Re: [OSL | CCIE_Voice] CUE licensing question
Did you check the install guide? El mar, 24-05-2011 a las 19:26 -0700, Cristobal Priego escribió: hello all i was just curious if you need to change the cue license on the lab, if you don't have an ftp server to do so what are your options to get the proper license on the CUE ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Home Lab Guide
What routers do you have? Any VWIC serial cards? DSPs? NM-4ESW? El lun, 25-04-2011 a las 22:04 -0500, Bill Lake escribió: do you want a full lab or a partial one to connect to rack rental? On Mon, Apr 25, 2011 at 9:27 PM, Abel ... midga...@gmail.com wrote: Hello everyone, This is my first post here, can someone help me with a guide to build a home lab. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading 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] pstn call error
Hey, if you have issued the command ccm-manager config and ccm-manager config server, there is a chance the gateway will continue to download the config file from CCM server and overwrite the gateway configuration. When this happens, CCM will instruct the gateway to use 24 channels though you have configure less channel on the t1 controller. The correct is to not configure ccm-manager config if you want to control all the channel and isdn settings from the gateway and not from CallManager. Cheers, El vie, 22-04-2011 a las 22:30 -0600, donny f escribió: i config to mgcp and work fine. what i might miss here? On Fri, Apr 22, 2011 at 7:41 PM, Hough, Earl earl.ho...@pcmallservices.com wrote: The reason for this error message is not due to H323. Look at the source of the messages. They are Q931 messages. What might cause the required circuit/channel not to be available on a PRI? Hint: look at your base ISDN configuration and compare that to the PSTN emulator for the same circuit. Earl Hough CCIE #16508 (RS/Security/Voice) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of donny f Sent: Friday, April 22, 2011 8:48 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] pstn call error hi all, what i miss in h323 config, that cause this ? debug from HQ (h 323) -- Cause i = 0x80AC - Requested circuit/channel not available *Apr 23 05:41:45.559: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8087 --- deug in PSTN --- Cause i = 0x80AC - Requested circuit/channel not available Apr 23 04:42:28.023: ISDN Se0/3/0:23 Q931: TX - RELEASE pd = 8 callref = 0x8087 Apr 23 04:42:28.031: ISDN Se0/3/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x0087 PSTN-WAN(config-controller)# Apr 23 04:42:56.107: %ENVMON-3-FAN_FAILED: Fan 1 not rotating _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading 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] Meet Me + Ad-hoc Participant (WB2 Lab1 6.2)
Hi Alex, I would check if the setting Advanced ad-hoc conferencing is enabled on CCM Service Parameters. This would allow you to chain conferences between each other. HTH, Randall S. El sáb, 23-04-2011 a las 19:12 +0800, Alex Goh escribió: Regards, Alex ___ For more information regarding industry leading 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] workbook 2 lab 8
Guys, If this was a problem with the SIP IP parameters, then he wouldn't be receiving any prompts at all. The fact that he receives the standard greeting is because CUC is not receiving the caller or redirected information on the SIP messages. As suggested, make sure you have the redirected header delivery setting. It may also matter if you are calling a forwarded extension or if you just press on the Messages button. Cheers, El sáb, 23-04-2011 a las 09:56 +0100, George Goglidze escribió: On CUE you most probably have a mismatch in what you've bind sip control ip to in voice service VoIP, and on cue sip server by default is the default gateway. Sent from my iPad On 23 Apr 2011, at 04:56, Cristobal Priego cristobalpri...@gmail.com wrote: Hello all I just finished my session and i was doing lab 8 from workbooks 2 something that i couldn't get to work properly was the unity integration through sip trunk the ring no answer and the busy were playing enter your id followed by pound instead of sorry extension ... is not available record your message at the tone i followed the proctor-guide and still it' didn't work also i couldn't get the MWI to work on CUE sip phones only my config looked like this voice register dn 1 number 3002 call-forward b2bua busy 3600 call-forward b2bua noan 3600 timeout 12 mwi name br2 phone 4 sip-ua mwi-server ipv4:10.10.202.2 i had the unsolicited notify enabled on the cue gui when i was doing a refresh of the mwi i saw unity express trying to ring my extensions on the default mwi extensions so i went ahead and configured the ephone dn's for mwi still didn't work also my sip srst didn't work i kept getting this error Apr 23 07:46:38.660: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bKe43aa9be From: sip:1002@10.10.201.1;tag=001ae22b12c500077cb0194d-395f52ee To: sip:1002@10.10.201.1;tag=17DB834-A33 Date: Sat, 23 Apr 2011 07:46:38 GMT Call-ID: 001ae22b-12c50006-85a48968-9fe91895@192.168.11.12 Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 REGISTER Content-Length: 0 my sip srst looked like this voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 cor incoming ld-css default call-forward b2bua busy 5600 call-foward b2bua noan 5600 timeout 13 codec g711u voice register global max-pool 2 max-dn 2 please help me out, 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 ___ For more information regarding industry leading 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] Slightly OT: Home Lab L2 Tunnel Question
Well, won't you share it?? :P I'm also very curious on how its done as I have asked myself the same question. Cheers. El jue, 31-03-2011 a las 23:34 -0400, Wayne Lawson escribió: Jeff - I know how it's done! ;-) Regards, Wayne A. Lawson II - CCIE #5244 (RS) Founder, President CEO - IPexpert, Inc., Proctor Labs, Inc. Platinum Solutions Group, LLC. Mailto: wlaw...@ipexpert.com Telephone: +1.810.334.1564 eFax: +1.810.454.0244 ::Message sent from iPhone Connect @ www.WayneLawson.com. On Mar 31, 2011, at 10:54 PM, Jeff Garvas j...@cia.net wrote: After having the chance to attend the IPX 5 day bootcamp (amazing experience!) I've been trying to figure out how they accomplished something. Those of you who have been there will know what I'm talking about, but for those of you who have not imagine this. The phones on your desk are powered by a local switch (or so I assume) but they're effectively plugged into your pods to the point that the MAC addresses appear on your pods in Michigan while you're sitting in San Jose or Columbus. This is probably exactly the same thing you'd see in the lab regardless of where you sit I would imagine. (Where the MAC of the phone in front of you shows up in the respective devices thousands of miles away) So how did they do that? I've sat here trying to come up with a 1:1 L2 bridge solution, maybe l2tp, or maybe some kind of per-port GRE tunnel and I keep running into limitations of the ipbase IOS I need to upgrade.. The answer is probably quite easy and I'm just missing it or over analyzing it. The first reason I'm asking is because I'm trying to find a way to simulate it for studying purposes. I have a 3750 at my desk and a 3750 in a cabinet with a trunk between them across a single ethernet connection and I'd like to hang phones off that local 3750 for POE and use the remote 3750 to patch into the back of BR2, the back of BR1, etc. I'm not trying to extend a simple vlan as much as I'm trying to make it appear as if each phone is plugged into its respective BR/HQ interface when in fact its not. The second reason is because it's driving me nuts that I can't figure out how they did it on my own. Anyone know the trick or have ideas? -Jeff Oh, and this is why you should rent rack time from proctorlabs. You spend too much time goofing around with your lab gear working on totally unrelated concepts unless you're thinking of taking the R/S later. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 回复： how to configure cucme to support phoneview
Try this: http://lmgtfy.com/?q=ipexpert+blog+phoneview Welcome to the Internet. El mié, 30-03-2011 a las 09:13 +0800, bruno escribió: hello bo, I have check the ipexpert blog ,can't find it. did u have the link? Best Regards, Bruno -- 原始邮件 -- 发件人: Bo Gaobga...@gmail.com; 发送时间: 2011年3月29日(星期二) 晚上9:15 收件人: brunobruno.juni...@gmail.com; 抄送: ccie_voiceccie_voice@onlinestudylist.com; 主题: Re: [OSL | CCIE_Voice] how to configure cucme to support phoneview Bruno, Please check IP Experts' website, I remember Vik had a blog about how to configure phoneview on it. Bo 2011/3/28 bruno bruno.juni...@gmail.com hello guys, ?? great news Unified FX release their lab version. how to configure cucme to support phoneview??? i can not find any tutorial on their website. could someone help？ ?? Best Regards, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST does not work
Looks like the calls are going through dial-peer 9 as soon a you hit the 1. What do you have on that dial-peer ? Another option would be to configure SIP dial rules so the phone will wait to get all 4 digits before sending the invite to CME. El lun, 28-03-2011 a las 20:11 +0200, Roig Borrell, Francesc Xavier escribió: Hi all, I have problems with SIP SRST. The SIP phone registers ok in SRST mode, but I can’t make or receive any type of calls (internal or external). Analyzing Internal calls I’ve seen that when PhoneA (1001) dials Phone B (1002). Only the first number of the called number is sent. The debug ccsip messages shows the first invite with the 1 but there are not any notify messages for kpml with all the other digits Have you found any problem with SRST? Is there anything I am missing in the config? Thanks in advance! voice service voip allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface FastEthernet0/0.240 bind media source-interface FastEthernet0/0.240 registrar server voice register global system message SRST max-dn 4 max-pool 4 ! voice register pool 1 id network 192.168.21.0 mask 255.255.255.0 dtmf-relay rtp-nte codec g711ulaw BRANCH1#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 1002 redirect ip2ip session target ipv4:192.168.21.69:5060 session protocol sipv2 dtmf-relay rtp-nte digit collect kpml codec g711ulaw bytes 160 after-hours-exempt FALSE dial-peer voice 40002 voip destination-pattern 1001 redirect ip2ip session target ipv4:192.168.21.68:5060 session protocol sipv2 dtmf-relay rtp-nte digit collect kpml codec g711ulaw bytes 160 after-hours-exempt FALSE BRANCH1# *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersCore: Calling Number=1, Called Number=1, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1 *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=9 *Mar 28 18:58:38.216: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1001, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.216: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40002 *Mar 28 18:58:38.220: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1001, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.220: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40002 *Mar 28 18:58:38.220: //-1/3968292780B0/DPM/dpAssociateIncomingPeerCore: Calling Number=1001, Called Number=1, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.220: //-1/3968292780B0/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40002 *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersCore: Calling Number=, Called Number=1, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1 *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersCore: Result=Partial Matches(1) after DP_MATCH_DEST *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersMoreArg: Result=MORE_DIGITS_NEEDED(1) Francesc Xavier Roig Borrell Network Senior Consultant Integración de Sistemas y Tecnología C/Santander, 49-51 Barcelona 08020 Tel. Fijo: (+34) 934965108 ext 64697 Tel. Móvil / Fax: (+34) 647 32 21 55 (24697) / (+34) 934965161 email: francesc.ro...@tecnocom.es http://www.tecnocom.es Por favor, antes de imprimir este mensaje, asegúrate de que es necesario. Ayudemos a cuidar el medio ambiente Este mensaje puede contener información confidencial o privilegiada. Si le ha llegado por error, rogamos no haga uso del mismo, avise al remitente y bórrelo. Consulte aviso legal This message may contain confidential or privileged information. If it has been sent to
Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incoming traffic
To me for incoming traffic means applying an inbound service policy on a switch interface. El dom, 27-03-2011 a las 19:08 +0300, Rogers Ochieng escribió: With CoS mapping to a queue and doing share/shape on the trunk port to router On 27 March 2011 18:43, a...@ipcomconsult.com wrote: Hey Experts Anybody can clarify on this topic? How to GUARANTEE bandwidth for incoming traffic on 3750? Thanks - Forwarded message from a...@ipcomconsult.com - Date: Sat, 12 Mar 2011 02:12:11 -0700 From: a...@ipcomconsult.com Reply-To: a...@ipcomconsult.com Subject: [OSL | CCIE_Voice] 3750 bandwidth guarantee for incoming traffic To: ccie_voice@onlinestudylist.com Hi guys Anybody can advise on how to GUARANTEE bandwidth for incoming traffic (let's say MGCP) on 3750? Policy-map as I understand can only police it but can not guarantee the bandwidth. Do you have to put it in Q2 removing any other traffic from it? Any alternative solutions? Tnx Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com - End forwarded message - ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Test VPIM between CUE and CUC - DNS client / server ?
I usually do all the DNS on the HQ router. The configuration is all on the IOS Configuration Guides. Not sure if it may really show up on the lab, if the lab would have a DNS server already setup. Would make sense to me to reduce configuration time from the DNS part as this is not what you are tested on, I guess. But for the lab practice, you could do it on IOS like I do, or any Windows server box. However, I just wanted to do learn it with IOS as I know is the resource that will be available for sure. HTH, El sáb, 19-03-2011 a las 13:52 +0300, Wael Agina escribió: Dear All,, To test VPIM between CUE and CUC we need DNS to resolve names to IPs. What is the simplest way to test this in my local Home LAB ? I have one server running windows 7 with vmware imaes and another normal win xp machine. WHat is the best way to run DNS easy and integrate it to CUE and CUC ? Any Idea ? Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MOH across GK-controlled trunk?
Shingei, That's a very good point, as it says so specifically on the documentation. What confuses me then is that on the debugs from Michael, you can see the IGMP request is taking place, so this means that the CME has actually sent it as it must have been instructed to do so through the H.245 messages. In such case, it could be a problem with multicast routing then, like Michael was saying. My suggestion to rule this out would be to configure have the phones on br2 fallback to srst while configuring cme-srst. Then make sure the phones are enabled for multicast moh and test by placing moh between two br2 phones to isolate other multicast routing problems. Anyway, my skills on troubleshooting multicast moh go only up to some point. As sounds like an interesting task, will see if I can try it tomorrow at lab. Cheeers, El lun, 21-03-2011 a las 12:31 +0800, ShinGei Yong escribió: Hi Michael, I believe you may just ignored that partThe br2 phone need to be registered with cucm,not with cme This condition is valid only when you have centralized multisite deployment in non-fallback/fallback mode. As per your configuration,obviously this is a distributed multisite deployment,which you have GK to controlled the 2 different clusters. I'm not too sure you're meet this criteria: Distributed Multisite Deployment source:http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/moh.html#wp1043948 Cisco Unified CM 7.1(2) and later releases support multicast MoH on intercluster calls using an intercluster trunk (ICT) or SIP trunk. This feature adds support for endpoints in one Unified CM cluster to hear multicast MoH streamed from another Unified CM cluster, while making more efficient use of intercluster bandwidth. A properly designed IP Multicast environment is required to take advantage of this feature. Earlier releases of Unified CM allow for only unicast MoH on intercluster calls, which requires configuration of at least one unicast MoH resource in each Unified CM cluster if MoH on the ICT or SIP trunk is desired. As the explanation above stated between UCM to UCM cluster,i'm not sure UCME will face the same.I might be wrong,but just the idea. Since your UCME is a standalone cluster, i would try out unicast on that, and believe that should no difference between using multi or uni on ur UCME. Thanks Shingei. On Mon, Mar 21, 2011 at 6:06 AM, Michael Luo hout...@gmail.com wrote: I was trying to understand this part - The br2 phone need to be registered with cucm. Not with cme. Why was that? From my troubleshooting, it looks like the CME phone was listening to the correct IP address/port - 239.1.1.1 16384. But didn't receive any MOH music. I was under the impression that the music was continuously being played by the router. The CME phones just need to tune to the right channel to receive music. Thanks! Michael On Sun, Mar 20, 2011 at 3:24 PM, Jimmy batraji...@yahoo.com wrote: Mate Coorect me if I am wrong. Doesn't matter if it's pstn or gatekeeper trunk. When cucm phone press hold button. The phone at cme site will will receive moh from cucm. For moh from flash to work. The br2 phone need to be registered with cucm. Not with cme. I hope I am making sense. I am not good at writing emails. Regards Sent from my iPad On Mar 21, 2011, at 8:17 AM, Michael Luo hout...@gmail.com wrote: One thing I noticed was the word stopped in show ip mroute output. This seems to be the problem by comparing with a working one. But I don't know how to fix it. Thanks! R3#sh ip mro IP Multicast Routing Table Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C - Connected, L - Local, P - Pruned, R - RP-bit set, F - Register flag, T - SPT-bit set, J - Join SPT, M - MSDP created entry, X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement, U -
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
It's difficult to ignore it if it is addressed to a specialized mailing list. It also takes extra time to sort out the crap that does not belong from the important emails. I'm sure you will get some attention at the cisco support forums and 0 rejection. El mié, 16-03-2011 a las 17:35 -0400, ccielabrat escribió: Right, thats why I included OT : (Off topic) in the subject line. That way, for people who don't want to be bothered can simply ignore it. On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.com wrote: 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Need suggestion on MVA
I don't understand the 2nd part either. Are you sure you are paraphrasing it correctly? Cheers. El mié, 16-03-2011 a las 23:58 +0530, Rahul Kapor escribió: Hi All, Finally My MVA started working ! Need your suggestion for following question PSTN phone line 4 can call in and make outbound Intl call. also allow SiteB for PSTN phone line 4 to call MVA , I guess SNR for number which is assigned to line 4 should be working and to allow siteB , do i need to create bogus remote destination number ie SNR for all site B user ? i am not sure about 2nd part ? what is your approach ? thx, Rahul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I failed because of MGCP issue- need help
That is not an error. El vie, 11-03-2011 a las 09:42 +0300, Support escribió: Dear Frineds, I failed in my first attempt as I could not register HQ and BR1 gateway in CUCM as MGCP GW. My router name for HQ was R1 and for branch R2 and Domain name configured on both routers as ccievoice.com So I added in CUCM as R1.ccievoice.com and R2.ccievoice.com, as it is hostname+domainname. But I could not get registered in CUCM I was getting error like below: Feb 20 13:00:00.581 : MGCP Packet received from 142.100.64.11:2427-à A UEP 7 S0/SU0/ds1...@r1.ccievoice.com MGCP 0.1 Thanks Raj __ Information from ESET NOD32 Antivirus, version of virus signature database 5943 (20110310) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volume 1 lab 5 (Transformation to 7 and 10 digits) to call 911
You don't need to check traces. You need to check your configuration. El sáb, 12-03-2011 a las 01:33 +0200, Ahmed Ellboudy escribió: Dear All, I am working on this lab and I can call 911, but the PSTN phone display the 4 digits only for HQ and BR1 while in the question it was requested to display 7 digits for HQ and 10 digits for BR1 , so can anyone guide me for the traces I need to collect to get my mistake ? Thanks, Ahmed Ellboudy | CCNP, CCVP. Networking Team Leader Raya IT - Professional Networking Services Mobile: +20100770837 Tel : +20238276000 Ext. 2338 Fax : +20238372930 Email : ahmed_ellbo...@rayacorp.com Address : El Motamayez District - 6th of October cid:image001.jpg@01CB8A26.89E6B660 Disclaimer: NOTICE The information contained in this message is confidential and is intended for the addressee(s) only. If you have received this message in error or there are any problems please notify the originator immediately. The unauthorized use, disclosure, copying or alteration of this message is strictly forbidden. Raya will not be liable for direct, special, indirect or consequential damages arising from alteration of the contents of this message by a third party or as a result of any malicious code or virus being passed on. Views expressed in this communication are not necessarily those of Raya.If you have received this message in error, please notify the sender immediately by email, facsimile or telephone and return and/or destroy the original message. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
