[OSL | CCIE_Voice] Outbound PSTN call drops
You may want to look into Bug ID: CSCsx67255...link below. http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetailsbugId=CSCsx67255from=summary *Inder Singh* CCIE™ Voice #38235 Sr. Network Engineer, Unified Communications ___ For more information regarding industry leading 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] need help guys..
Amit. On proctor labs, even though there is configuration on the UCCX server, this config is not propagated to the CUCM. Make sure you initiate a configuration sync on the UCCX to propagate to CUCM. I am having proctor labs...' when i am working for cucm and uccx integration task.. it is already done by default for us in lab... but when check in cucm not able to see rmcm user that used in uccx ... and when i try to add ipcc extension in cucm end user...it is not having any option to add it... is it any config issue or i missed something that need to do for fix it? -- 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
[OSL | CCIE_Voice] Best practice for binding interfaces to sccp and CUE
I prefer to use a loopback interface for media resources and CUE. The proctor may ask you to bind the CUE to one interface and the media resources to another...so please read the question carefully. If you do decide to bind the CUE to the loopback your mask on the interface will have to be something other than a 32 bit mask so use the command ip ospf network point-to-point in order to propagate the network over ospf. -- Message: 1 Date: Wed, 26 Jun 2013 20:32:55 +0530 From: singh singh8...@in.com To: ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Best practice for binding interfaces to sccp andCUE Message-ID: 1372258975.57342f6b95854ad89e9c4088ab94a...@mail.in.com Content-Type: text/plain; charset=utf-8 Hello All,Wondering what is the best practice to bind sccp interface and CUE to .For example if I have an interface on the router which is my voice interface and a loopback then which as per the best practice can be used for binding singh ___ For more information regarding industry leading 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] SIP Trunk with CUC and possible AAR
Hi All, A big thanks to Ash, Amit and Raees for replying...you guys gave me a lot of good information to think about. Before posting this issue I was about 90% of the way there...please see the scenario in my original email below. As you all probably know AAR is not supported with SIP trunks. Therefore in order to provide HA in the situation where you have WAN congestion, a route list with sip trunk as the first choice and PSTN gateway as the second choice must be used. However the problem occurs when trying to redirect to voicemail on NoAnswer or Busy. In this case if the CUC is to recognize the redirecting number you either have to forward four digits out the PSTN or at the HQ site you need to create a translation pattern and use the calling party mask to cut the number down to four digits. I hope this helps others in their journey. Cheers!! Inder. To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR Message-ID: CA+53e6vAJciztxfr=zerprzmboidgafrtaqr-dykezv6z5e...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. It then asks for calls that are sent to VM from BR1 to be redirected out the PSTN when there is WAN congestion. I have looked high and low but I can't find any reference where this can be done with AAR...or am I totally missing something. If AAR is possible can someone point me in the right direction? If it is not possible can someone let me know how you might achieve this otherwise? I tried using a route list with the SIP trunk as the primary RG and the PSTN GW and the secondary RG. The issue is redirecting the caller, called and redirect on no answer. We need the BR1 phone to be able to press the message key and retrieve messages (this I was able to do with alternate extensions) and also for callers to be redirected to voicemail for the called party (this I was not able to do with the route list scenario). Thanks in advance for any help you can provide. Regards. Inder. ___ For more information regarding industry leading 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] HOWTO?: Unified CM User Options /ccmuser
Hi Anthony, I don't think it is possible to get to the UCM user page from the CUC server...it must be accessed from the UCM server. Regards. Inder. Subject: [OSL | CCIE_Voice] HOWTO?: Unified CM User Options /ccmuser page from Unity Connection not working Message-ID: CADkWibdTMPUzVfKAay9G-re=yt0oug7uac7oka5pmhehko8...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 How do you allow a user to get to the ccmuser Cisco Unified CM User Options page from Unity Connection (*not* CUCM)? https://10.10.210.13:8443/ccmuser/showHome.do The username/password is accepted but I just get bounced back to the login page. All Feature Services/Network Services are running. Voicemail and the PCA page https://10.10.210.13:8443/ciscopca/home.do are both working From CUCM, the CM User Options page works; https://10.10.210.10:8443/ccmuser/showHome.do I just can't seem to enter from Unity Connection. Anthony ___ For more information regarding industry leading 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] need help with ras issue.
Hi Cecil, A couple of things: 1. If you are not using a defualt gw tech prefix please make sure that you are tagging the ras dialpeer with a tech-prefix and stripping that prefix at the CCM Also: 2. post the gatekeeper config as well as sh gatek end and sh gatek gw 3. post the gateway config including the interface used to register with the GK and the ras dialpeer itself. also include the sh gateway output from the gateway. Regards. Inder. Subject: Re: [OSL | CCIE_Voice] need help with ras issue. Message-ID: caogsoawz0vgp7dopecse0z34-b87wbkp5ypwfcxm9hjaj_u...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, On Gatekeeper, post the output of debug gatekeeper main 10. Regards, Mohd Baqari On Tue, Oct 18, 2011 at 10:04 PM, Cecil Wilson cecil...@gmail.com wrote: Hello I am having problem with ras config from BR2 to HQ. the HQ phone will not ring. I can from BR2 to HQ over pots dial-peer but I can not call HQ from BR2 using ras. I can get from HQ to BR2 ok. can someone help? thanks Cecil ___ For more information regarding industry leading 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] Conference Resource
Hi, If BR1 is hosting the conference then it will be BR1's HW-conf bridge resource which will be invoked. Regards. Inder. Subject: [OSL | CCIE_Voice] Conference Resource Message-ID: cacsoyaoddo95atrqu4umjtiyznztb68xtdaatrxkdpztyu_...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Lets say HQ Phone1 is attending a conference @ BR1, and BR1 Phone 1 is the host of the conference, HQ Phone1 has HQDP--HQMRGL--HQMRG--HQXcoder and HW-Conf , BR1 Phone1 has BR1DP--BR1MRGL--BR1RG--Br1Xcoder and BR1HW-Conf So what hardware resources HQ Phone1 will use when attending a conference hosted by BR1 Phone 1, as per region settings HQ --BR1 Codec is g729. ___ For more information regarding industry leading 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] SIP Trunk with CUC and possible AAR
Hello All, I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. It then asks for calls that are sent to VM from BR1 to be redirected out the PSTN when there is WAN congestion. I have looked high and low but I can't find any reference where this can be done with AAR...or am I totally missing something. If AAR is possible can someone point me in the right direction? If it is not possible can someone let me know how you might achieve this otherwise? I tried using a route list with the SIP trunk as the primary RG and the PSTN GW and the secondary RG. The issue is redirecting the caller, called and redirect on no answer. We need the BR1 phone to be able to press the message key and retrieve messages (this I was able to do with alternate extensions) and also for callers to be redirected to voicemail for the called party (this I was not able to do with the route list scenario). Thanks in advance for any help you can provide. Regards. Inder. ___ For more information regarding industry leading 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] Dear All my Multicast MOH is not working
Hello Darshan, Since you have set the max hops to 1 and are trying to invoke MOH on the BR site I will assume you are trying to spoof multicast MOH. 1. In this case make sure that on the branch router you have the following: ccm-manager music-on-hold 2. and either in the call-manager-fallback or the telephony-service construct assure you have the following: moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route *local loopback address ip address of local voice subnet* 3. On the BR router flash make sure you have the music on hold file. 4. On the UCM at HQ make sure that you create a region that is g711 everywhere and that you create a device pool for this region and apply it to your multicast music on hold server. Regards. Inder. Message: 1 Date: Sat, 17 Sep 2011 00:33:30 +0300 From: darshan ccievoice0...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Dear All my Multicast MOH is not working Message-ID: bay154-ds144ec9686601cc347d2a52b1...@phx.gbl Content-Type: text/plain; charset=us-ascii Multicast Moh config. 1. Media Resource file 1..select Multicast 2. MOH server PUB and selecting MOH region and select MOH with IP 239.1.1.1 and max hobs 1 3. MRG ..in MRG I have selected all resource except MOH-2 . 4. MRGL 5. Selected MRGL in siteB Device pool 6. In Global Config mode in SIteB I have given ccm-manager music-on-hold. Regards Darsh ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME Extension Mobility Issue
Hello All, I am working on Lab 9A in volume 1 and have setup extension mobility for CME. Here is the configuration: voice logout-profile 1 pin 1234 number 3002 type normal ! voice user-profile 1 max-idle-time 10 user br2ph3 password adgjm number 3102 type normal speed-dial 1 3006 ! em logout 7:0 19:0 23:0 url authentication http://10.10.202.1/CCMCIP/authenticate.asp cisco cisco ! ephone 2 description YOUR 79XX PHONE or IP BLUE mac-address EC44.761F.E658 max-calls-per-button 5 busy-trigger-per-button 3 type 7965 logout-profile 1 The problem I am having...when I log into extension mobility I see the speed dial number on the screen but do not see the extension 3102. Has anyone run into this? Thanks. Inder. Inder Singh (403) 992-6956 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME Extension Mobility Issue
Hello All, I just wanted to update you on the fix. After a couple people replied it was all too simple ;-). I had missed creating an ephone-dn for the number is the user profile. Thanks to all that replied. Regards. Inder Singh (403) 992-6956 On Sat, Sep 10, 2011 at 9:55 AM, Inder Singh ising...@gmail.com wrote: Hello All, I am working on Lab 9A in volume 1 and have setup extension mobility for CME. Here is the configuration: voice logout-profile 1 pin 1234 number 3002 type normal ! voice user-profile 1 max-idle-time 10 user br2ph3 password adgjm number 3102 type normal speed-dial 1 3006 ! em logout 7:0 19:0 23:0 url authentication http://10.10.202.1/CCMCIP/authenticate.asp cisco cisco ! ephone 2 description YOUR 79XX PHONE or IP BLUE mac-address EC44.761F.E658 max-calls-per-button 5 busy-trigger-per-button 3 type 7965 logout-profile 1 The problem I am having...when I log into extension mobility I see the speed dial number on the screen but do not see the extension 3102. Has anyone run into this? Thanks. Inder. Inder Singh (403) 992-6956 ___ For more information regarding industry leading 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 Presence Client Install File
Hi All, I am on volume 1 lab 13 and am being asked to install the CUPC client. Does anyone know where the install files are hidden? Thanks. Inder. Inder Singh (403) 992-6956 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
