need either internal switches or external switches. If you plan on doing Video you will need Routers (29xx's) with PVDM3's and 9971 phones or other video compatible phones. Michael Sears, CCIE(V)#38404 1. Accessing SB cluster using single browser (Amdo Ngawa) Message: 1 Date: Tue, 17 Jun 2014 10:54:43 -0400 From: Amdo Ngawa datapack...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Accessing SB cluster using single browser Message-ID: CA+d2o73vWbFQYqBN4HrMLfet0gZ-v=wy_c9rkrthd8tpjkc...@mail.gmail.com Content-Type: text/plain; charset=utf-8 Hi Folks: My Esxi server has two ethernet ports; one port connected to HQ and then other to SB. I installed CUCM 9 on both clusters. My Esxi is configured with 10.10.100.x and I can access HQ Publisher from my browser (windows 7). Is it necessary to have a separate workstation to access the SB cluster or is there any way to access it from a single workstation? Thank you and have a good one. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Buttons on 7975 IP Phones (Andy Thanh)
Minh, In response to your query regarding monitoring the line availability from your 7975 phones. Do the 7975's have expansion modules? How many lines will you need to monitor? If feasible you can add all the numbers to the 7975 phones making them shared lines. When they are in use they will turn red on everyone's phone except the person that is using a particular line. This will also restrict the use of these lines to one user at a time. Michael Sears, CCIE(V)#38404 3. Re: Buttons on 7975 IP Phones (Andy Thanh) On Mon, Jun 2, 2014 at 7:24 PM, Minh Dang dangquangm...@vnpro.org wrote: Dear group, We have some 7975s and would like to program the buttons on those to monitor the status of DID number of the sip trunk. Our users want to know if the DID numbers are available or not. Then they press the button to select the line and make the call. Please advise some ways to do this. Thank you for your time, Minh ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Toll free number translated to my desk cisco phone
Dharambir, First question I have for you is what protocol your using on the gateway, MGCP, SIP or H323. The suggestions from Justin Carney are valid suggestions on how to get this working. If your using an H.323 gateway you could just write an inbound voice translation rule and apply to dial-peer. --Michael Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) E911 Infrastructure Specialist (CER) Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, April 17, 2014 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 98, Issue 14 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Toll free number translated to my desk cisco phone (Dharambir kumar varma) 2. Re: Toll free number translated to my desk cisco phone (Andr? de Castro) 3. Re: Toll free number translated to my desk cisco phone (Justin Carney) -- Message: 1 Date: Thu, 17 Apr 2014 17:47:46 +0530 From: Dharambir kumar varma dharambi...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Toll free number translated to my desk cisco phone Message-ID: CA+iWkJTSu2n6c3Jg1tg=roo_6x5rkfpa0kzrpp9ktsyo4x-...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi all I have purchased one DID number 8000 toll free . I want when pstn user dial this number it must be forwarded to my one of desk extension(which is DID capable) can we do some translation in cucm Thanks in advance for reply -- Regards, Dharambir Kumar -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20140417/5692d90b/attachment-0001.html -- Message: 2 Date: Thu, 17 Apr 2014 11:18:47 -0300 From: Andr? de Castro aocbra...@gmail.com To: Dharambir kumar varma dharambi...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Toll free number translated to my desk cisco phone Message-ID: can9dzmeygjga7wayes9b7fq89q5x2hd0l8uvxja0wbnom3a...@mail.gmail.com Content-Type: text/plain; charset=utf-8 Hi Dharambir, I guess the siginifcant digits in your gateway configs is what you are looking for. Regards, On Thu, Apr 17, 2014 at 9:17 AM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi all I have purchased one DID number 8000 toll free . I want when pstn user dial this number it must be forwarded to my one of desk extension(which is DID capable) can we do some translation in cucm Thanks in advance for reply -- Regards, Dharambir Kumar ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc -- Andr? de Castro -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20140417/29cc9d37/attachment-0001.html -- Message: 3 Date: Thu, 17 Apr 2014 10:47:57 -0400 From: Justin Carney justin.s.car...@gmail.com To: Andr? de Castro aocbra...@gmail.com Cc: Dharambir kumar varma dharambi...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Toll free number translated to my desk cisco phone Message-ID: caex4d8yms3-kj7awqeedt-fotehoestfbjaxifmb5ty-t5j...@mail.gmail.com Content-Type: text/plain; charset=utf-8 You have several options... Either on the gateway CLI (if using H.323 or SIP, however this does not apply if your vgw is MGCP): - match the incoming DNIS on a dial peer and perform a translation to the desired internal DN Or in CUCM: - on the gateway you can apply significant digits if the # of digits aligns to you dial plan and the TF number happens to match the identical significant digits as your target IP phone DN - on the gateway you can apply a callED party transformation pattern - you could define a translation pattern to match the TF number and translate it to your target DN*
[OSL | CCIE_Voice] CCIE COLLABORATION LAB
Greetings, I've passed my written collaboration and have questions regarding studying for the LAB. I'm wondering what people are using for their home LABs to study for the LAB. Is anybody using their voice lab equipment? I've priced out the equipment on the equipment list and is about 17k, way too much. Any suggestions would be appreciated. Thank you, Michael Sears CCIE (V) 38404 ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] CCIE COLLABORATION LAB
I'm looking for dual status. Michael Sears, CCIE(V)#38404 From: Joe Tansey [mailto:joetans...@hotmail.com] Sent: Tuesday, April 15, 2014 11:42 AM To: Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CCIE COLLABORATION LAB I thought there was the option to re-characterize your Voice CCIE as a Collaboration one if you've passed the written Collab exam? Has something changed? Or you are looking for dual-status? ~Joe From: michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Date: Tue, 15 Apr 2014 16:19:51 + Subject: [OSL | CCIE_Voice] CCIE COLLABORATION LAB Greetings, I've passed my written collaboration and have questions regarding studying for the LAB. I'm wondering what people are using for their home LABs to study for the LAB. Is anybody using their voice lab equipment? I've priced out the equipment on the equipment list and is about 17k, way too much. Any suggestions would be appreciated. Thank you, Michael Sears CCIE (V) 38404 ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinchttp://www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] CUCM and CUC Demo Licenses
Chris, With version 9.1.2-1-28 or 9.x you have to use CUWL licensing not DLU's. Go to the following link: (you will need a CCO login) https://tools.cisco.com/SWIFT/LicensingUI/demoPage Go to GET DEMOSUNIFIED COMMUNICATIONS PRODUCTSCisco Unified Communications Demo License Version 9.x and later and paste the content of the License Request from the Enterprise License Manager(ELM)/Cisco Prime License Manager into the empty box: This will get you a license good for 90 days for CUCM and CUC. After burning servers CUCM and CUC and apply the license in ELM take a snapshot immediately so you can revert back like Abel suggests. Now a question for you. What equipment, routers and switches are you using in your lab? Hope this helps :) Michael Sears, CCIE 38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems So was so happy to update my home CCIE Collaboration lab to 9.1 images to be current on the new lab, as I want to sit for this in the next 90 days:) Only thing is licensing has expired and I am guessing there is no longer enough trial DLU's to support even my lab phones, clients? Even with 8.6 there were plenty to run the Voice lab, now with ELM it's killed services, thoughts? What is the best recommendation to or what have others with home labs been doing to keep your LAB Enviorment licensed without having to reinstall, as that takes FOREVER! Any help is appreciated... Chris _ Export your config and return the VMware snapshot to the moment after the installation Abel ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] how can I test these 911, 9911 dial peers to make sure they work.
Minh, What type of gateway are you using. Are you using 9 for secondary dial-tone. As mentioned earlier you should use the urgent priority for 911 or 9911 calls. If you could share more about your topology and types of equipment your using it would be helpful. It sounds like your using an H323 gateway and sending your calls to the gateway where they match a dial-peer for outgoing calls. In this case to test your emergency dialing create the following: Create a voice translation-rule (for example convert 911 to your cell phone number) Apply voice translation-rule to voice translation profile Apply voice translation profile to dial peer voice translation-rule 911 rule 1 /911/ /18005551212/ ! voice translation-profile 911 translate called 911 ! dial-peer voice 911 pots description ### EMERGENCY DIALING TEST ### translation-profile outgoing 911 destination-pattern 911 no digit-strip port 0/0/0:23 Conversely if your using an MGCP gateway just add a called party transformation mask to the route pattern in CUCM. This is how I test my CER integrations without sending calls to the PSAP. After this testing it is imperative that you place calls to the PSAP to insure the carrier is handling the calls correctly. Hope this helps :) Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, March 23, 2014 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 97, Issue 28 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: interdigit timeout, comfor noise (Ryan Maxam) 2. Re: interdigit timeout, comfor noise (Minh Dang) 3. Re: interdigit timeout, comfor noise (Ryan Maxam) 4. Re: Wireshark pcap questions for SIP and H323 call flows (Shrinivas Varanasy) -- Message: 1 Date: Sat, 22 Mar 2014 17:40:06 -0400 From: Ryan Maxam ryan.ma...@gmail.com To: Minh Dang dangquangm...@vnpro.org Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] interdigit timeout, comfor noise Message-ID: 97761eb9-160b-48b6-bfb6-cec766344...@gmail.com Content-Type: text/plain; charset=us-ascii Also, for route patterns like 911, 9911 or any other very specific route pattern, checking the Urgent Priority box in the route pattern will route the call without waiting for the interdigit timeout. Ryan Maxam Sent from my iPad On Mar 22, 2014, at 7:15 AM, Minh Dang dangquangm...@vnpro.org wrote: Thanks Andy Thanh Shrinivas. Created more dial peers and it get improved. -Original Message- From: Shrinivas Varanasy [mailto:voip...@me.com] Sent: Friday, March 21, 2014 5:57 PM To: Minh Dang Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] interdigit timeout, comfor noise If you have more than one patterns starting with 9 (like 91 , 9011 etc, specifically 911) after user press 9 it will wait for the next digit before getting dial tone. With above examples. User will not get dial-tone until he press 91 or 9011 or 9 followed by digit other than 0 or 1. You may use 8 for local calls. On Mar 20, 2014, at 6:32 AM, Minh Dang dangquangm...@vnpro.org wrote: Hi group, Our users make complaint about the delay when they press 9 to dial outside. Is there anyway to make the users feel more comfortable? Or I just adjust the interdigit timeout to 1 second? Thank you ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc -- Message: 2 Date: Sat, 22 Mar 2014 20:20:22 -0400 From: Minh Dang dangquangm...@vnpro.org To: 'Ryan Maxam' ryan.ma...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] interdigit timeout, comfor noise Message-ID: 01cf462d$b0f6f140$12e4d3c0$@org Content-Type: text/plain; charset=us-ascii Then how can I test these 911, 9911 dial peers to make sure it work? Thanks Ryan -Original Message-
Re: [OSL | CCIE_Voice] 911 on phone without DN
Mike, I do this all the time with my clients. Basically I use a dummy DN as Jeff suggested and create line CSS with partitions that only allow 911, 9911 and internal calling. You must assign a DN or you won't get dial tone. Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) E911 Infrastructure Specialist (CER) Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, March 14, 2014 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 97, Issue 21 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. 911 on phone without DN (Mike O'Nan) 2. Re: 911 on phone without DN (Jeffrey Girard) -- Message: 1 Date: Fri, 14 Mar 2014 07:56:14 -0500 From: Mike O'Nan mdona...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 911 on phone without DN Message-ID: 8qdxiwdwrp35d18wnjdx8yn9.1394801772...@email.android.com Content-Type: text/plain; charset=utf-8 Sometimes in our environment they ask to delete the extension as a person has left and don't want to hear their phone ring. I am wondering if there is a way to dial 911 when there is no DN associated with the phone. Any ideas how I can make this work?? I have roughly 200 remote sites with MGCP gw. Some using LRG...if that helps the thought process any. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20140314/9c63ae18/attachment-0001.html -- Message: 2 Date: Fri, 14 Mar 2014 13:02:42 + From: Jeffrey Girard jeffrey.gir...@girardinc.com To: Mike O'Nan mdona...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] 911 on phone without DN Message-ID: 522c88512d547748ad4a2828a23b13cc7b975...@hood.girardinc.com Content-Type: text/plain; charset=utf-8 Have you concerned simply adding a dummy DN in a partition that is not reachable by any CSS? You can use the same dummy DN as a shared line appearance on all currently un-owned phones Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mike O'Nan Sent: Friday, March 14, 2014 8:56 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 911 on phone without DN Sometimes in our environment they ask to delete the extension as a person has left and don't want to hear their phone ring. I am wondering if there is a way to dial 911 when there is no DN associated with the phone. Any ideas how I can make this work? I have roughly 200 remote sites with MGCP gw. Some using LRG...if that helps the thought process any. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20140314/2a77dac8/attachment-0001.html -- ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc End of CCIE_Voice Digest, Vol 97, Issue 21 ** ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
[OSL | CCIE_Voice] IS THIS LIST GOING TO BECOME THE COLLABORATION STUDY LIST FOR IPX?
Greetings all, I believe that this list will become the IPExpert Collaboration IE Study List. I may be wrong but that only seems logical for those of us that are pursuing the Collaboration IE LAB and refuse to give up our CCIE Voice Certification. --Michael Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) E911 Insrastructure Specialist (CER) Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, February 28, 2014 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 96, Issue 18 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: CCIE COLLABORATION STUDY GROUP(WWW.COLLABORATIONIE.COM) (Wayne Lawson) -- Message: 1 Date: Thu, 27 Feb 2014 12:28:26 -0500 From: Wayne Lawson waynelawson-...@ipexpert.com To: stevechaves9 stevechav...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE COLLABORATION STUDY GROUP (WWW.COLLABORATIONIE.COM) Message-ID: 8d70c9ad-c485-47a5-a814-bd8a77f85...@ipexpert.com Content-Type: text/plain; charset=us-ascii You're banned. Regards, Wayne A. Lawson II Founder CEO - iPexpert CCIE #5244 / Emeritus :: World-Class Cisco Certification Training Mobile: +1.810.334.1564 :: Free Videos :: Free Training / Product Offerings :: CCIE Blog :: Twitter On Feb 27, 2014, at 9:27 AM, stevechaves9 stevechav...@gmail.com wrote: Hi Friends got great forum link to join together. Lets join the study group so that we can share and-pass the lab Whoever is interested can join thought to inform all as the same way we worked in voice but unfortunately no success but surely i am hoping this time. All can use the below link to join the group whoever is interested thanks (http://collaboration-ie.com/index.php?/topic/625-collaboration-study- group/) Chey jejejeje ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20140227/03076f27/attachment-0001.html -- ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc End of CCIE_Voice Digest, Vol 96, Issue 18 ** ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Voice CCIE versus Collaboration.
Hello Chris, The instructor led Collaboration LAB is CLUS14-5844 Lab: Instructor-Led (8 hour duration) You need to register for Cisco Live in order to take it. The price for the LAB is $1095.00. After registering for Cisco Live you can register for the LAB. Hope this helps. --Michael Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice From: Chris Avants [mailto:cava...@gmail.com] Sent: Sunday, February 16, 2014 11:26 AM To: Sears, Michael (msears) Subject: Re: [OSL | CCIE_Voice] Voice CCIE versus Collaboration. What's the instructor led collaboration lab? Where can I get more details:) On Feb 15, 2014 11:24 AM, michael.se...@compucom.commailto:michael.se...@compucom.com wrote: I have decided that the Voice CCIE I worked so hard to get is to0 important to just give it over to Cisco. I will be taking the Collaboration LAB and have already passed the written. I will be using option 3. In addition I will be taking the Instructor lead Collaboration LAB at Cisco Live coming up in May in San Francisco. Option 3: Pass both the CCIE Collaboration Written Exam and the CCIE Collaboration Lab Exam and then receive a CCIE Collaboration certification in addition to the previously earned CCIE Voice certification. CCIE Voice holders who have passed the CCIE Collaboration written and lab exams will be granted both the CCIE Voice and CCIE Collaboration certifications. CCIE Voice holders with a valid CCIE Voice Written Exam or CCIE Collaboration Written Exam are eligible to register for the CCIE Collaboration Lab Exam (available starting February 14, 2014). Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) E911 Infrastructure Specialist (CER) Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.commailto:ccie_voice-requ...@onlinestudylist.com Sent: Saturday, February 15, 2014 10:00 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 96, Issue 6 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.commailto:ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.commailto:ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: RIP CCIE-Voice :-) (m george) 2. Re: RIP CCIE-Voice :-) (Abel ...) -- Message: 1 Date: Sat, 15 Feb 2014 10:52:50 +0500 From: m george m.george00...@gmail.commailto:m.george00...@gmail.com To: Abel ... midga...@gmail.commailto:midga...@gmail.com, OSL Group ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-) Message-ID: CANY2=oaehghqbys2cdp7qacatvtotthctpejuju0+++hwvv...@mail.gmail.commailto:oaehghqbys2cdp7qacatvtotthctpejuju0%2b%2b%2bhwvv...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Abel, That's not true.Last date is feb 13,2016. Read here : *Option 2*: Pass the CCIE Collaboration Written Exam and then *permanently*convert your CCIE Voice certification to a CCIE Collaboration certification between November 21, 2013 and February 13, 2016 http://www.cisco.com/web/learning/other/pop_quote.html On Sat, Feb 15, 2014 at 10:21 AM, Abel ... midga...@gmail.commailto:midga...@gmail.com wrote: If you are planning the upgrade with the written exam, is too late. Last date wast the last 14th of February. On 15/02/2014 3:40 PM, m george m.george00...@gmail.commailto:m.george00...@gmail.com wrote: Will anyone here who already passed voice lab preparing to undertake collaboration lab for 2nd CCIE title ? I have talked to many folks me my colleagues we plan to convert our titles to Collab IE with written rather than going for another hectic lab. What's your guys take on this ? What will you do ? On Sat, Feb 15, 2014 at 3:54 AM, Abel ... midga...@gmail.commailto:midga...@gmail.com wrote: Upgrading my home lab already, kind of expensive with new 29xx. But just for the knowledge sake. On Sat, Feb 15, 2014 at 7:15 AM, wilson.sam...@bt.commailto:wilson.sam...@bt.com wrote: Aha Nicolas, you have
[OSL | CCIE_Voice] Cisco MediaSense
Greetings Professionals, Anyone out there with MediaSense experience willing to do a knowledge share? I'm doing an install with CUBE and CUCM version 9.1.1 and finding the Cisco Documentation a little vague. Thank you for any information, Michael Sears ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Voice CCIE versus Collaboration.
I have decided that the Voice CCIE I worked so hard to get is to0 important to just give it over to Cisco. I will be taking the Collaboration LAB and have already passed the written. I will be using option 3. In addition I will be taking the Instructor lead Collaboration LAB at Cisco Live coming up in May in San Francisco. Option 3: Pass both the CCIE Collaboration Written Exam and the CCIE Collaboration Lab Exam and then receive a CCIE Collaboration certification in addition to the previously earned CCIE Voice certification. CCIE Voice holders who have passed the CCIE Collaboration written and lab exams will be granted both the CCIE Voice and CCIE Collaboration certifications. CCIE Voice holders with a valid CCIE Voice Written Exam or CCIE Collaboration Written Exam are eligible to register for the CCIE Collaboration Lab Exam (available starting February 14, 2014). Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) E911 Infrastructure Specialist (CER) Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Saturday, February 15, 2014 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 96, Issue 6 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: RIP CCIE-Voice :-) (m george) 2. Re: RIP CCIE-Voice :-) (Abel ...) -- Message: 1 Date: Sat, 15 Feb 2014 10:52:50 +0500 From: m george m.george00...@gmail.com To: Abel ... midga...@gmail.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-) Message-ID: CANY2=oaehghqbys2cdp7qacatvtotthctpejuju0+++hwvv...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Abel, That's not true.Last date is feb 13,2016. Read here : *Option 2*: Pass the CCIE Collaboration Written Exam and then *permanently*convert your CCIE Voice certification to a CCIE Collaboration certification between November 21, 2013 and February 13, 2016 http://www.cisco.com/web/learning/other/pop_quote.html On Sat, Feb 15, 2014 at 10:21 AM, Abel ... midga...@gmail.com wrote: If you are planning the upgrade with the written exam, is too late. Last date wast the last 14th of February. On 15/02/2014 3:40 PM, m george m.george00...@gmail.com wrote: Will anyone here who already passed voice lab preparing to undertake collaboration lab for 2nd CCIE title ? I have talked to many folks me my colleagues we plan to convert our titles to Collab IE with written rather than going for another hectic lab. What's your guys take on this ? What will you do ? On Sat, Feb 15, 2014 at 3:54 AM, Abel ... midga...@gmail.com wrote: Upgrading my home lab already, kind of expensive with new 29xx. But just for the knowledge sake. On Sat, Feb 15, 2014 at 7:15 AM, wilson.sam...@bt.com wrote: Aha Nicolas, you have a point sir. Anyway, I just wanted to make the passage of the track / version somewhat memorable that's all. No need to get serious on this now (note to myself as well) Lets get the Colloboration done.. Btw, who is attempting it on tihs forum and how you have prepared for it? Lab Gear?? Regards -- *From:* Mergenthal, Chase [chase.mergent...@bestbuy.com] *Sent:* Friday, February 14, 2014 3:01 PM *To:* Nicolas MICHEL; Samuel,W,Wilson,JKH3 R *Cc:* Online Study *Subject:* RE: [OSL | CCIE_Voice] RIP CCIE-Voice :-) It's funny you mention that, on my second or so attempt; at the end of the exam UCCX wasn't working at all... I got 100% on UCCX... -- Chase Mergenthal *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Nicolas MICHEL *Sent:* Friday, February 14, 2014 1:41 PM *To:* wilson.sam...@bt.com *Cc:* Online Study *Subject:* Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-) Wilson, I am already a CCIE in RS so I know what to expect when I am taking a CCIE exam. When you skip the UCCX task because you ran out of time and when you score report says : UCCX = 100%, to me it means complete nonsense
Re: [OSL | CCIE_Voice] Live Record.
Karen, What does your telephony-service, ephone-dn and CUE configuration look like? If it worked don't think is your configuration, maybe need to reset the CUE module. --Michael Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) E911 Infrastructure Specialist (CER) Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: Karen Johnson [mailto:karen.johnson...@yahoo.ca] Sent: Saturday, August 10, 2013 6:13 PM To: Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: Re: Live Record. hi Mike, Thanks, I just tested. And when I ended Live record, it did end recording. However when i try to call 4250 again or press LiveRecord again, it always busy. Do you know if any command i missed? K From: michael.se...@compucom.commailto:michael.se...@compucom.com michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Cc: karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca Sent: Friday, August 9, 2013 10:26:29 AM Subject: RE: Live Record. Karen, To stop the live record session you should press the live record softkey again and it will end the recording and send to voicemail. If you just disconnect the recording will continue. --Michael Message: 6 Date: Fri, 9 Aug 2013 07:27:47 -0700 (PDT) From: Karen Johnson karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] LiveRecord Message-ID: 1376058467.46146.yahoomail...@web163906.mail.gq1.yahoo.commailto:1376058467.46146.yahoomail...@web163906.mail.gq1.yahoo.com Content-Type: text/plain; charset=us-ascii hi folks. After I press Live Record and press disconnected to end conversation , why the Live Record session still stay? is this expected or any configuration we need? 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
Re: [OSL | CCIE_Voice] Live Record.
Karen, To stop the live record session you should press the live record softkey again and it will end the recording and send to voicemail. If you just disconnect the recording will continue. --Michael Message: 6 Date: Fri, 9 Aug 2013 07:27:47 -0700 (PDT) From: Karen Johnson karen.johnson...@yahoo.ca To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] LiveRecord Message-ID: 1376058467.46146.yahoomail...@web163906.mail.gq1.yahoo.com Content-Type: text/plain; charset=us-ascii hi folks. After I press Live Record and press disconnected to end conversation , why the Live Record session still stay? is this expected or any configuration we need? 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
Re: [OSL | CCIE_Voice] Ringlist.
Karen, The first thing I noticed is that your uploading the Ringlist to Publisher. If your phones are registered to the Subscriber if you have one you will have this problem. You must upload ringlist to both the publisher and subscriber. If your phones are registered to publisher then is different issue. Let me know. I've had this problem in the past and requires trouble-shooting skills to determine the root cause of problem. Michael Sears, CCIE(V)#38404, CCNA, CCNA-Voice, CCNP-Voice Cisco Certified Unified Communications Computing Systems Specialist (UCS) Cisco Certified Unified Communications Manager Express Specialist (CUCME) Certified E911 Infrastructure Specialist (CER) 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
Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect.
Karen, Tom is right on. Go to CUCM service parameter Drop Ad Hoc Conference and change from default setting Never to When Conference Controller Leaves. But does the question explicitly state that all callers should be dropped from the conference in the event the leader leaves the conference? In some cases they may want the bridge to stay open if the organizer leaves the bridge. This is a case where you have to read the question very carefully. It may not matter if the bridge stays open. 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
Re: [OSL | CCIE_Voice] AXL Password
The simplest way to approach this is to use the administrator account on CUCM and make all your passwords and usernames the same for lab purposes not a production environment. Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems Message: 1 Date: Mon, 15 Jul 2013 15:49:44 +0530 From: Dharambir kumar varma dharambi...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] AXL password Message-ID: ca+iwkjtygpn7gvnkqzekxm_v63bcksmef1qy1vnc_ndxb8_...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hi i have one cucm publisher/ and one cucm subcriber. My cisco presense server is integated with Piblisher by using security password and AXL username and password. the same Axl username /Password we are using to login CUCM through webpage. can we change this AXL username/password. if i change then is there any impact on CUCM--Presense server ... -- 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
Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect.
Well doesn't work will have to test in my lab and find solution. Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: Karen Johnson [mailto:karen.johnson...@yahoo.ca] Sent: Monday, July 15, 2013 9:08 PM To: ccie2k12; Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect. yes doesnt work From: ccie2k12 ccie2...@gmail.commailto:ccie2...@gmail.com To: 'Karen Johnson' karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca; michael.se...@compucom.commailto:michael.se...@compucom.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Sent: Monday, July 15, 2013 3:09:00 PM Subject: RE: [OSL | CCIE_Voice] Conf Meetme did not disconnect. have u tried it? Drop Ad Hoc Conference doesn't work for meet-me. Regards, From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson Sent: Monday, July 15, 2013 8:46 PM To: michael.se...@compucom.commailto:michael.se...@compucom.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect. hi Mike and Tom, Thanks, I will, just prepare if they ask me to drop. K From: michael.se...@compucom.commailto:michael.se...@compucom.com michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Cc: karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca Sent: Monday, July 15, 2013 7:10:03 AM Subject: Re: Conf Meetme did not disconnect. Karen, Tom is right on. Go to CUCM service parameter Drop Ad Hoc Conference and change from default setting Never to When Conference Controller Leaves. But does the question explicitly state that all callers should be dropped from the conference in the event the leader leaves the conference? In some cases they may want the bridge to stay open if the organizer leaves the bridge. This is a case where you have to read the question very carefully. It may not matter if the bridge stays open. 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
[OSL | CCIE_Voice] ESRST and SRST Manager.
Greetings all, I'm working a solution for a customer and wonder if anyone has any experience with SRST Manager and ESRST. Yes I've googled it and loaded the OVA, but I'm looking for someone's opinion about it who has hands on experience with it. I'm setting up 1000 sites for SRST and looking at all the options. Any feedback would be appreciated since I've never used it before. Would just plain old SRST be best? Thank you, 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
Re: [OSL | CCIE_Voice] How to configure Clocking for GW's
First of all the question itself is ambiguous as clocking occurs at the physical layer of the OSI model. The DCE side is the PSTN and the DTE side is the user side and both reside at layer 1. The question really doesn't make sense. ! Question: Take clocking for Layer 1 from Network side. Your PRI clocking of layer 2 should be user side. No clocking at layer 2 ! In answer you your query about verification use the show controller t1 0/0/0 to determine if the circuit is running clean. Another useful command is show interface serial 0/0/0:23. ! With show controller your verifying, framing, line coding, clocking and you're looking for errors such as: ! Data in current interval (90 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs ! The show interface command will show you MTU, Bandwidth, Delay, Reliability, and transmit and receive load as well as possible errors: ! R1#show interface serial 0/0/0:23 Serial0/0/0:23 is up, line protocol is up (spoofing) Hardware is DSX1 MTU 1500 bytes, BW 64 Kbit/sec, DLY 2 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulation HDLC, loopback not set Last input 00:00:13, output never, output hang never Last clearing of show interface counters never Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0 Queueing strategy: weighted fair Output queue: 0/1000/64/0 (size/max total/threshold/drops) Conversations 0/1/256 (active/max active/max total) Reserved Conversations 0/0 (allocated/max allocated) Available Bandwidth 48 kilobits/sec 5 minute input rate 0 bits/sec, 0 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec 613735 packets input, 2458376 bytes, 0 no buffer Received 0 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 613719 packets output, 2456979 bytes, 0 underruns 0 output errors, 0 collisions, 0 interface resets 0 unknown protocol drops 0 output buffer failures, 0 output buffers swapped out 1 carrier transitions Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags ! ! Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: Karen Johnson [mailto:karen.johnson...@yahoo.ca] Sent: Tuesday, July 02, 2013 11:12 PM To: Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: Re: How to configure Clocking for GW's tks Mike, very comprehensive - what is the verification command ? do we need to sh controller t1 to see if any noise on PRI line ? From: michael.se...@compucom.commailto:michael.se...@compucom.com michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Cc: karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca Sent: Monday, July 1, 2013 9:09:31 AM Subject: RE: How to configure Clocking for GW's How to set clocking for PSTN Router and Branch Routers Example using MGCP gateway: ! ! Question: Take clocking for Layer 1 from Network side. --Means PSTN is Network Side Your PRI clocking of layer 2 should be user side. --Means Branch takes clock from PSTN ! ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! ! BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! ! !Hope this helps clarify the clocking issues and configuration. ! ! Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems
Re: [OSL | CCIE_Voice] How to configure Clocking for GW's
How to set clocking for PSTN Router and Branch Routers Example using MGCP gateway: ! ! Question: Take clocking for Layer 1 from Network side. --Means PSTN is Network Side Your PRI clocking of layer 2 should be user side. --Means Branch takes clock from PSTN ! ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! ! BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! ! !Hope this helps clarify the clocking issues and configuration. ! ! 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
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 89, Issue 4
NETWORK SIDE: ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! USER SIDE BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! 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
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 89, Issue 4
When you say L1 and L2 are you talking about layer 1 and 2 of the OSI Model? Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: CISCO CCIE VOICE [mailto:ccievoic...@gmail.com] Sent: Monday, July 01, 2013 12:58 PM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: CCIE_Voice Digest, Vol 89, Issue 4 Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2 USER SIDE in that case controller t1 0/0/0 clock source line -L1 Network Side (This is by default enable no need to add it) and for L2: USER SIDE do we need to add any additional commands under serial interface 0/0/0:23 ? Thanks On Mon, Jul 1, 2013 at 9:29 PM, Sears, Michael (msears) michael.se...@compucom.commailto:michael.se...@compucom.com wrote: NETWORK SIDE: ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! USER SIDE BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! 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
Re: [OSL | CCIE_Voice] TCL applied but when calling 4000 call disconnected...
Amit, I would recommend that you verify your script. You can do so by going to the following link: ! http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html ! Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service (B-ACD) ! ! Your script is missing service app-b-acd with duplicates and other errors and your calling a non-existent service-name callq. ! You will need to know how to find this link for writing BACD scripts: Products Voice and Unified Communications IP Telephony Unified Communications Platform Cisco Unified Communications Manager Express Configure Configuration Guides Cisco Unified CME B-ACD and Tcl Call-Handling Applications Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service (B-ACD) ! See the following sections: Embedded Call-Queue and AA Tcl Scripts: Example (You can use this script and modify) Cisco Unified CME B-ACD with Drop-Through Option: Example (add parameters from the drop through example) ! In addition you don't need the following POTS dial-peer just the VoIP dial-peer will do: dial-peer voice 1000 pots service app-b-acd-aa incoming called-number 4000 ! Amit I know when I first started my journey BACD was very difficult for me and confusing until I found this link. ! Hope this helps you out. ! 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
Re: [OSL | CCIE_Voice] Translation-rule help
Regis, If I understand you correctly you're placing an international call and want to do variable digit dialing, most likely for either SRST or for an H323 gateway. In this case where your using 9 for the secondary dial tone and 011 for international calling you wouldn't use a voice translation to remove the 9. The translations would be used to mark the traffic as international and send out the calling number as E164. The example below indicates how I use translations for international dialing on H323 gateway and SRST. Since 9011 is an explicit match it will automatically be dropped and you add the 011 back in using the prefix 011 as stated by Regis. ! voice translation-rule 4 rule 1 /^4...$/ /+1888404/ type any international plan any isdn ! voice translation-rule 14 rule 1 // // type any international plan any isdn ! voice translation-profile international translate calling 4 translate called 14 ! dial-peer voice 9011 pots translation-profile outgoing international destination-pattern 9011T port 0/1/0:23 prefix 011 ! 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
Re: [OSL | CCIE_Voice] Overlapping dial plan
I would recommend you review the Dial Plan SRND. I just did a roll-out with 500 remote sites so you can imagine how many overlapping digits there were. I used 10 digits for Voice Mail with 5 digit internal dialing using variable digit dialing with site codes. There's several ways to do it. The SRND goes over all the dial plans for best practice and overlapping dial plans. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/dialplan.html#wp1150620 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
Re: [OSL | CCIE_Voice] BLUE Screen on SC Phones.
Karen, I really can't explain it, technically I can't explain why this happens. My screens on SC phones turned bright blue with no information and the phones were registered. Finally, in desperation I tried changing the phones back to the default background image and it worked. I also put other background images on the phones and it worked also. Good Luck. Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Infrastructure Specialist [3-2-2013 3-02-38 PM] [UCS SPECIALIST1] Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: Karen Johnson [mailto:karen.johnson...@yahoo.ca] Sent: Thursday, June 13, 2013 10:13 AM To: Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: Re: BLUE Screen on SC Phones. ah will test that in moment and update you. but why is that ? From: michael.se...@compucom.commailto:michael.se...@compucom.com michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Cc: karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca Sent: Wednesday, June 12, 2013 7:24:54 PM Subject: BLUE Screen on SC Phones. Karen, I experienced the same thing when studying for the IE in my home lab. The solution for me to get rid of the very blue screen was to check the background image and revert to the standard image. Then my phones showed showed correct information. Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Insrastructure Specialist Compucom Systems Western Region inline: image001.jpginline: image004.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] BLUE Screen on SC Phones.
Karen, I experienced the same thing when studying for the IE in my home lab. The solution for me to get rid of the very blue screen was to check the background image and revert to the standard image. Then my phones showed showed correct information. Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Insrastructure Specialist Compucom Systems Western Region ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Spanning Tree Portfast
Recommended for HQ Switch Access Ports: interface GigabitEthernet1/0/20 description ==HQ Phone 1 switchport access vlan xxx switchport mode access switchport voice vlan xxx switchport priority extend cos 0 spanning-tree portfast Recommended for use with HWIC-4ESW: interface FastEthernet0/3/0 description ==R2 Phone 1 switchport trunk native vlan xxx switchport mode trunk switchport voice vlan sxx switchport priority extend cos 0 switchport trunk encapsulation dot1q Portfast Command: Typical STP convergence time is around 50 seconds by default, so basically every port takes around 50 seconds to initialize and be in say forwarding state, this is a lot of time and is not needed to be spent to check for loops in your network especially if you know there won't be any network loops through that port. The port bypasses the listening, learning, filtering steps when using portfast increasing convergence times. It is advisable to utilize this command on access ports in the older versions of code. The newer versions of code have the command spanning-tree portfast trunk for use on trunk ports. I would recommend that you use spanning-tree portfast on server ports and phone ports, but it has no effect on trunked ports. Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Insrastructure Specialist 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
Re: [OSL | CCIE_Voice] LAB GRADED WITH ALL 0%.
Ravi, All 0% does sound a little strange. I've never heard of anyone getting all 0% before. Possibly your configurations were wiped out somehow. If you feel that your lab was not graded correctly or there was a problem with grading call the following number and open a case with the CCIE Certifications Team. 1.800.553.6387 #4 #1 --ms Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Infrastructure Specialist Dear All, I got CCIE lab exam for the first time and I am really unsatisfied about the marks I got. unfortunately I was not able to complete my lab properly but I have configured few sections in good manner. Unfortunately I didn't knew that I removed a cable from a phone and replugged due to phones were not registered with CUCM correctly. when I got a result sheet it was really strange and they given me 0% for each and every section. So is there any way to appeal the exam to cisco and get another chance to attain this valuable certification. I know that I am technically competent. but now I am totally depressed. 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
Re: [OSL | CCIE_Voice] CUE NTP - Best practices
Hey MJ, DBreplication issues are very time consuming to fix and troubleshoot and out of my four attempts I didn't have any issues with replication. I know several others who are on their 7th and 8th attempts and they too have NOT had DBreplication issues, especially with regards to NTP. There are two ways that I'm aware of to check the status, ReportsDatabase Summary and RTMT, although sometimes they are incorrect. I did ask the proctor in RTP if there was a DBreplication issue was that considered part of the lab. He responded that you are expected to fix it although I haven't experienced it. A good question to the group I think would be just to ask if anyone has had DBreplicaation issues while sitting the lab. Also, maybe there is a DBreplication guru out there that can speak to DBreplication issues better than myself. In my lab with severe DBreplication issues it was usually quicker to revert to a good snapshot then spend much time on trying to fix it. There is no need to restart any services on the Publisher or Subscriber when you add NTP to the Publisher. The Subscriber will get its time from the Publisher and the Publisher will get its time from HQ1 loopback if that's what the question calls for. No need to reset anything. In the case of the HQ1 loopback it gets it time from the PSTN. In my lab I used the atomic clocks in Boulder, CO and my configuration on my PSTN router for NTP looked like this: PSTN# ntp source Loopback1 ntp master 1 ntp update-calendar ntp server 132.163.4.101 prefer burst iburst ntp server 132.163.4.102 burst iburst ntp server 132.163.4.103 burst iburst This assumes that you have internet access from the PSTN router. To trouble shoot ntp I use the following: PSTN-WAN#debug ntp ? adjustNTP clock adjustments all NTP all debugging on core NTP core messages eventsNTP events packetNTP packet debugging refclock NTP refclock messages PSTN#debug ntp all You can also use utils diagnose test to check for any dns issues or ntp issues related to DBreplication. admin:utils diagnose test test - ntp_reachability : Passed test - ntp_clock_drift : Passed test - ntp_stratum: Passed If you do have a problem in the LAB with DBreplication better to fix it sometimes DB repair works than try to determine the root cause. Otherwise you won't finish your lab. I didn't focus much on DBreplication when I took my attempts, maybe I should have, but I passed without any issues with NTP or DBreplication issues. Well not sure if any of this helps, but good luck in your studies. You may want to NOT focus so much on this one topic although your questions are valid and the information is good to know. --ms Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Insrastructure Specialist [3-2-2013 3-02-38 PM] [UCS SPECIALIST1] Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: sanity insanity [mailto:networksanitytoinsan...@gmail.com] Sent: Monday, June 10, 2013 6:38 PM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: CUE NTP - Best practices hi Guys, Still waiting to hear back Thanks again On Sun, Jun 9, 2013 at 10:22 PM, sanity insanity networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com wrote: hi Guys, Thanks for your replies. If the DB replication does go out of sync is there any troubleshooting step that can be executed . Hope it does not take too much time to sync since we would then loose the time to complete other tasks in the process. 1) Also does restart of NTP on Publisher and subscriber fix all issues related to DB replication caused due to NTP? 2) Is it recommended that we restart NTP service on Publisher and subscriber after adding NTP server to the Publisher? 3) What can be done to determine the cause of the issue? -MJ On Wed, Jun 5, 2013 at 10:55 PM, Sears, Michael (msears) michael.se...@compucom.commailto:michael.se...@compucom.com wrote: 1) which is the best way to bind the service module to the router is it required to bind it to the loopback of the router or the interface voice VLAN. Answer: It depends on the question it could be either the Loopback or the Voice VLAN it should say in the LAB. In my practice I use both alternating, one lab I'll use the Voice VLAN and then on another run through the lab I'll use the Loopback. Note that people have had issues using the Loopback so best to figure it out before setting the LAB. 2) On my HQ router I am configuring it to sync with a back NTP server. I am also required to sync the CUCM publisher with the loopback of this HQ router . Here are my questions... a) I have the following configuration on the HQ router... ntp source Loopback0 ntp server 177.26.1.100 HQ ntp configuration: ntp source
Re: [OSL | CCIE_Voice] CUE NTP - Best practices
1) which is the best way to bind the service module to the router is it required to bind it to the loopback of the router or the interface voice VLAN. Answer: It depends on the question it could be either the Loopback or the Voice VLAN it should say in the LAB. In my practice I use both alternating, one lab I'll use the Voice VLAN and then on another run through the lab I'll use the Loopback. Note that people have had issues using the Loopback so best to figure it out before setting the LAB. 2) On my HQ router I am configuring it to sync with a back NTP server. I am also required to sync the CUCM publisher with the loopback of this HQ router . Here are my questions... a) I have the following configuration on the HQ router... ntp source Loopback0 ntp server 177.26.1.100 HQ ntp configuration: ntp source Loopback0 ntp server 177.26.1.100 burst iburst 3) On CUCM PUB I have added the NTP server and given it the ip address of loopback of HQ router. Now my questions are... 1) Is a reboot of CUCM required? No 2) If the ntp does not sync will my DB replication break after a few hours? Answer: If the NTP Server does not synchronize need to determine why. When I've had issues in the past with NTP not synchronizing it did not break dbreplication over a period of hours. Although I suppose there's that possibility. Michael Sears, CCIE(V)#38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration
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
Re: [OSL | CCIE_Voice] Tranfer to Voicemail button for CME
MJ, Your configuration looks ok but I think you might be forwarding the * to VM. I would recommend the following: voice translation-rule 10 rule 1 /.*\(4...\)$/ /\1/ voice translation-profile vm-redirect translate redirect-called 10 ! dial-peer voice 4220 voip translation-profile outgoing vm-redirect destination-pattern 4220$ session protocol sipv2 session target ipv4:(IP Address of Service Engine) dtmf-relay sip-notify codec g711ulaw no vad ! ephone-dn 3 number 4001 label 4001 (Don't use Label will Default to 4001) description +85224044001 name SiteC1 (This gets listed in the local directory) 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 Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Insrastructure Specialist 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
Re: [OSL | CCIE_Voice] LAN QOS (CCIEing)
CCIEing, See page 111 of SRND CAT2970(config)#mls qos srr-queue output cos-map queue 1 threshold 3 5 ! Maps CoS 5 to Queue 1 Threshold 3 (Voice gets all of Queue 1) Remember Queue 1 is the priority Queue by default. You should download the SRND and use it when configuring LAN QoS. It makes it a cut and paste task. http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/Enterprise_QoS_SRND.pdf Michael Sears, CCIE(V)#38404 ___ For more information regarding industry leading 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
First I used the SRND for LAN QoS and I believe that to be the best way, but I'm sure there are many other flavors. I found using the SRND for LAN QoS to be quick and easy, page 107. It's very efficient way to do it and eliminates mistakes. For WAN QoS I used auto qos voip trust or auto qos voip depending on the question wording. In addition I did QoS first and I know some people disagree with that, but it worked for me speed wise using the device based approach. I did not use the SRND for WAN QoS, but agree is a good reference to have since it's on the desktop of the lab. I used the 95% rule as well as removing the rmon commands. I passed. I agree with most of the comments especially read the question very carefully. Michael Sears Designing and Implementing Cisco Unified Communications on Unified Computing Systems -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, April 09, 2013 7:09 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 86, Issue 58 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Dial-peer Preference (Josh Petro) 2. Re: WAN QoS Calculations (Barrera, Hugo) -- Message: 1 Date: Tue, 9 Apr 2013 20:43:26 -0400 From: Josh Petro josh.pe...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Dial-peer Preference Message-ID: ca+m12bxfzo1q61o5jv8bh-h4e3+bkndxbs2xvhd3md713g4...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, I know this is a silly question, but it's been bugging me. Does the lab script care if your dial-peers have preference 0 (no preference configured) or Preference 1 / 2 configured as it is below? I realize that preference 0 (no preference) would be the dial-peer used if there is a match on both peers, but would the below be graded differently? I'm used to assigning a preference and leaving no preference always make me feel like I missed something. dial-peer voice 100 pots preference 1 destination-pattern 9.[2-9]... port 0/0/0:15 ! dial-peer voice 200 pots preference 2 destination-pattern 9.[2-9]... port 0/0/0:15 Thanks much. Josh -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130409/c53eaa82/attachment-0001.html -- Message: 2 Date: Wed, 10 Apr 2013 01:08:35 + From: Barrera, Hugo hugo.barr...@nexusis.com To: William Bell b...@ucguerrilla.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] WAN QoS Calculations Message-ID: 22e3eddc-0e84-41f5-a668-f64e1a8d9...@nexusis.com Content-Type: text/plain; charset=windows-1252 I will review thanks Bill. Regards, Hugo On Apr 9, 2013, at 5:27 PM, William Bell b...@ucguerrilla.commailto:b...@ucguerrilla.com wrote: 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 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 :) 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: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto: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
Re: [OSL | CCIE_Voice] HQ as ntp master Required
Greeting Guys, QUESTION: If HQ-router is ntp reference for branches AND HQ router has some external ntp reference. Should you configure ntp master on HQ router? RECOMMENDATIONS: This is an issue I toyed with for months, do I use ntp master on HQ or no. Finally, after several discussion on the forum, and testing in my lab I would make the following recommendation. 1. Do NOT use ntp master command on any of your routers or switches it lowers the stratum level by a factor of 7-10. Try it in your lab and you'll see. NOT recommended. 2. Do USE on HQ the ntp source loopback0 command but not on other routers or switch. 3. Use the command ntp server [ip address] burst iburst to increase your synchronization times. With experimentation you'll see that the ntp master command basically makes your time on the other routers less accurate by increasing the stratum level to instead of like 2 to 8 or more. Cheers, Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Specialist 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
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
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 Practice, Practice, Practice. Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Specialist 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
Re: [OSL | CCIE_Voice] HQ as ntp master Required
Use ntp server [ip address] burst iburst on all devices, including switch or at least I configure ntp on switch. If I understand your question accurately, I don't configure ntp reference for anything but SIP phones. Not required. I do configure a DTG for all DP's, HQ, SB and SC and even MOH and GK if it calls for it. If lab does not require locations like no CAC involved I don't configuration locations at all. --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 [3-2-2013 3-02-38 PM] Designing and Implementing Cisco Unified Communications on Unified Computing Systems From: Vikky Kumar [mailto:vikkyne...@gmail.com] Sent: Tuesday, April 02, 2013 12:41 PM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com; Josh Petro (josh.pe...@gmail.com) Subject: Re: HQ as ntp master Required Hi Michael, Where do where you prefer # ntp server [ip address] burst iburst ? i mean, hq only or braches only or all locations Also do you recommend to use ntp reference and location time zone in CUCM datetime group or just create DT group and specify timezone for different branches located in different timezone. Regards, Vikky On Tue, Apr 2, 2013 at 5:47 PM, Sears, Michael (msears) michael.se...@compucom.commailto:michael.se...@compucom.com wrote: Greeting Guys, QUESTION: If HQ-router is ntp reference for branches AND HQ router has some external ntp reference. Should you configure ntp master on HQ router? RECOMMENDATIONS: This is an issue I toyed with for months, do I use ntp master on HQ or no. Finally, after several discussion on the forum, and testing in my lab I would make the following recommendation. 1. Do NOT use ntp master command on any of your routers or switches it lowers the stratum level by a factor of 7-10. Try it in your lab and you'll see. NOT recommended. 2. Do USE on HQ the ntp source loopback0 command but not on other routers or switch. 3. Use the command ntp server [ip address] burst iburst to increase your synchronization times. With experimentation you'll see that the ntp master command basically makes your time on the other routers less accurate by increasing the stratum level to instead of like 2 to 8 or more. Cheers, Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Specialist Designing and Implementing Cisco Unified Communications on Unified Computing Systems inline: image003.jpg___ For more information regarding industry leading 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
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 [3-2-2013 3-02-38 PM] 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.commailto: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 Practice, Practice, Practice. Michael Sears, CCIE(V)#38404 Cisco Certified Unfied Communications Computing Systems Specialist E911 Specialist Designing and Implementing Cisco Unified Communications on Unified Computing Systems ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com inline: image001.jpg___ For more information regarding industry leading 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] hugo.barr...@nexusis.com
Greetings Hugo, What it really comes down too. What is the question and what are results you're being asked to provide? Without knowing what the results are it is difficult to answer accurately appropriately. Michael Sears CCIE (V) 38404 Message: 1 Date: Sun, 24 Mar 2013 16:01:11 + From: Barrera, Hugo hugo.barr...@nexusis.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME Presence Message-ID: 0bebb671101b304eb5b3b3d587c6d75231b08...@nxca05exch10.nexusis.com Content-Type: text/plain; charset=iso-2022-jp 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130324/a0c85e3c/attachment-0001.html -- Message: 2 Date: Sun, 24 Mar 2013 16:27:18 + From: Leslie Meade leslie.me...@lvs1.com To: Barrera, Hugo hugo.barr...@nexusis.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Presence Message-ID: f64719604b4e6f41bdbb2af38e7609f449a95...@lvscgyex03.longviewsystems.com Content-Type: text/plain; charset=iso-2022-jp It will be stated in the lab which way? but if they do not it is up to you Leslie Meade From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Barrera, Hugo Sent: Sunday, March 24, 2013 9:01 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME Presence 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130324/f6c4761d/attachment-0001.html -- Message: 3 Date: Sun, 24 Mar 2013 13:03:00 -0500 From: Brad McAllister b...@bdmcomputers.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] cRTP 2-4 bytes Message-ID: cagnbhgc1attstcdwxtwdj9ar5alf1jfqkvrfcpvezue87nx...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, It seems that most calculations I see use 2 bytes when header compression is turned on. On Occasion I also seem 4 bytes used. If I understand correctly, 4 bytes should be used if udp checksum is enabled. My questions is: What enables/disables udp checksum? Is it safe to always use 2 bytes for this value? Thanks, - Brad -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130324/decff277/attachment-0001.html -- Message: 4 Date: Sun, 24 Mar 2013 14:18:02 -0400 From: William Bell b...@ucguerrilla.com To: Barrera, Hugo hugo.barr...@nexusis.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Presence Message-ID: 3c147625-f7b9-4d2c-b1fd-c5c409e58...@ucguerrilla.com Content-Type: text/plain; charset=iso-2022-jp 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
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE.
Greetings Vikas, Can you call 4101 from either HQ or SB or do you get a rapid busy or if you call from HQ to SB to 4101 and no other calls up does the call reroute over the PSTN. If you get rapid busy or if your call immediately reroutes over the PSTN that isn't right. You could have a locations issue from what you explain I'm a little confused. Also it appear that you haven't put into place a rsvp bandwidth statement which is required to perform rsvp calls. On HQ and SC need the following statements: interface Serial0/1/0:0.102 point-to-point ip rsvp bandwidth 112 (to allow 4 calls) Michael Sears Compucom Systems Western Region Senior Infrastructure Solution Consulting Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax: +1.978.863.0740 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, March 22, 2013 7:34 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 85, Issue 78 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: HQ and Branch1 phones cannot call CUE (Vikky Kumar) 2. Re: SRST to voicemail without Alternate Extension (Leslie Meade) -- Message: 1 Date: Fri, 22 Mar 2013 16:17:09 +0300 From: Vikky Kumar vikkyne...@gmail.com To: Sears, Michael (msears) michael.se...@compucom.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE Message-ID: ca+4dtjfua+8z2fgff9ajxulz5n5zfalxwez9kiyuigyty3g...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Williams, Thanks for your email. Earlier message on the calling hq-phones was not enough bandwidth when i tried to place the call to 4220, after reload whole lab now its Ring out and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 phone call goes smooth no prob upto 4 calls after that , not enough bandwidth, Rerouting on display of phone and call goes via PSTN network I think this is expected ok This happens vice versa, sc to hq. I configured sdspfarm transcode session 4 under telephony-service. my configuration is CME-SRST My problem is HQ and BR1 Phones can't call CUE VM Pilot . as mentioned above I am using RSVP, its configures exactly how you have explained working ok. Automated Alternate Routing in Service Parameters = TRUE. HQ and BR2 you need to configure MTP resources = Done configure Location RSVP setting to mandatory between HQ --BR2 = Done I have not configured ip precedence under sccp, does that matters, I have added now but situation remain same. i dont have serial uplink its ethernet only, but works as per my testing 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-card 0 dspfarm dsp services dspfarm ! interface Loopback0 ip address 172.16.30.254 255.255.255.0 ip pim dense-mode ip ospf network point-to-point ! interface FastEthernet0/0 ip address 192.168.1.4 255.255.255.0 duplex auto speed auto ip rsvp bandwidth 112 ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 172.16.30.253 255.255.255.0 service-module ip default-gateway 172.16.30.254 ! interface Vlan31 ip address 172.22.30.254 255.255.255.0 ip pim dense-mode ! interface Vlan32 ip address 172.32.30.254 255.255.255.0 ! ! sccp local Loopback0 sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-conf associate profile 1 register sc-mtp associate profile 2 register sc-xcode ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 8 associate application SCCP ! Regards, Vikky On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) michael.se...@compucom.com wrote: Greetings Vikas, First, do you get any message on the calling phone, like not enough bandwidth when
Re: [OSL | CCIE_Voice] UCCX agent routing and script
Can you call from HQ Phone 1 to 4101? By chance do you have RSVP configured? I had same issue and in my case RSVP wasn't working. If you have it setup the easy way to test is to turn off mandatory in locations and try to call from HQ Phone 1 to 4101 again. If this works try UCCX again. This may not be your problem, but I had exactly the same problem when RSVP wasn't working correctly and it sounds like UCCX is trying to forward the call so this may not be a problem with UCCX but another issue outside of UCCX call routing, i.e. being able to forward calls through HQ. Hope this helps. Michael Sears CCIE Voice 38404 ___ For more information regarding industry leading 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 on CME Not Working
Hello Hugo, As mentioned earlier do you have an ad-hoc conference dial-peer setup? dial-peer 4001 pots number A4001 no-reg both description CbargeAdHoc conference ad-hoc In addition do you have conference hardware, no privacy and privacy-on-hold (if you using privacy button) setup under telephony-service? telephony-service conference hardware no privacy privacy-on-hold You can do a show sccp to make sure your conference and xcode are actually registered to CME. Michael Sears CCIE 38404 ___ For more information regarding industry leading 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
Greetings Vikas, First, do you get any message on the calling phone, like not enough bandwidth when you try and place the call. What happens when you try and dial 4 digit extension of BR2 phone from HQ or BR1 phone, Rapid busy, dead air, not enough bandwidth on display of phone, Rerouting? First this could just be a simple case of not configuring sdspfarm transcode session 10 under telephony-service. It all depends on your configuration and if SRST is involved. I just read in your header HQ and BR1 Phones can't call CUE VM Pilot so the following may not help and your problem is local on BR2 router. How are you trying to trigger CAC? Are you in fact trying to configure RSVP based CAC or plain simple locations based CAC? It is unclear what it is your trying to accomplish. If your trying to perform RSVP Based CAC how many calls do you want to permit. Let's say for example you want to permit 4 calls then reroute across the PSTN using AAR. In this case you would need to turn on Automated Alternate Routing in Service Parameters. Then on HQ and BR2 you need to configure MTP resources. In addition you need to configure Location RSVP setting to mandatory between HQ --BR2. You also need to configure on HQ and BR2: interface Serial0/1/0:0.102 point-to-point ip rsvp bandwidth 112 (to allow 4 calls) ! sccp local Loopback0 sccp ccm [ip address] identifier 1 priority 1 version 6.0 sccp ccm [ip address] identifier 2 priority 2 version 6.0 sccp ccm [ip address] identifier 3 priority 3 version 6.0 sccp ip precedence 3 sccp ! sccp ccm group 1 description sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 1 register sc-mtp associate profile 2 register sc-xcode associate profile 3 register sc-conf registration timeout 3 registration retri 3 keepalive timeout 3 keepalive retri 3 switchback met imm switchback interval 15 switchover met imm ! dspfarm profile 1 mtp description dspfarm profile 1 mtp codec pass-through (I always use codec pass-through some do not it seem to work both ways) codec g729r8 rsvp maximum sessions software 10 associate application SCCP ! dspfarm profile 2 transcode description dspfarm profile 2 transcode ! dspfarm profile 3 conference description dspfarm profile 2 conference ! To reroute the calls once the upper limit of four is reached you will need to do the following: Create aar group Create pt-aar Create css-aar containing pt-aar Create two route lists one for HQ and one for BR2 Create two route patterns using the appropriate partitions and apply correct route list Calls will by default use the External Phone Number Mask to reroute the call by matching the route pattern, which is assigned to appropriate route list directing the call out the appropriate gateway. You might want to paste in your configuration so that everyone can have a look :) Michael Sears CCIE 38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Can't reach GUI for CUE.
Hello Jason, It is strange that you can http to GUI when using another interface, like the Interface Voice VLAN, for example, but I would through this out there anyway. Do you have all the CME/CUE files downloaded to right directories in flash? Do you have the following setup: ip http server ip http path flash:/gui ip http authentication local telephony-service web admin system name administrator password cisco create cnf You can also try to telnet to 10.32.60.253 80 to see if you can hit port 80 using telnet from the HQ router. Although I have not had the problem myself I've seen it time and time again on the list. Have you tried using http://[CMEIPADDRESS]. What does your telephony-services configuration look like? Specifically, what IP are you using for your source address in telephony-service and port number. The loopback or Voice VLAN and port 2000? From all your results this just does not appear to be a routing issue. Especially since you can ping the CUE address from local and other routers. Also, as someone already suggested use ip ospf network point-to-point. When you try and reach CUE using HTTP://10.32.60.253/ what is the error. Do you get HTTP - 404? M ichael Sears CCIE 38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Can't reach GUI for CUE.
I started thinking about you issue and have another question for you. What does your show ip route look like in CUE? Michael Sears Compucom Systems Western Region Senior Infrastructure Solution Consulting Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax: +1.978.863.0740 -Original Message- From: Sears, Michael (msears) Sent: Tuesday, March 19, 2013 8:44 PM To: ccie_voice@onlinestudylist.com Cc: 'scubajas...@gmail.com' Subject: Can't reach GUI for CUE. Hello Jason, It is strange that you can http to GUI when using another interface, like the Interface Voice VLAN, for example, but I would through this out there anyway. Do you have all the CME/CUE files downloaded to right directories in flash? Do you have the following setup: ip http server ip http path flash:/gui ip http authentication local telephony-service web admin system name administrator password cisco create cnf You can also try to telnet to 10.32.60.253 80 to see if you can hit port 80 using telnet from the HQ router. Although I have not had the problem myself I've seen it time and time again on the list. Have you tried using http://[CMEIPADDRESS]. What does your telephony-services configuration look like? Specifically, what IP are you using for your source address in telephony-service and port number. The loopback or Voice VLAN and port 2000? From all your results this just does not appear to be a routing issue. Especially since you can ping the CUE address from local and other routers. Also, as someone already suggested use ip ospf network point-to-point. When you try and reach CUE using HTTP://10.32.60.253/ what is the error. Do you get HTTP - 404? M ichael Sears CCIE 38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [MVA NOT WORKING] Mobile Voice Access not working since many
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 of the CUCM Pub dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 5 pots service cmm incoming called-number 3300 no digit-strip 2) here is the MVA service url ! application service cmm http://ip address of the CUCM Pub:8080/ccmivr/pages/IVRMainpage.vxml ! 3) I am stripping 3033300 coming from pstn to last 4 digits using a translation-rule on the voice-port level . That is 3033300 becomes 3300 when it reaches CUCM. 4) On CUCM in the service parameters... Enable Mobile Voice access is set to True Mobile voice access number is 3300 Matching caller id with Remote Destination is Partial Match Number of digits of Caller ID Partial Match is 7 5) The Mobility softkey has been added for on hold and connected at the softkey template level and applied to the phone ( SB PH1) 6)At the User SB phone 1 I have enabled Enable Mobility and Enable Mobile Voice Access also selected the MAC address of the phone 7) Created a Remote Dest profile and selected user id of sb ph1 and the correct calling search space for the phone 8) Added a Remoted Destination number of 9525 9) Also went to device phone and selected the Owner User ID of SB Ph1 10) Cisco Unified Mobile Voice Access Service is running on both Sub and Pub on CUCM Questions : 1) I now dial from the pstn line 9525 on the pstn phone to 3033300 . The prompt I get asks me for a pin . I enter 1 and the call drops . I Even tried entering 12345 ( which the pin for user SB Phone 1) and still the call drops after the prompt. Anything wrong the above config? Anything missing in the config ? Any suggestions? 2) I am strip the number to last 4 digits ( as in step 3) . Is this correct procedure? 3) There is also no QOS setup in the config for now . Anything related to Bandwidth here? Please help! -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
[OSL | CCIE_Voice] DHCP WITH ONE POOL USING STATIC MAPPING.
Requirements: I want to configure a DHCP server on a router . The requirement is that just 1 DHCP pool is required for the phone. I am also asked to assign ip addresses of 14.10.66.13 and 14.10.66.14 for my phones. Solution: #conf t Enter configuration commands, one per line. End with CNTL/Z. ROUTER1(config)#ip dhcp database flash:origin.txt ROUTER1(config)#no service dhcp ROUTER1(config)#service dhcp ROUTER1(config)#do more origin.txt *time* Mar 14 2013 06:54 AM *version* 4 !IP address Type Hardware address Lease expiration VRF !IP address Type Hardware address Interface-name !IP address Interface-name Lease expiration Server IP address Hardware address Vrf *end* Open Notepad and modify origin.txt as below: *time* Mar 14 2013 06:55 AM *version* 4 !IP address TypeHardware addressLease expiration 142.102.66.13 /24 id 010024142EFF10 infinite 142.102.66.14 /24 id 016C504DDACC3D infinite *end* Don't forget the mask or it won't work and no dots are required in the Hardware Address. Copy the file to Publisher using tftp. Download the file to Router flash using tftp from Publisher. no ip dhcp database flash:origin.txt ip dhcp excluded-address 142.102.66.1 142.102.66.12 ip dhcp excluded-address 142.102.66.15 142.102.66.254 ip dhcp pool voice origin file flash:origin.txt option 150 ip [CUCM or CME IP Address] depends on if you're doing CUCM or CME default-router [ip address of Voice VLAN] Hope this helps Michael Sears CCIE (V) #38404 ___ For more information regarding industry leading 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] VM in SRST
To make VM work in SRST one method is to mask the calling number, i.e., on the hunt pilot to . At times this method may conflict with other requirements. You can also utilize alternate extensions in UC if the question does not say you can't use alternate extensions to accomplish this. Hope this helps. Michael Sears CCIE (V) #38404 1. VM in SRST (CISCO CCIE VOICE) Message: 1 Date: Wed, 6 Mar 2013 12:00:13 +0300 From: CISCO CCIE VOICE ccievoic...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VM in SRST Message-ID: cabpd02qegoruwbq3dmgklf_x0_xnitcbeshyyq9yh_61p7n...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 HI All When the Branch-1 is in SRST Proper Mode (call-manager-fallback) When i am trying t press voice mail button its playing system greeting not the user greeting ,its there any thing i need do on CUCM in order to play user greeting when the phones are in SRST.. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] RSVP a big problem (sanity insanity)
--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
Re: [OSL | CCIE_Voice] One button login UCCX
You need to go the application user rm and associate the agent phones to this user. --ms Michael Sears CCIE (V) 38404 Message: 2 Date: Wed, 06 Mar 2013 22:10:21 +0530 From: singh singh8...@in.com To: ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] One button login UCCX Message-ID: 1362588021.c00193e70e8e27e70601b26161b4a...@mail.in.com Content-Type: text/plain; charset=utf-8 I have one button login set for my uccx agents and have verified that the agent id and password for the users association of rmuser with the phones resource group contains the agentshowever I am seeing the following error on one button login...Unable to log you in due to conf error ( phone is not associated withRM JTAPI Provider user ID) contact your system admin for helpwhat else to check?singhGet Yourself a cool, short @in.com Email ID now! -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130306/f5ed7540/attachment-0001.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CCIE VOICE PASSED
Amit, Thank you for your congratulations. In order to pass the CCIE Voice LAB I followed a self-established protocol to follow when performing the LAB. It took me 4 attempts which is nothing I'm ashamed of and all attempts were a learning experience in developing my Strategy to actually pass this monster. First, briefly read the lab and determine what it is your be asked to do focused on the switch and gateways. I use the Device Approach. I do tasks on the gateways and switch first including QoS tasks that may or may not be on the LAB exam according to the Voice blueprint available on the Cisco Website. I would strongly recommend having strong QoS feature knowledge and practice it daily along with all of the other Layer 1, 2 and 3 technologies so that you can perform tasks quickly and efficiently. Once I leave the Switching and Gateways I don't have to go back to them. I then move on to CUCM configuration and all the associated tasks required to complete CUCM as quickly as possible. Speed plays a big part in the real LAB. Then on to Unity Connection and all the associated tasks. Then on to UCCX. I then spend any remaining time troubleshooting and testing, several times if time allows, fixing mistakes which you will make. After three attempts I performed 2-3 practice labs from, INE, IPExpert and Cisco 360 a day for about 45 days until my fourth attempt and passing. Any non-NDA breaking questions welcomed. Michael Sears CCIE (V) 38404 From: Amit Sharma [mailto:aryan231...@gmail.com] Sent: Sunday, March 03, 2013 10:12 PM To: Sears, Michael (msears) Subject: Re: [OSL | CCIE_Voice] CCIE VOICE PASSED many congrats michael! Can you share your tips to us so that we can go for getting our ccie number? !thanks On Sun, Mar 3, 2013 at 6:37 PM, michael.se...@compucom.commailto:michael.se...@compucom.com wrote: Well I finally cleared the LAB, what a relief. I would like to thank the people on the list for all the posted questions, output and responses, also INE, Mark Snow and IPExpert, Vic Malhi for all their assistance on my journey. I would also wish the best for those who are still striving to attain this goal as it is quite challenging. Michael Sears CCIE VOICE #38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading 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] Queries in relation to Route Plan
1) On site B I am doing the digit strip on the CUCM and not the gateway is the right way to get this done ? Everyone develops their own strategy for where they perform the digit strip. My strategy is to use predot on the dial-pattern so that the 9, 91 and 9011 is striped from the phone display and it only shows the actual number without the 9, 91 and 9011. For the H323 gateway I add the 9, 91 or 9011 back in on the route list and the mgcp gateway requires no modifications. The dial peers I create on the H323 gateway all begin with 9 with the exception of 911. 2) when the wan is up and operational I have the dial-peer configured as ... dial-peer voice 4 pots destination-pattern .T port 0/0/0:23 forward-digits all I would not recommend using this approach as you can't mark the type and plan of the outbound calls. In addition your going to want to add the + back in for international calls since the H323 gateway will automatically strip the + for international calls. I create dial-peers for 911, subscriber, national, international, inbound VOIP calls to CUCM and SRST and use the appropriate voice translation-rule and profile, i.e., national or international depending on requirements. The use of one dial-peer won't allow you to mark traffic which will be required as your calls will go out the gateway as unknown, unknown. This is in reference to the H323 gateway. 3) In srst mode I will need separate dial-peers for the digit strip. Currently I am creating separate dial-peers for srst . Would you see this as the right approach? Absolutely create dial peers for SRST, subscriber, national, and international requirements and mark the traffic type and plan using voice translation-rules and profiles and apply the appropriate profile to the dial-peers so you can send the appropriate number of digits and type and plan. Also consider how you're going to route calls to UC voicemail so that you get the appropriate response from UC 4) For srst mode I am not using the calling and called party type and number plan but when wan is up I have it set on call-manager. Is the calling and called party number type and plan required in srst mode? Yes the calling and called party is required in SRST and the phones should meet the requirements of the question. Number of digits to send and calling and called party type and plan. 5) I have route patterns set as the following on the CUCM ( callmanger) for my site B ( h323 gateway) for emergency , local , long distance and International as the following 911 local route group 9.[2-9]XX--- local route group ( strip predot) 91.[2-9]XX[2-9]XX -- local route group( strip predot) 9011.! --- local route group ( strip predot) The dial-plan is going to become a lot more complex than just sending everything out one gateway. You need to consider TEHO and redundant calls that will utilize other gateways for a single route pattern. You also need to develop a strategy as to where you will perform your digit manipulation. I do all my digit manipulation on the route list. Your route patterns are solid as far as the MGCP gateway goes and depending on how you write your dial-peers in H323 gateway are possibly not the most efficient way to do this. I write my dial peers for subscriber, national, international in the H323 Gateway using 911, 9[2-9].., 91[2-9]..[2-9].. prefix 1, 9011T prefix 011. 911 requires no digit manipulation. 6) Use of Standard Local Route List for same route patterns for MGCP and H323 gateways This is how I do it and perform marking on the route patterns. This is a good methodology in my opinion. You just have to make sure to use the appropriate partition on the route patterns and it works great. I would suggest using the Standard Local Route List whenever or where ever you can for most efficient call routing with the least number of route lists. I'm not saying this is the right or wrong approach, but has been effective for me. There are many different ways of performing the dial plan and it can become quite complex. I would suggest in your practice to make local for both MGCP and H323 redundant where H323 gateway backs up the MGCP gateway. I would further suggest that in your practice you have national calls go out the opposite gateway first as a TEHO national to subscriber call. For international calls use both gateways one backing up the other. I have found this to be a good practice technique for developing good skills with the dial-plan. I would further suggest that you use the Standard Local Route List whenever possible in your dial-plan. Hope this clears a few things up for you. Michael Sears CCIE VOICE #38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
[OSL | CCIE_Voice] CCIE VOICE PASSED
Well I finally cleared the LAB, what a relief. I would like to thank the people on the list for all the posted questions, output and responses, also INE, Mark Snow and IPExpert, Vic Malhi for all their assistance on my journey. I would also wish the best for those who are still striving to attain this goal as it is quite challenging. Michael Sears CCIE VOICE #38404 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] GATEKEEPER ISSUE
Greetings All. I'm having an unusual issue with my Gatekeeper which I have not seen before. My gatekeeper is on the mgcp HQ router. When I do show gatekeeper end I only see GK_Trunk_1, the publisher and CME. The Subscriber is missing. I have reset the trunk and gatekeeper many times. Although the database shows 412's and 2's I performed a DB repair all. I have tried everything to get GK_Tunk_2, the subscriber to come up but it won't. Anyone out there experience this issue and did you find a resolution? If so please respond ASAP as my lab is in 9 days and if this happens to me in the lab I'd like to know what I have to do to fix it. Any input appreciated. Thank you. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] GATEKEEPER ISSUE
Thank you much. You were right on. Somehow I set the device pool incorrectly I'm just upset with myself for not checking this, but thank you much was driving me crazy. Hopefully, I won't make this stupid mistake in my lab attempt. Oh and thanks for responding so quickly. --ms Michael Sears -Original Message- From: Nicolas MICHEL [mailto:mcl.nico...@gmail.com] Sent: Wednesday, February 20, 2013 6:10 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GATEKEEPER ISSUE Hi Michael Check that the device pool associated to the trunk got a Call Manager group with both Pub and Sub. Beside of that I don't know Regards PS: Never ask to answer ASAP please :) We are like you and owe nothing but help if we can :) Nic Le Wednesday, February 20, 2013 1:43:06 PM, michael.se...@compucom.com a écrit : Greetings All. I'm having an unusual issue with my Gatekeeper which I have not seen before. My gatekeeper is on the mgcp HQ router. When I do show gatekeeper end I only see GK_Trunk_1, the publisher and CME. The Subscriber is missing. I have reset the trunk and gatekeeper many times. Although the database shows 412's and 2's I performed a DB repair all. I have tried everything to get GK_Tunk_2, the subscriber to come up but it won't. Anyone out there experience this issue and did you find a resolution? If so please respond ASAP as my lab is in 9 days and if this happens to me in the lab I'd like to know what I have to do to fix it. Any input appreciated. Thank you. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE
Greetings Joe, Could you post or send me a copy of your complete configuration for Site C. Sure sounds like a transcoding issue but who knows. Do calls complete and roll to CUE VM? From Site A? What happens when calls roll to VM get rapid busy??Is transcoder get invoked when you place call through GK? What does the GK dial-peer look like on Site C? --ms Michael Sears Compucom Systems Western Region Senior Consultant Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax:+1.978.863.0740 [Description: Description: ccnp_voice_sm] inline: image001.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] AAR ISSUES/BUSY SIGNAL
Hugo, I'm trying to understand as I don't have that lab book. Is HQ mgcp and BR1 mgcp or h323? Michael Sears Compucom Systems Western Region Senior Consultant Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax: +1.978.863.0740 On Nov 14, 2012, at 7:08 PM, Barrera, Hugo hugo.barr...@nexusis.com wrote: Hi, ** ** I am doing AAR, Lab 6 Volume 1, but I just can?t get it to work correctly. The following is my programming task list and what I witnessed: ** ** Call flow = HQ Phone 2 x5002 calls BR1 Phone 2 x1002 ** ** 1. Enabled AAR globally in service parameters 2. Created my AAR-PT and AAR-CSS o The AAR-CSS contains the AAR-PT and Internal-PT 3. Both my phones have their corresponding Ext Phone Mask (10 digits) 4. Created my (1) AAR Group named = HQ-BR1-AARGroup o Prefix for this AAR group is ?91? 5. Assigned the above AAR group at the line and device level of both phones 6. Built out separate AAR Rout Patterns pointing to new Route List ** ** ** ** Troubleshooting: I ran my debugs on both gateways and could see the call from CUCM to the HQ-RTR I could also see the incoming call from PSTN to BR1-RTR however I still received a busy? ** ** Any ideas 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
[OSL | CCIE_Voice] Unified FX Problems.
Greetings, For help with Phone View contact: http://www.unifiedfx.com/ in the lower right hand corner of page click on Questions. Talk to Stephen Welsh stephen.we...@unifiedfx.com if you can he's the expert and has keep me going through the ups and downs of Phone View which works perfect for me. You can also email them at: email. m...@unifiedfx.com and I'm sure they will be able to assist you with any problems. Michael Sears Hi Guys I am facing some issues with Unified FX application to get Screenshots of the phones. I can remote manage the phones alright but I am simply not able to view their screens. Here is what I have done: 1. Created an app userpvadmin with Server Monitoring, EM Authentication Tab Sync User Groups. 2. Created an end user pview with Standard CTI Enabled. 3. Associated all phones with pview end user. Here is the error that I receive when I try to see screenshot: Command (Cmd:Screenshot) sent to device (MAC) using thread (0) with response (XML Error response from phone) I am using the Version 2.1.37 which is the latest one. Please let me know if anyone has faced similar issues. Thanks Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voiceview
Voice View is not supported on the AIM Module. Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Saturday, October 20, 2012 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 80, Issue 35 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Voiceview Express and CUCM integration - message playback issues (Peter Simmons) -- Message: 1 Date: Sat, 20 Oct 2012 09:45:45 +0100 From: Peter Simmons pe...@grayrigg.com To: Kevin Spicer ke...@kevinspicer.co.uk Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Voiceview Express and CUCM integration - message playback issues Message-ID: 508264b9.5070...@grayrigg.com Content-Type: text/plain; charset=iso-8859-1; Format=flowed Kevin, This is on my lab here at home. I've trawled the bug toolkit briefly for anything obvious, but didn't find anything immediately that jumped out at me. These are the versions. Phones: 7965Gs Phone load SCCP45.8-4-1S CUCM Version: 7.0.1.11000-2 AIM CUE Version: 7.0.6 Router: 2811 IOS 12.4(24)T4 I'll do some traces on CUE today and see what that shows up, if anything, then next up I plan to see if it works using CME integration, that'll be for next week though :-) Many thanks for coming back to me on this, your input is appreciated. regards Peter On 10/20/2012 9:03 AM, Kevin Spicer wrote: Is this on proctorlabs or your own kit? If your own kit maybe a bug in the specific version, did you check.bug toolkit. Also, which model phones are you using? On 20 Oct 2012 08:48, Peter Simmons pe...@grayrigg.com mailto:pe...@grayrigg.com wrote: Dan, I appreciate you coming back to me on this. I've done that already as part of the testing (Sorry, this got missed off in the original message). I've tried adding ALL the various CTI permissions to the cue jtapi user, adding all possible permissions, and just leaving it at the required values. I've pretty much done what is stated in this document (old but still OK for CUCM 7.x/CUE7.x as far as I can see): http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel2_3/cliadmi n/ch_vview.html along with the CUE/CUCM integration following this document: http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configu ration_example09186a0080289ef0.shtml No change whatever I try, I'm assiduously resetting everything (including CUCM several times) between tests just in case. Today I'll try some traces on the CUE module for a while and see if this draws anything else out I haven't seen already. Again, thankyou for your feedback - I appreciate it. regards Peter On 10/19/2012 11:36 PM, Dan Quinlan (daquinla) wrote: Try adding standard CTI enabled to the jtapi user. I've seen issues with the control all devices in the past. DQ Dan Quinlan, CCIE #36129 daqui...@cisco.com mailto:daqui...@cisco.com On Oct 19, 2012, at 1:09 PM, Peter Simmons pe...@grayrigg.com mailto:pe...@grayrigg.com wrote: Dear all, I have run into problems trying to get Voiceview message playback working with Call Manager (not CME - please note!) I have done the following: 1) Checked CUE license is correct for CCM integration 2) Enabled VoiceView on CUE 3) Created a phone service on CUCM pointing to the CUE URL ( http://CUE-hostname/voiceview/common/login.do) 4) Assigned phone service to users 5) Associated CUE jtapi user with the relevant phones 6) Assigned CUE jtapi user correct permissions (mainly allow CTI control of all devices) 7) Reset phones and CUE CUE VMX works just fine, all normal functions are working as expected, and I can leave and retrieve VMX without issues via the handset. I can log in to the Voiceview service and look at the messages in the inbox, and this session shows up on the CUE (using the show voiceview sessions CLI command) - so I know the URL in the CUCM phone
Re: [OSL | CCIE_Voice] LAN QOS QUESTION
Krishna, I added the mls qos srr-queue output dscp-map commands so that the cos-map and dscp-map match with equivalent values. This is not a requirement as your question is stated. In addition for the link to the router you need: mls qos trust dscp On the server ports: mls qos trust dscp On the phone ports: mls qos trust cos mls qos trust device cisco-phone This is assuming there are no other stated requirements for marking traffic requiring class-maps and policy-maps and service policies. Michael Sears From: Krishna [mailto:vinayak_...@yahoo.com] Sent: Sunday, October 14, 2012 6:59 PM To: Sears, Michael (msears) Cc: Online Study Subject: Re: LAN QOS QUESTION michael, thanks for your input on this query.. i am using 3750 switch provided by proctor labs.. I am well aware lan qos queuing and thresholds , but not sure where should i have to keep those values unless if they are explicitly stated. adding to that question, i assume i need to put mls qos trust dscp for the traffic coming from router to switch isn't it?? I see some guys putting as mls qos trust cos which i feel incorrect since layer 2 to laye3 mappings were done using cos-dscp mapping. please correct me if i am wrong. thank you krishna. From: michael.se...@compucom.commailto:michael.se...@compucom.com michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Cc: vinayak_...@yahoo.commailto:vinayak_...@yahoo.com Sent: Sunday, October 14, 2012 4:50 PM Subject: RE: LAN QOS QUESTION On a Cisco 3750 -- I used the Enterprise_QoS_SRND [1] pages 105 through 112 to put together the following QoS configuration to meet your stated requirements. I would recommend you follow the SRND when writing your QoS configuration it's pretty much cut and paste. Requirements: assign cos 5 to priority queue assign cos 3 4 to queue 2 cos 4 exceed 60% should be dropped Solution: mls qos mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos srr-queue output cos-map queue 1 threshold 3 5 mls qos srr-queue output cos-map queue 2 threshold 3 3 mls qos srr-queue output cos-map queue 2 threshold 1 4 mls qos srr-queue output dscp-map queue 1 threshold 3 46 mls qos srr-queue output dscp-map queue 2 threshold 3 24 mls qos srr-queue output dscp-map queue 2 threshold 1 32 mls qos queue-set output 1 threshold 2 60 100 100 100 Note how cos 3 and 4 are both in queue 2 but in different thresholds. Then since cos 4 is in threshold 1 it is marked down to 60% to discard cos 4 above 60%. Hope this helps. Keep in mind QoS will vary depending on switch type. INE has some great Videos on understanding LAN QoS and the relationships between Queues and Thresholds and related configurations. http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book.html What switch type are you using? Michael Sears Message: 1 Date: Sat, 13 Oct 2012 18:40:10 -0700 (PDT) From: Krishna vinayak_...@yahoo.commailto:vinayak_...@yahoo.com To: Online Study ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lan qos- cos 5 threshold 1 in queue 1 doesn't show in running config Message-ID: 1350178810.93468.yahoomail...@web164602.mail.gq1.yahoo.commailto:1350178810.93468.yahoomail...@web164602.mail.gq1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 hi guys, i was wondering whether i am doing right way of doing lan qos or not ?? the requirements are assign cos 5 to priority queue , cos 3 4 ?to queue 2 with 60% exceed of cos 4 should be dropped. so here is my configuration for that mls qos? mls qos srr-queue output cos-map queue 1 threshold 1 5 mls qos srr-queue output cos-map queue 2 threshold 2 3 mls qos srr-queue output cos-map queue 2 threshold 1 4 mls qos queue-set output 2 threshold 3 60 100 100 272 when i issued show run | i ?mls commands, i see every ?mls qos command except the cos 5 which is assigned to q1 t1. ?Is my approach is correct in dealing this question correctly?? does it matter whether we assign cos values to t1 or t2 or t3 in the queues??? your input is much appreciated. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CERTIFICATION SUPPORT PHONE NUMBER
Contact: Career Certifications and Support 800.553.6387 #4 #1 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] UNITY CONNECTION HELP[
I just finished burning my UC Server and setting up the Phone System and I'm getting the following error. Failed to send message to remote AXL server. Please check error log for more details. Has someone seen this before and where is the log file? Any assistance greatly appreciated haven't seen this one yet. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Please Ignore: UNITY CONNECTION HELP[
-Original Message- From: Sears, Michael (msears) Sent: Thursday, October 11, 2012 8:47 PM To: ccie_voice@onlinestudylist.com Subject: UNITY CONNECTION HELP[ I just finished burning my UC Server and setting up the Phone System and I'm getting the following error. Failed to send message to remote AXL server. Please check error log for more details. Has someone seen this before and where is the log file? Any assistance greatly appreciated haven't seen this one yet. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] H323 Voip Interface commands comment.
Bill, In reference to using or not using: h323-gateway voip interface (Used for Gatekeeper Configurations) The h323-gateway voip interface command dictates the source address that being used to communicate with gatekeeper. Unfortunately some are under the misunderstanding that this must me the loopback add which of course we know not to be true. h323-gateway voip bind srcaddr (Used to build the H323 Gateway) Sets the source IP address to be used for this H323 gateway. The ip address attribute indicates the address to be used as the source IP address for the gateway. Use this command for the interface that contains the IP address to which you want to bind for all H323 traffic. Caveat: I had an associate that argued vehemently that the h323-gateway voip interface command was required when setting up the H323 Gateway and explained that he had it up and running in his lab. I looked at his lab and sure enough the command was there and NOT registered to his gatekeeper. After further inspection I realized he had NOT enabled the gateway command on his H323 gateway. So you can get away with using it, not correct, by not enabling the gateway command or just do it the right way. Thanks Bill for all the valuable information regarding the use of these commands. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUCME in SRST mode IOS Version 15 / Toll Fraud Prevention.
Raynard, http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/feature/guide/srst8_1.html I recently had a client that upgraded all their routers to 3900's Series routers and along with that of course 15.x code. SRST in original configuration stopped working with the 15.x code. We were seeing debugs similar to these and others (debug voice ccapi inout): *Aug 14 19:54:32.507: %VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected): IEC=1.1.228.3.31.0 on callID 3 GUID=AE5066C5883E11DE8026A96657501A09. Cisco now enables IP Address Trusted Authentication by default. This was in an environment where two routers were being used as SRST gateways with crossbar H323 voip route between the routers in this MGCP to accommodate cross router calling while in SRST. The following configuration resolved our issue. voice service voip ip address trusted list ipv4 172.19.245.1 ipv4 172.19.247.1 ipv4 172.19.243.1 ipv4 171.19.245.1 ipv4 171.19.10.1 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip Adding trusted lists in this scenario resolved our SRST problem. No configuration changes were made with the exception of add the trusted list, in this case, we added all IP Interfaces to the trusted list on each router including the loopbacks. Hope this helps. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [UCCX AGENT PHONES SAY NOT READY.
Need a little help with UCCX. My IPPA service comes up on the phone but won't let me change the status to ready. It flash's registering very quickly then goes back to not ready. I've scrubbed google.com and haven't come up with anything. This is happening on two different gateways and two types of phone, 7965 and 7962. Same behavior on both, on both gateways and phones are registered with call manager I've tried removing the phones from rm and putting the back. Also have reset CTI Manager and have reloaded the server. No luck. Have also scrubbed the configuration for errors and everything looks good. Any help appreciated. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] BACD AUDIO PROBLEM.
Greetings Sanjay, http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, September 14, 2012 11:40 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 79, Issue 42 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Vol1 Lab 9a -BACD cannot hear audio ? (Sanjay P) 2. Re: Vol1 Lab 9a -BACD cannot hear audio ? (Kevin Spicer) -- Message: 1 Date: Fri, 14 Sep 2012 17:07:36 +0100 (BST) From: Sanjay P sp1...@yahoo.co.uk To: OSLGroup ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol1 Lab 9a -BACD cannot hear audio ? Message-ID: 1347638856.37034.yahoomail...@web132306.mail.ird.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Hi, Using PL Pod 29. [ using 7962 hardware phones on VPN link] ? Having trouble? with Vol1 lab 9a BACD ? I cannot hear? the welcome prompt when the BACD aa service is invoked. I can see dialpeer 3500 ?hit to invoke service aa?and then some output from debug voip application script? but no audio is played?, I ?then fast-busy after 15 secs. ? [ what is the difference between paramspace english location flash://bacdprompts/? --- written Vol1 solution v1800 paramspace english location flash:bacdprompts?--- on Viks Vol1 walkthrough ? I have tried both (reload in between) and have the same outcome as below] ? ? My? abbreviated config? is as below ? .. voice service voip ?allow-connections h323 to h323 ?allow-connections h323 to sip ?allow-connections sip to h323 ?allow-connections sip to sip ?no supplementary-service h450.2 ?no supplementary-service h450.3 ?h323 ? ras rrq ttl 120? margin 15 ? no call service stop ?sip ? registrar server ? ..voice translation-profile BACD ?translate redirect-called 3500 ! ! voice-card 0 ! ! application ?service queue flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl ? param queue-len 15 ? param aa-hunt10 3006 ? param number-of-hunt-grps 2 ? param aa-hunt2 3210 ? param queue-manager-debugs 1 ?! ?service aa flash:bacdprompts/app-b-acd-2.1.2.2.tcl ? paramspace english index 1 ? param number-of-hunt-grps 2 ? param handoff-string aa ? paramspace english language en ? param max-time-vm-retry 2 ? param aa-pilot 3500 ? paramspace english location flash://bacdprompts/ ? param second-greeting-time 60 ? param welcome-prompt _bacd_welcome.au ? param call-retry-timer 15 ? param voice-mail 53002 ? param max-time-call-retry 90 ? param service-name queue ? ? telephony-service ?no auto-reg-ephone ?em logout 0:0 0:0 0:0 ?max-ephones 5 ?max-dn 10 ?ip source-address 10.10.110.3 port 2000 ?url services http://10.10.202.2/voiceview/common/login.do ?url authentication http://10.10.202.1/CCMCIP/authenticate.asp? ?load 7960-7940 P00308000500 ?load 7962 SCCP42.8-3-3S ?time-format 24 ?date-format dd-mm-yy ?max-conferences 8 gain -6 ?moh music-on-hold.au ?transfer-system full-consult ?transfer-pattern .T ! ? dial-peer voice 3500 voip ?service aa ?destination-pattern 3500 ?session target ipv4:10.10.110.3 ?incoming called-number 3500 ?dtmf-relay h245-alphanumeric ?codec g711ulaw ?no vad ! dial-peer voice 3600 voip ?translation-profile outgoing BACD ?destination-pattern 3600 ?session protocol sipv2 ?session target ipv4:10.10.110.3 ?dtmf-relay sip-notify ?codec g711ulaw ?no vad ! !BR2-RTR#sh flash: | inc .au? 6??? 21658 Jul 17 2012 03:11:34 bacdprompts/en_bacd_allagentsbusy.au 7??? 83291 Dec 18 2008 13:49:52 bacdprompts/en_bacd_disconnect.au 8??? 63055 Dec 18 2008 13:49:54 bacdprompts/en_bacd_enter_dest.au 9??? 37952 Dec 18 2008 13:49:54 bacdprompts/en_bacd_invalidoption.au 10? 496521 Dec 18 2008 13:50:00 bacdprompts/en_bacd_music_on_hold.au 11? 123446 Dec 18 2008 13:50:02 bacdprompts/en_bacd_options_menu.au 12?? 42978 Dec 18 2008 13:50:04 bacdprompts/en_bacd_welcome.au 13?? 34794 Dec 18 2008 13:50:04 bacdprompts/en_bacd_xferto_operator.au 16?? 14618 Jul 21 2012 03:14:48 bacdprompts/en_thanks.au ? ? Sep 14 19:54:30.942: //-1/D4D6BA388049/DPM/dpMatchPeersMoreArg: ?? Result=SUCCESS(0) ?? List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=3500 Sep 14 19:54:30.962: //-1/D4D6BA388049/DPM/dpAssociateIncomingPeerCore: ?? Calling Number=3002, Called Number=3500, Voice-Interface=0x0, ?? Timeout=TRUE, Peer Encap
Re: [OSL | CCIE_Voice] iDivert/DND in SRST
SRST requirements for a given scenario may include retaining the functionality of the phone as it was in non-SRST mode. In this scenario you need to use telephony-services. The way to accomplish this and retain the functionality is by using the DND softkey which you have to setup under an ephone-template and apply to the phones as follows. ! ephone-template 1 softkeys ringing Answer Dnd ! ephone 1 device-security-mode none mac-address 0024.142E.FF10 ephone-template 1 type 7965 button 1:1 2:3 ! ephone 2 device-security-mode none mac-address 0021.55D5.3962 ephone-template 1 type 7965 button 1:2 2:4 ! Michael Sears From: Ramy Abdelrahim ramyoth...@hotmail.commailto:ramyoth...@hotmail.com Date: Tuesday, August 7, 2012 3:11 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] iDivert/DND in SRST Dear All, When the phone is registered to UCM it has iDivert softkey button to transfer a call to VM while ringing. When this site goes into SRST, iDivert is not there. Do I have to preserve this feature in SRST? And if it's the case then how? Can anyone help on this? Regards, Ramy ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed
In order for your Remote Destination Number to be recognized for example you RDN is one of the numbers from the PSTN 5551212 but the call in coming into the gateway as 800.555.1212 and is h323 gateway apply a inbound voice translation-rule and profile on the mva dialpeer to strip the 800 from the call. In order for calls to complete use the calling search space of the phone you are using for mva as the rerouting calling search space and insure that there is a route pattern that matches the number your dialing. The way to test is dial the same number from the phone you have setup for mva and number should complete. If it does then you're missing your rerouting calling search space. If it doesn't complete you don't have call routing setup correctly. Also be sure that you have enabled the mva service in service parameters and have entered the mva number there. Michael Sears From: ccielabrat ccielab...@gmail.com Date: Tuesday, August 7, 2012 2:09 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed To All, I'm hoping the group can help me understand the call flow for an MVA call. I'm able to call into the MVA pilot number , have my remote destination number recognized and be prompted for my PIN and to dial . But I get the message Your call can not be completed as dialed for anything I try to call. I understand that the number configured under the mobile voice access page is used as an anchor , as per Vik's vlecture, but I'm unclear what device is referenced regarding CSS and what should and should be reachable. Can anyone please help get closure on this last piece of the puzzle. -Lab Rat ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Beggining Advice Help
Below is a URL which is an Expansion of the CCIE Voice Lab v3.0 Exam Topics (Blueprint). You will need to know these technologies for the LAB Exam. http://ciscovoiceguru.com/wp-content/uploads/2011/01/CCIE-Voice-Expanded-Blueprint.pdf I too am a Voice CCIE candidate. I would strongly recommend investigating INE, i.e., Mark Snow teachings as well as IPExpert, Vic Malhi teachings as well. In addition I would recommend that you look into building your own rack if you can afford it. It works much better than using Rack Rentals. Good luck to you in your efforts towards the CCIE Voice LAB. Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, August 05, 2012 5:26 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 78, Issue 19 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: [CCIE Beggining Advice Help] (Kadambari) 2. Passed (steven moran) 3. Re: Passed (muhammad nouman) -- Message: 1 Date: Sun, 5 Aug 2012 14:52:34 -0700 From: Kadambari kbeel...@yahoo.co.in To: Dulip Ravindra duliprb2...@yahoo.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [CCIE Beggining Advice Help] Message-ID: 733244.47997...@smtp124-mob.biz.mail.ac4.yahoo.com Content-Type: text/plain; charset=us-ascii Dulip, I just started my ccie voice. Pls let me know if we can tall about study plan and strategy -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120805/b9f6ae16/attachment-0001.html -- Message: 2 Date: Mon, 6 Aug 2012 09:06:25 +1000 From: steven moran smoran...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Passed Message-ID: cafjqmcjks1o3rr8_u0luwrc_cihp-nhvueq9w4jvda9v0oh...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 just to let you all know I finally passed the exam, thanks for all the help and replies. If anyone in the Sydney, Australia area is interested in buying a full CCIE Voice Equipment lab plus may extra's please let me know. Best regards and good luck to all Steven Moran -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120806/8990a63e/attachment-0001.html -- Message: 3 Date: Sun, 5 Aug 2012 16:26:24 -0700 (PDT) From: muhammad nouman nouman_n...@yahoo.com To: steven moran smoran...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Passed Message-ID: 1344209184.83519.yahoomail...@web161702.mail.bf1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Congratulation Steven From: steven moran smoran...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, 6 August 2012 9:06 AM Subject: [OSL | CCIE_Voice] Passed just to let you all know I finally passed the exam, thanks for all the help and replies. ? If anyone in the Sydney, Australia area is interested in buying a full CCIE Voice Equipment lab plus may extra's please let me know. ? Best regards and good luck to all ? Steven Moran ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120805/6d42d598/attachment.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 78, Issue 19 ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ntp master- is it necessary
The only situation in which I use the ntp master command is in a situation where for example HQ is providing clock for Servers and HQ clock is synchronizing with an external reliable source. In my lab I synchronize my PSTN with the Boulder Atomic Clock and synchronize my HQ router with the PSTN. HQ loopback provides clock for Servers, Switch, CUE and other branches. The following is what I use to accomplish this. Other than this scenario I do not use the ntp master command. ntp server 10.1.1.1 ntp source loopback0 ntp master Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, August 05, 2012 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 78, Issue 14 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Switch QOS query (Justin McIntyre) 2. Re: ntp master- is it necessary (Bruno Nonogaki) 3. Re: Switch QOS query (Bruno Nonogaki) 4. Re: ntp master- is it necessary (Justin McIntyre) -- Message: 1 Date: Sun, 5 Aug 2012 09:10:00 -0400 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Switch QOS query Message-ID: 0aed4c84-05d7-4f60-b91f-eeca67495...@blackbox.com Content-Type: text/plain; charset=us-ascii So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your requirements. Have the priority-queue enabled but then also give queue 1 30% bandwidth. If priority-queue out is enabled then this over-rides the bandwidth command for that queue. I know you had some other questions as well specifically about how to drop certain traffic if a queue were 80% full. My suggestion to you would be to review Vik Mahlis QOS blog on the IPEXPERT website. Go to blog.ipexpert.com and select the voice blog on the left. Then look for the QOS section. I think this will clear up most of your questions and get you on your way. Thanks, Justin McIntyre This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. -- Message: 2 Date: Sun, 5 Aug 2012 10:14:44 -0300 From: Bruno Nonogaki brun...@gmail.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Message-ID: CAP_RLdVVgHfN=fGytf6w7K5n5-nS6g3JnRiZfM5=tgsxozp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello Krishna, Yes, you are right. ntp master is not required. If you do ntp master, it may synchronize with its internal clock. It is a big mistake a lot of people do, including me before the OWLE Bootcamp, which I really recommend. Regards, Bruno On Sun, Aug 5, 2012 at 2:37 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, i see some guys posts on ntp master command on the hq router ... i was wondering why one would be needing ntp master command when it is already being synchronized with external ntp server ntp master will infact mess up the time if not configured correctly since ntp master takes the stratum from the hardware(device) and be careful when putting the command ntp master .. if it is required then it is advised to keep the stratum number high compared to the extrenal ntp server... please correct me guys if i m wrong precisely, i felt that ntp master command is not required if that device is synchronized with external ntp server.. any comments on my advice is much appreciated... thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML
Re: [OSL | CCIE_Voice] ntp master- is it necessary
Thanks Dan. I'm not trying to make it work yes it works fine without the command, but instead I'm trying to replicate what Cisco might be looking for on the lab. Frankly I don't know what they are looking for in these types of scenarios but don't want to take the chance of losing points because they think the command should be there. Just cause it works doesn't mean you will get the points. Michael Sears -Original Message- From: Dan Quinlan (daquinla) [mailto:daqui...@cisco.com] Sent: Sunday, August 05, 2012 11:27 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Michael, all, You do NOT need the NTP master command on a router whenever that router is synchronizing with an external source, even if other devices are to synchronize with that router. Michael - in your config, you can remove the NTP master command and everything will synchronize with the HQ router just fine. Ntp master should really only be used when there is no external clock. DQ d...@cisco.com Sent from my iPhone On Aug 5, 2012, at 12:38 PM, michael.se...@compucom.com michael.se...@compucom.com wrote: The only situation in which I use the ntp master command is in a situation where for example HQ is providing clock for Servers and HQ clock is synchronizing with an external reliable source. In my lab I synchronize my PSTN with the Boulder Atomic Clock and synchronize my HQ router with the PSTN. HQ loopback provides clock for Servers, Switch, CUE and other branches. The following is what I use to accomplish this. Other than this scenario I do not use the ntp master command. ntp server 10.1.1.1 ntp source loopback0 ntp master Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, August 05, 2012 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 78, Issue 14 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Switch QOS query (Justin McIntyre) 2. Re: ntp master- is it necessary (Bruno Nonogaki) 3. Re: Switch QOS query (Bruno Nonogaki) 4. Re: ntp master- is it necessary (Justin McIntyre) -- Message: 1 Date: Sun, 5 Aug 2012 09:10:00 -0400 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Switch QOS query Message-ID: 0aed4c84-05d7-4f60-b91f-eeca67495...@blackbox.com Content-Type: text/plain; charset=us-ascii So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your requirements. Have the priority-queue enabled but then also give queue 1 30% bandwidth. If priority-queue out is enabled then this over-rides the bandwidth command for that queue. I know you had some other questions as well specifically about how to drop certain traffic if a queue were 80% full. My suggestion to you would be to review Vik Mahlis QOS blog on the IPEXPERT website. Go to blog.ipexpert.com and select the voice blog on the left. Then look for the QOS section. I think this will clear up most of your questions and get you on your way. Thanks, Justin McIntyre This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. -- Message: 2 Date: Sun, 5 Aug 2012 10:14:44 -0300 From: Bruno Nonogaki brun...@gmail.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Message-ID: CAP_RLdVVgHfN=fGytf6w7K5n5-nS6g3JnRiZfM5=tgsxozp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello Krishna, Yes, you are right. ntp master is not required. If you do ntp master, it may
[OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs
If the question specifically states plan unknown and type unknown I would interpret that to mean set the type to type any unknown plan any unknown. But there must be more to the question. For example are we talking about called or calling party or both. Can we get more information on the question. Is this MGCP or H323? Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, July 30, 2012 1:06 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 77, Issue 55 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: ANI/DSNI TON on Proctor Labs (Rrcrumm) 2. Re: Prompt recording (Rrcrumm) 3. Re: Prompt recording (Jason Aarons (AM)) -- Message: 1 Date: Mon, 30 Jul 2012 11:38:14 -0700 From: Rrcrumm rrcr...@yahoo.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs Message-ID: f3878a37-0374-45b9-9230-619eec35d...@yahoo.com Content-Type: text/plain; charset=us-ascii Hi It is with MGCP gw Sent from my iPhone On Jul 30, 2012, at 9:55 AM, Krishna vinayak_...@yahoo.com wrote: if it is h323 gateway, i will create a translation rule and apply at the dial-peer... if it is mgcp gateway do it on the call manager route list detail level... thank you krishna. From: Randall Crumm rrcr...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Sent: Monday, July 30, 2012 11:33 AM Subject: [OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs Hello, I have noticed some different behaviors and was wondering what you recommend If the question asks for plan unknown and type unknown should you set to 1. plan unknown and type unknown or 2. plan isdn type unknown or 3. plan call manager type call manager For me I have tried the above and it seems like call manager/call manager is what is working(actually allowing the call to go through). It goes through as unknown/unknown Any thoughts? Thanks! Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120730/d05c6c96/attachment-0001.html -- Message: 2 Date: Mon, 30 Jul 2012 12:00:40 -0700 From: Rrcrumm rrcr...@yahoo.com To: Mann Chaddha mann.chad...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Prompt recording Message-ID: 3f8330ca-e532-4f40-87de-17a1a5ff4...@yahoo.com Content-Type: text/plain; charset=us-ascii I don't see a way to download the file. Under options I only see upload Download is grayed out Thanks Randall Sent from my iPhone On Jul 30, 2012, at 10:48 AM, Mann Chaddha mann.chad...@gmail.com wrote: Hi The best the fastest way I have bee using is with the CUC Call Handlers. Then I grant greeting admin privileges change customer keypad mappings for a user to record the same. Do you see any error whilst looking at the greetings on the CH. hth Mann On Mon, Jul 30, 2012 at 9:30 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Prompt recording (Ken Wyan) -- Message: 1 Date: Mon, 30 Jul 2012 17:42:31 +0530 From: Ken Wyan kew...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Prompt recording Message-ID: capbg9bku7prygp3mc52bzc9kds70xp-pv1fslosmyewbdmn...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 If
[OSL | CCIE_Voice] Prompt recording
Randall, Have you tried just logging into Unity Connection from UCCX and recording your prompt there? Go to Unity Connection-- Users. Pick a user and go to greetings and select standard greeting. Click on Play/Record--click on options--Playback and Recording--Play Back Device select phone--Recording Device--Phone--Active Phone Number-put in a phone number from your pod--Performance--Play message while downloading. Then click on record and it will ring the phone you selected answer and record your prompt. When finished recording click on options--save file as--you can save file.wav where you like and upload your prompt to UCCX. I hope I haven't over simplified, but this is how I record my prompts for UCCX scripts. If you can't reach Unity from UCCX for some reason there is a single line script that records prompts for you. Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, July 30, 2012 1:06 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 77, Issue 55 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: ANI/DSNI TON on Proctor Labs (Rrcrumm) 2. Re: Prompt recording (Rrcrumm) 3. Re: Prompt recording (Jason Aarons (AM)) -- Message: 1 Date: Mon, 30 Jul 2012 11:38:14 -0700 From: Rrcrumm rrcr...@yahoo.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs Message-ID: f3878a37-0374-45b9-9230-619eec35d...@yahoo.com Content-Type: text/plain; charset=us-ascii Hi It is with MGCP gw Sent from my iPhone On Jul 30, 2012, at 9:55 AM, Krishna vinayak_...@yahoo.com wrote: if it is h323 gateway, i will create a translation rule and apply at the dial-peer... if it is mgcp gateway do it on the call manager route list detail level... thank you krishna. From: Randall Crumm rrcr...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Sent: Monday, July 30, 2012 11:33 AM Subject: [OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs Hello, I have noticed some different behaviors and was wondering what you recommend If the question asks for plan unknown and type unknown should you set to 1. plan unknown and type unknown or 2. plan isdn type unknown or 3. plan call manager type call manager For me I have tried the above and it seems like call manager/call manager is what is working(actually allowing the call to go through). It goes through as unknown/unknown Any thoughts? Thanks! Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120730/d05c6c96/attachment-0001.html -- Message: 2 Date: Mon, 30 Jul 2012 12:00:40 -0700 From: Rrcrumm rrcr...@yahoo.com To: Mann Chaddha mann.chad...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Prompt recording Message-ID: 3f8330ca-e532-4f40-87de-17a1a5ff4...@yahoo.com Content-Type: text/plain; charset=us-ascii I don't see a way to download the file. Under options I only see upload Download is grayed out Thanks Randall Sent from my iPhone On Jul 30, 2012, at 10:48 AM, Mann Chaddha mann.chad...@gmail.com wrote: Hi The best the fastest way I have bee using is with the CUC Call Handlers. Then I grant greeting admin privileges change customer keypad mappings for a user to record the same. Do you see any error whilst looking at the greetings on the CH. hth Mann On Mon, Jul 30, 2012 at 9:30 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject
[OSL | CCIE_Voice] [PLUS DIALING CONSIDERATION-MODIFYING NUMBER DISPLAYED ON PHONE]
I'm placing a call to 95551212. I need the phone to display to --5551212. I'm using predot and prefixing a 9 on the route list and sending 10 digit calling number. I'm trying to figure out how to strip the 9 on the display of the phone without modifying the h323 dial-peers on the gateway. I've tried called party transformations on the gateway and the device pool and it changes what is displayed on phone, but only sends the transformed number 5551212 to the h323 gateway and the call doesn't complete since it doesn't send the 9. Any suggestions for help appreciated. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [PLUS DIALING CONSIDERATION-MODIFYING NUMBER DISPLAYED ON PHONE] RESOLVED
Thanks everybody for replying so fast. The answer is to apply: voice service voip no supplementary-service h225-notify cid-update It works great and is much better than adding dial peers. Thanks especially to Kevin and Jason you were right on the mark. Michael Sears From: kevinspice...@gmail.com [mailto:kevinspice...@gmail.com] On Behalf Of Kevin Spicer Sent: Friday, July 13, 2012 10:54 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [PLUS DIALING CONSIDERATION-MODIFYING NUMBER DISPLAYED ON PHONE] Hi Michael, Voice service voip No supplementary-service h225-notify cid-update Then the phone should display the number as manipulated y the RP. On 13 Jul 2012 17:19, michael.se...@compucom.commailto:michael.se...@compucom.com wrote: I'm placing a call to 95551212. I need the phone to display to --5551212. I'm using predot and prefixing a 9 on the route list and sending 10 digit calling number. I'm trying to figure out how to strip the 9 on the display of the phone without modifying the h323 dial-peers on the gateway. I've tried called party transformations on the gateway and the device pool and it changes what is displayed on phone, but only sends the transformed number 5551212 to the h323 gateway and the call doesn't complete since it doesn't send the 9. Any suggestions for help appreciated. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUCM NTP SERVER USING ATOMIC CLOCK for ACCURATE TIME.
I had a client with the requirement to point to an accurate time server like in your explanation. I used the Boulder Atomic Clock the most accurate clock in the world: http://voices.yahoo.com/boulder-home-atomic-clock-most-accurate-clock-6136932.html NTP Servers: ntp server 132.163.4.101 ntp server 132.163.4.102 ntp server 132.163.4.103 To configure CUCM to utilize this time source: Cisco Unified Operating System AdministrationSettingsNTP ServersAdd New As long as CUCM has access to the Internet these servers will meet your requirement. They worked very well for my client. You just need to make sure your Date/Time groups are accurate. Hope this helps Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [How to test Privacy]
I have two phones setup with a shared line, 3012. The phones also have privacy configured on them on button 6. How do I test to see that Privacy is working or not? As you can tell I'm new to Privacy and could use some information as my lab is 3 weeks away and really need some information. Or if you have a link you could send I would greatly appreciate it. Thanks Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Mobile Voice Access and H323 Gateway
An H323 gateway is required to enable Mobile Voice Access service. The H323 gateway is where you setup the service mva. Configure on H3233 Gateway: Application Service mva http://Unified CM cluster Publisher IP Addr:8080/ccmivr/pages/IVRMainpage.vxml In CUCM go to help. Then search for mobile voice and look for Configuring an H.323 Gateway for System Remote Access by Using PRI. Michael Sears -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Saturday, June 30, 2012 10:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 76, Issue 68 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Mobile Voice Access (Felton (Feng) Xu) -- Message: 1 Date: Sat, 30 Jun 2012 05:06:09 -0700 (PDT) From: Felton \(Feng\) Xu felto...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Mobile Voice Access Message-ID: 1341057969.38483.yahoomail...@web113809.mail.gq1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Hi, I am doing the mobile voice access lab, I found most of the configuration guide will include the H.323 gateway to play the IVR according to the VXML file. For my current lab, I only have the CUCM over VMware, and an Asterisk server acting as PSTN/PLMN via SIP trunk to CUCM. Is that possible to have mobile voice access working in my environment, is H.323 GW is a must for this scenario? Scenarios: 1. mobile number (#6140918 from Asterisk) called to MVA number (#6999) 2. CUCM replied with 404 Not Found. 3. The CUCM log file indicate the following: 06/30/2012 21:41:24.504 CCM|DbMobility: getMatchedRemDest starts: cnumber = +6140918|CLID::StandAloneClusterNID::192.168.1.101LVL::DetailedMASK::ff 06/30/2012 21:41:24.504 CCM|DbMobility: getMatchedRemDest: partial match|CLID::StandAloneClusterNID::192.168.1.101LVL::DetailedMASK::ff 06/30/2012 21:41:24.504 CCM|DbMobility: removeLeadingPlus: Calling Party Number starts with a leading +, +6140918 , removing it ... |CLID::StandAloneClusterNID::192.168.1.101LVL::DetailedMASK::ff 06/30/2012 21:41:24.504 CCM|DbMobility: getMatchedRemDest: number to partial match is 6140918|CLID::StandAloneClusterNID::192.168.1.101LVL::DetailedMASK::ff 06/30/2012 21:41:24.504 CCM|DbMobility: getMatchedRemDest: need to match 8 digits|CLID::StandAloneClusterNID::192.168.1.101LVL::DetailedMASK::ff 06/30/2012 21:41:24.504 CCM|DbMobility: getMatchedRemDest can't find remdest +6140918 in map|CLID::StandAloneClusterNID::192.168.1.101LVL::DetailedMASK::ff Regards,? --- Feng Xu (Felton) felto...@yahoo.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120630/74c2defa/attachment-0001.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 76, Issue 68 ** ___ For more information regarding industry leading 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 test Privacy]
Ah man thank you. That's too easy was making it too complicated. Following your instructions I see exactly how it works. Privacy just blocks the other phone with the shared line from seeing the information. Thank you Michael Sears Compucom Systems Western Region Senior Consultant Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax:+1.978.863.0740 [cid:image001.jpg@01CD5794.7A0F6CE0] From: kevinspice...@gmail.com [mailto:kevinspice...@gmail.com] On Behalf Of Kevin Spicer Sent: Sunday, July 01, 2012 2:13 PM To: Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [How to test Privacy] [Please keep the list in cc] I'm a little unclear from your description how you are testing this. So lets say you have 2 x HQ phones with a shared line (2010, say) and a privacy button on button 6. You press the privacy button on HQ phone 1 so that the black dot appears, then place a call from the PSTN to 2010. Both phones see th inbound call and its details. Then answer it on HQ phone 1. HQ Phone 1 has a call in progress, but HQ phone 2 no longer shows any information about this. Then press the privacy button again, so that the dot becomes unshaded, you should see details of the call in progress appear on HQ phone 2. So the privacy button prevents the sharing of information on in progress calls. On Sun, Jul 1, 2012 at 9:02 PM, michael.se...@compucom.commailto:michael.se...@compucom.com wrote: Ok. You see what I'm trying to do. I have privacy setup on HQ Phones with shared line. When I press the privacy button and make a call to the shared line it shows all the caller ID information. I'm trying to get it so it won't display the Caller ID information or am I testing wrong. Thanks for your information obviously I'm confused how this works. The question says Make sure when this button is pressed, the other phone cannot see the calling number of the shared line. When I call from the shared line to another phone I still see the phone number. Any other information you can add so I can test and understand this. I believe I have it configured correctly just don't know how to test it. Thanks again, --ms Michael Sears Compucom Systems Western Region Senior Consultant Office: +1.720.344.6833tel:%2B1.720.344.6833 Mobile: +1.303.328.5590tel:%2B1.303.328.5590 Fax:+1.978.863.0740tel:%2B1.978.863.0740 [cid:image001.jpg@01CD5794.7A0F6CE0] From: kevinspice...@gmail.commailto:kevinspice...@gmail.com [mailto:kevinspice...@gmail.commailto:kevinspice...@gmail.com] On Behalf Of Kevin Spicer Sent: Sunday, July 01, 2012 1:53 PM To: Juan Carlos Anzola Cc: Sears, Michael (msears); ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [How to test Privacy] Even without barge configured you should see that, when privacy is on, the shared lines do not see information (callerID etc.) for calls in progress. On Sun, Jul 1, 2012 at 8:44 PM, Juan Carlos Anzola juancarlosanz...@gmail.commailto:juancarlosanz...@gmail.com wrote: Hi Michael, Privacy feature ussually works in conjuction with Barge feature. To test privacy, you can make a call to the shared line, answer it in either phone. * If privacy is enabled you won't be able to barge from the other phone. * If privacy is disabled you will be able to barge in. By default, even when privacy is enabled, as soon as you put the call on hold (from the phone with privacy enabled) the other party will be able to take that call. If you want to override this behaviour, you can go to Service Parameter and search for Enforce privacy on held calls or something like that. HTH -- Juan Carlos Anzola ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com inline: image001.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [MVA - CALLER PARTY NAME]
Greetings, I'm working on a MVA lab that's asking for the 4 digit calling number and the name. MVA is working perfectly, but nothing I do seems to enable the calling party name. Has anyone been able to accomplish sending the calling name when making an MVA call? If so please share how you did it. If I make the MVA call it only presents the four digits which is ok, but it won't show the calling name until the call is answered. I've read many articles, some of which provide solutions that don't work and some that say it's not supported so I'm looking for the definitive answer on whether this can be done or not. Any feedback appreciated. Thank you, --ms ___ For more information regarding industry leading 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] Cannot Conference on CME
Greetings Krishna, What does you telephony service configuration look like. Below is a working example that I have built in my lab with CME registered to gatekeeper. 10.1.10.1 is the loopback address sccp local Loopback0 sccp ccm 10.1.10.1 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 2 register conference associate profile 1 register transcoder ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 6 conference-join custom-cptone join conference-leave custom-cptone leave associate application SCCP telephony-service sdspfarm units 6 sdspfarm transcode sessions 3 sdspfarm conference sessions 3 sdspfarm tag 1 transcoder sdspfarm tag 2 conference no privacy conference hardware no auto-reg-ephone xml user pvadmin password cisco 15 max-ephones 10 max-dn 20 no-reg ip source-address 10.1.10.1 port 2000 strict-match url authentication http:// 10.1.10.1 /CCMCIP/authenticate.asp pvphone cisco time-zone 42 time-format 24 voicemail 4220 max-conferences 8 gain -6 web admin system name admin password cisco transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Jun 4 2012 02:36:34 Hope this helps, Best Regards --ms -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, June 04, 2012 3:24 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 76, Issue 7 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: cannot conference on CME (Krishna) 2. Re: cannot conference on CME (Jeff Mchugh) 3. Re: cannot conference on CME (chase mergenthal) -- Message: 1 Date: Mon, 4 Jun 2012 12:02:20 -0700 (PDT) From: Krishna vinayak_...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cannot conference on CME Message-ID: 1338836540.25074.yahoomail...@web46008.mail.sp1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Hi folks, i configured the transcoder and conference resources on the CME, but couldn't make them to work. When i want to conference the line, it says cannot complete the conference. here is my config: Did i miss any configuration part in this below config sccp local FastEthernet0/0 sccp ccm 10.50.5.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 ?bind interface FastEthernet0/0 ?associate ccm 1 priority 1 ?associate profile 1 register Conference ?associate profile 2 register mtp(mac-address of sourceinterface) ?keepalive retries 5 ?switchover method immediate ?switchback method immediate ?switchback interval 5 ! dspfarm profile 2 transcode ?codec g711ulaw ?codec g711alaw ?codec g729ar8 ?codec g729abr8 ?codec g729r8 ?codec g729br8 ?maximum sessions 24 ?associate application SCCP ! dspfarm profile 1 conference ?codec g711ulaw ?codec g729r8 ?codec g729br8 ?codec g729abr8 ?codec g729ar8 ?maximum sessions 4 ?associate application SCCP here is the status of sccp: SCCP Admin State: UP Gateway Local Interface: FastEthernet0/0 ? ? ? ? IPv4 Address: 10.50.5.1 ? ? ? ? Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.50.5.1, Port Number: 2000 ? ? ? ? ? ? ? ? Priority: N/A, Version: 7.0, Identifier: 1 ? ? ? ? ? ? ? ? Trustpoint: N/A Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.50.5.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 48, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 Conferencing Oper State: ACTIVE - Cause Code: NONE
[OSL | CCIE_Voice] CUE/CUCM INTEGRATION WIZARD WON'T RUN-CTI ROUTE POINT WON'T REGISTER
Greetings and salutations group --Having an issue with and integration of CUE with CUCM. Problem is CUE won't register with the CTI Route Point. I've tried running the wizard to do this and the browser locks up and ping times to the CUCM go to 5000+ms. They are usually 1-4ms from CUE to CUCM--when run the Wizard they spike to the greater value. I've also configured CUE manually and the CTI Route Point still will not register. I've tried this on an AIM module as well as NM module. I'm running version c2811nm-adventerprisek9_ivs_li-mz.124-24.T6.bin IOS and CUE version 7.0.6. I've checked the routing without the wizard trying to reach CUCM and it's only two hops with great ping times. I've tried everything I know to alleviate the problems and can't get to the root cause here. Any suggestions would be appreciated. cue# show ccn status ccm-manager JTAPI Subsystem is not registered with any Call Manager cue# show ccn subsystem jtapi Cisco Call Manager: 10.100.1.11,10.100.1.12 CCM JTAPI Username: cuejtapi CCM JTAPI Password: * Call Control Group 1 CTI ports: 3221,3222,3223 Call Control Group 1 MWI port: CSS for redirects from route points:ccm-default CSS for redirects from CTI ports: redirecting-party Again, any input would be appreciated and thanks. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Reply--Phoneview Issue
There's a great video on the Unified FX site: http://www.unifiedfx.com/Videos Goes through Exactly how to set it up correctly and if your using version 2.0 you have to have .net framework 4.0. ___ For more information regarding industry leading 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_Voice Digest, Vol 65, Issue 114
Thanks much for the detail have to be able to find it when sitting the Lab in 30 days. Michael Sears From: Mohamed Gazzaz [mailto:mgaz...@hotmail.com] Sent: Friday, April 27, 2012 11:15 PM To: Sears, Michael (msears); ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 65, Issue 114 Hello Michael, From this url, http://www.cisco.com/cisco/web/psa/default.html?mode=home Select Products -- Voice and unified communications -- Customer Collaboration -- Cisco Unified Contact Center Products -- Cisco Unified Contact Center Express -- Configuration Examples and TechNotes -- Configure a One Button Login for IP Phone Agents Regards, Mohamed Gazzaz From: michael.se...@compucom.commailto:michael.se...@compucom.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Date: Fri, 27 Apr 2012 20:10:19 -0400 Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 65, Issue 114 Greetings all, I'm trying the find the IPPA URL in CUCM help or the ICON that will be on the candidate desktop without any luck. Can anyone provide assistance on how to find this URL when sitting in the lab. Thank you, Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://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_Voice Digest, Vol 65, Issue 114
Greetings all, I'm trying the find the IPPA URL in CUCM help or the ICON that will be on the candidate desktop without any luck. Can anyone provide assistance on how to find this URL when sitting in the lab. Thank you, Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [Hyderabad India Dialplan]
I've had a new site come up in Hyderabad India. Wondering if someone could share information regarding the dial-plan used there. Any information would be appreciated. Thank you, --ms Michael Sears ___ For more information regarding industry leading 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] Istalling CUE license file
I had the same issue. Are you using Filezilla? That's what I used and I watched the logs as the download took place or failed to take place. Initially I also had some problems with setting up Filezilla, but once I got it setup correctly everything worked as you are doing it. If you can ping the ftp server --problem is most likely the configuration of Filezilla or the location in which you have placed the license file. In the IPX walk through tutorials on CUE Vic does a great job of explaining how to do it. Hope this helps. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [CERTIFICATE ERROR WHEN CONNECTING TO SERVERS]
When I'm at home and VPN into my LAB eveything works fine I can RDP to AD and UCCX and HTTPS into CUCM, UC and CUPS. But when I VPN into my network from outside my network I can connect to everything Routers, Switches, RDP but have problems connecting to any of my Cisco Servers Using HTTPS. I come up to the There is a problem with this website's security certificate as usual and click on Continue to this website (not recommended) as usual but can't connect and get Certificate Error. I've tried adding exceptions as trusted sites and about eveything else. I've tried several different browsers but no luck. Any suggesions would be appreciated. Thank you --ms ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] E1 configuration (Eliot Ngwa)
I read in the thread below I am using simple crossover cable (Ethernet crossover). This cable will not work. You need a T1 cross over cable: http://www.google.com/search?q=t1+crossover+cablehl=enprmd=imvnstbm=ischtbo=usource=univsa=Xei=3gP-Tp2AIpS5twf-ufXPBgsqi=2ved=0CF4QsAQbiw=1072bih=804 !!! If your using an Ethernet cross over it won't work need T1 cross over 1--4, 2--5, 4--1, 5--2 http://www.ebay.com/itm/T1-Crossover-cable-3FT-/160570999135?pt=LH_DefaultDomain_0hash=item2562c7015f Usually you can pick one up at local computer store or ebay real cheap depending on the length. Hope this helps. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] PSTN-WAN Enabled Secret
All is not lost: There are only a few methods you can use to recover from a lost password, but all of them destroy the start-up configuration. *Routers that have nonvolatile RAM (NVRAM) chips can be removed and reseated. The NVRAM is implemented with battery-backed up static RAM (SRAM). If you remove the SRAM, the contents of NVRAM are erased as well as the no service password-recovery configuration. Be sure to use proper anti-static procedures when you handle the NVRAM. Some of these routers are 3640 and 3660. *Other routers, such as the 1700, 2600, and 3620, use an electrically erasable programmable read-only memory (EEPROM) in order to hold the configuration. The EEPROM does not erase when you remove it. *Another method is to reload or boot the router with console access, and press CTRL-BREAKhttp://www.cisco.com/en/US/products/hw/routers/ps133/products_tech_note09186a0080174a34.shtml within five to ten seconds of the Cisco IOS software image decompressing or roughly when the Image text-base:... part of the banner begins. You are then prompted to reset the router to factory default (erase start-up configuration). Hope this helps --ms Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MGCP Gateway has CSS set to ALL?
Sure enough - it is a CSS named all containing a ton of partitions. Thanks Randall - should have figured this out. This doesn't seem to be a very efficient way of doing things. Can someone explain the behavior when you assign a CSS to a Gateway with multiple partitions spanning multiple sites? Does it search through the CSS until an appropriate match is found for the phones partition? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CSS'S, PT and COS Question
Greetings, I'm building a site and want to restrict Extension Mobility (non-logged in phones) to calling internally and emergency services only. On the device level I've assigned the International CSS which has all partitions. On the line level I have assigned the Services CSS which contains partitions or only emergency numbers and internal phones. Will this restrict the calls to internal and emergency only? I'm thinking that the line will be used? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MGCP Gateway has CSS set to ALL?
Greetings all - I'm researching a cluster located in Dublin and it has an MGCP Gateway support the Paris Site. I'm reviewing the gateway under the E1 PRI and the CSS is set to ALL. Anyone know what this means?? Does it search through all the CSS's in the list for a match?? Never seen this before. Any input appreciated. --ms ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany]
Greetings - I am seeking input on developing a dial plan for a site that has been thrown my way in Munich Germany. I'm new to ISDN ERA and have been using NANP for years. Any input regarding developing a dial plan for Munich Germany, including sample configurations of CUCM and Gateways, would be greatly appreciated or if you can point me to resources ( in the right direction) would be immensely appreciated. Thank you. ___ For more information regarding industry leading 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] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany]
Thanks Ash - Yeah I'll speak to BT thought this was going to be a tough one. -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: Thursday, December 01, 2011 11:39 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany] Germany Dial-plan is the most complicated dial-plan in the world because its not organized as the NANP or UK Dial plans , they don't have a fixed pattern in the access codes or length , you have to speak to someone from Germany who know about the telecom there or maybe the German telecom itself can give you guide about that , Ash On Thu, Dec 1, 2011 at 7:53 AM, michael.se...@compucom.com wrote: Greetings - I am seeking input on developing a dial plan for a site that has been thrown my way in Munich Germany. I'm new to ISDN ERA and have been using NANP for years. Any input regarding developing a dial plan for Munich Germany, including sample configurations of CUCM and Gateways, would be greatly appreciated or if you can point me to resources ( in the right direction) would be immensely appreciated. Thank you. ___ For more information regarding industry leading 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 Problem requested channel unavailable
ISDN Problem requested channel unavailable: 1. Which side disconnects the call with this reason in which direction? 2. You might be requesting bchannels in ascending while the telephony switch is hoping to allocate b-channel in decending manner, hence the swicth will reject the call with requested channel unavailable. a. Try using the command isdn bchan-number-order ascending / decending to adjust. 3. The companding type might not be properly matched. a. Are you set for A-law or U-law. Verify configurations. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
