Re: [OSL | CCIE_Voice] MVA partial match
It is ..yes confusing Partial is number of digits selected from left ( decide the number based on your "deb isdn q931" ) not displayed on phone screen. Complete match is the number in the MVA configuration. If you select complete match debug isdn q931 should show exact number. Look for page help for more info. Thanks Shrini On 3/27/2011 10:20 PM, Erwan Erwan wrote: hi all, can someone explain bit on MVA logic or flow ? I config MVA "Complete match " service parameter in UCM , but still I called from any number , it prompt for pin (as if it match) Thanks in advyou ___ 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] CME: assign blf-speed-dial to a specific button?
I think you'll have to configure 3 place holder ephone-dn's with no number say ephone-dn 5 dual-line ephone-dn 6 dual-line ephone-dn 7 dual-line then button 2:5 3:6 4:7 this will force the blf speed dial to be pushed to button 5 On 28 March 2011 07:31, Michael Luo wrote: > I'm using CME 7.1 and 7965 phone. > > Let say, I want to configure a blf-speed-dial on the 5th button on the > phone. How should I configure that? I know the "blf-speed-dial" command. > But I couldn't figure out how to assign it to a specific button. > > Thanks! > Michael > > ___ > 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
[OSL | CCIE_Voice] Voiceview error
hi all, When I press the Voiceview "Listen" , it said :Unknown Error ,report to System Administrator " all other Voiceview work fine, wondering what i miss here? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MVA partial match
hi all, can someone explain bit on MVA logic or flow ? I config MVA "Complete match " service parameter in UCM , but still I called from any number , it prompt for pin (as if it match) Thanks in adv ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME: assign blf-speed-dial to a specific button?
Michael, It automatically assigns to the next available button. Thanks Shrini _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Luo Sent: Sunday, March 27, 2011 9:32 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME: assign blf-speed-dial to a specific button? I'm using CME 7.1 and 7965 phone. Let say, I want to configure a blf-speed-dial on the 5th button on the phone. How should I configure that? I know the "blf-speed-dial" command. But I couldn't figure out how to assign it to a specific button. Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME: assign blf-speed-dial to a specific button?
I'm using CME 7.1 and 7965 phone. Let say, I want to configure a blf-speed-dial on the 5th button on the phone. How should I configure that? I know the "blf-speed-dial" command. But I couldn't figure out how to assign it to a specific button. Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization
I'm actually doing that lab right now :) Make sure that "use device pool calling transformation css" is not checked and assign the css to the phone. Reboot the phone. Might require a no mgcp and mgcp on the gateways. On Sun, Mar 27, 2011 at 10:50 AM, George Goglidze wrote: > Hi Roger, > > I checked the replication on a cli on both, PUB/SUB: > admin:show perf query class "Number of Replicates Created and State of > Replication" > ==>query class : > > - Perf class (Number of Replicates Created and State of Replication) has > instances and values: > ReplicateCount -> Number of Replicates Created = 412 > ReplicateCount -> Replicate_State= 2 > > > I know the config is correct, because if I stop Call manager service on the > Sub, or remove it from the Call Manager group, everything works fine... So > it has to be good. > > I don't want to loose much time with this, as I'm with proctorlabs online > rack rental, so I refer to finish the lab. > at the end if I finish it much earlier, then I might do it, and reset the > replication between sub/pub. > > Thanks for anwering Roger! > > Cheers, > > On Sun, Mar 27, 2011 at 3:45 PM, Roger Carpio wrote: > >> Hello George, >> >> Very rarely I've seen replication status 2 and weird behaviors happen. >> Have you checked replication anywhere else? CLI / Unified Reporting? >> >> If you've done this many times and you're sure the configuration has been >> done correctly; drop this replication and reset it "cluster wide". I hope it >> helps. >> >> Regards, >> Roger Carpio. >> >> On Sun, Mar 27, 2011 at 8:28 AM, George Goglidze wrote: >> >>> Hi all, >>> >>> I was wondering if anyone has seen this before. >>> >>> There is a requirement to show in Missed/Received directory numbers in >>> E164 format, but in alerting/connected state it should be localized for >>> user. >>> Subscriber to 7digit, national to 10digit, international to 011 >>> >>> I have configured correct prefixes on the incoming gateways, so >>> Missed/Received directory is fine! It shows correct values with the +sign >>> E164 format. >>> >>> And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani, >>> which are in their corresponding css-norm-hq-ani, css-norm-br1-ani, >>> css-norm-br2-ani >>> and I've applied them to the corresponding Device Pools, in "Calling >>> Party Transformation CSS". >>> >>> But when the call comes in, I still get in alerting/connected state E164 >>> format. >>> >>> All my devices phones and gateways are registered to Sub first, and in >>> failed to Publisher. >>> >>> If I change this and I register all the devices to the Publisher, instead >>> of a subscriber, changing it in Callmanager group configuration, then it all >>> works as expected. >>> Therefore I thought maybe Sub is not replicating correctly, but I've >>> checked and the replication Status is 2! which means it's replicating fine. >>> I have even restarted the Subscriber but it didn't fix the issue. >>> >>> I am out of ideas, and I've done it many times in prior Labs, and it has >>> always worked fine! >>> I've searched through CCM Service Parameters, and couldn't find anything >>> that could affect this behaviour. >>> >>> If anyone has seen this, please let me know... >>> >>> Regards, >>> >>> ___ >>> 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] Fwd: 3750 bandwidth guarantee for incomingtraffic
Brian, I was thinking about this similar way. I guessed it could be achieved by putting the traffic in question into PQ, removing everything else from there and then policing it as required with service policy. However I suspect it could be not the way they will want as the solution does not look very clear. Can anybody advise any other way to accomplish this? Thanks Alex Sent from my BlackBerry Wireless Handheld -Original Message- From: Brian Mulgrew Sender: ccie_voice-boun...@onlinestudylist.com Date: Sun, 27 Mar 2011 21:55:17 To: Randall SaborÃo Cubero Cc: Subject: Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incoming traffic ___ 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] Fwd: 3750 bandwidth guarantee for incomingtraffic
Tnx Randall I think this way u can only police the traffic but not guarantee the bandwidth. Alex Sent from my BlackBerry Wireless Handheld -Original Message- From: Randall Saborío Cubero Date: Sun, 27 Mar 2011 14:19:46 To: Rogers Ochieng Cc: ; Subject: 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, 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] Fwd: 3750 bandwidth guarantee for incoming traffic
I find this a confusing one as I would read it as associating particular traffic types to the inbound priority queue, but I dont really understand how to do this: On egress we would define the dedicated PQ and map DSCP / COS values to it - in the event of congestion PQ is always served first so BW is guaranteed. On ingress as both queues are always shared (even when PQ is enabled) - then how do we allocate our DSCP / COS values to the inbound PQ? We can set them to various queue ids and thresholds but I cant see a way to set them to the PQ/ Is this done implicitly by setting the queue id as PQ and threshold ID as 3? e.g. mls qos srr-queue input dscp-map queue 2 threshold 3 46 Any thoughts appreciated! Brian On Sun, Mar 27, 2011 at 9:19 PM, Randall Saborío Cubero wrote: > 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, 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 > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
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, 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] Home lab PSTN
Thank you very much Wael, Matt, CCIE_Voice, duy and Bill for your quick and helpful responses. Regards, Shaun P On Sun, Mar 27, 2011 at 1:36 PM, Wael Agina wrote: > Dear SHaun > > IP expert has two models of numbering though volume two. > So if you checked you will find that you can use one model only and arrange > yourself with it. > I am doing this in my home lab and doing all IPX labs. > > Regards, > Wael Agina > > > On Sun, Mar 27, 2011 at 7:19 PM, Shaun P wrote: > >> Thank you Wael for the information. >> The IPExpert configuration has initial and final pstn configuration for >> every lab in Volume 1. Do we need to load the pstn configuration on the pstn >> router for each Volume 1 lab? >> >> Best, >> >> Shaun P >> >> >> On Sat, Mar 26, 2011 at 8:37 PM, Wael Agina wrote: >> >>> Dear Shaun, >>> >>> I have 2811 as both PSTN and FR WAN RTR, so you can use single >>> router. >>> I think they connect the PSTN to FR just to have an IP Connectivity - so >>> you can manage it remotly - and to simulate remote GK scenarios. >>> >>> You can use single 2811 as both and it will work fine. >>> >>> Regards, >>> Wael Agina >>> >>> On Sun, Mar 27, 2011 at 1:52 AM, Shaun P wrote: >>> Hi Voicers: I have a home lab with 2811 as PSTN router and a 2522 as frame relay router. Do I need to add a serial interface to my PSTN router and connect it to the frame relay router? I see the IPexpert PSTN configuration has ospf in it. What would be the minimum PVDM2 requirement to complete all the labs? Thank you in advance and keep labbing Shaun P ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com >>> >>> >>> -- >>> >>> Thanks and Best Regards, >>> Wael Agina >>> >> >> > > > -- > > Thanks and Best Regards, > Wael Agina > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA using sip trunk
no just play around w sip trunk --- On Sun, 3/27/11, mihal caro wrote: From: mihal caro Subject: Re: [OSL | CCIE_Voice] MVA using sip trunk To: "Erwan Erwan" Cc: ccie_voice@onlinestudylist.com Received: Sunday, March 27, 2011, 6:14 PM hi there I had the same problem SIp trunk MVA is support on CUCM 7.13 . If is for the CCIE lab you need to configure it as H323. regards. On 27 March 2011 07:24, Erwan Erwan wrote: hi all, I tried to config MVA in HQ site, Scenario - I dial in from PSTN to 5999 (MVA #) it gave me prompt for pin and press "1" to dial any outside number (911) 1. If I use SIP trunk It will gave me error saying (below is the complete "debug ccsip message" "Sent: SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' " 2. However if I use, H323 GW , call out by MVA to outside (ie :911 ) work fine Question - - wondering any parameter in SIP\ trunk I need to check ? tks in adv Complete debug === PSTN-WAN# Mar 27 06:07:26.026: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' Via: SIP/2.0/TCP 10.10.210.10:5060;branch=z9hG4bK1861aabffa From: ;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-20651851 To: ;tag=1343344-12F6 Call-ID: 7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10 CSeq: 101 INVITE Reason: Q.850;cause=100 Content-Length: 0 ___ 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] MVA using sip trunk
hmm i use, proctor lab . not sure what ver ? 7.0 maybe --- On Sun, 3/27/11, George Goglidze wrote: From: George Goglidze Subject: Re: [OSL | CCIE_Voice] MVA using sip trunk To: "Erwan Erwan" Cc: ccie_voice@onlinestudylist.com Received: Sunday, March 27, 2011, 6:11 PM which version of CUCM are you using? because MVA on SIP is supported starting 7.1.3 I think... On Sun, Mar 27, 2011 at 7:24 AM, Erwan Erwan wrote: hi all, I tried to config MVA in HQ site, Scenario - I dial in from PSTN to 5999 (MVA #) it gave me prompt for pin and press "1" to dial any outside number (911) 1. If I use SIP trunk It will gave me error saying (below is the complete "debug ccsip message" "Sent: SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' " 2. However if I use, H323 GW , call out by MVA to outside (ie :911 ) work fine Question - - wondering any parameter in SIP\ trunk I need to check ? tks in adv Complete debug === PSTN-WAN# Mar 27 06:07:26.026: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' Via: SIP/2.0/TCP 10.10.210.10:5060;branch=z9hG4bK1861aabffa From: ;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-20651851 To: ;tag=1343344-12F6 Call-ID: 7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10 CSeq: 101 INVITE Reason: Q.850;cause=100 Content-Length: 0 ___ 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] Remove prefix 9 from called number on phone screen
To get the called party transformation of the RP to show on the display of the phone with a H.323 gw you need to add this to your configuration on the H.323 gw voice service voip no supplementary-service h225-notify cid-update Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: Michael Luo [hout...@gmail.com] Skickat: den 27 mars 2011 20:07 Till: ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen I'm using CUCM 7.1.2 with H.323 gateway. In H323 GW, the dial-peer for local number is 9 with 7-digit. For example "destination-pattern 9...". When user make a local call, the IP phone will display the called number with prefix 9. For example "To: 95551212". If we're not allowed to change dial-peer on GW, what's the easiest way to remove the prefix 9 from phone screen? I tried to use route pattern to discard the 9 then prefix 9 in route group. But it's still showing up. Do I have to do called-party transform pattern? I was trying to avoid that. Thanks! Michael___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Home lab PSTN
Hi Shaun: I have 4 2811s in my lab. The minimum DSP resource I have on each is 2 PVDM2-16s. I have a voice T1/E1 from each site to the PSTN and a data T1/E1 to the PSTN. The OSPF is handled over the data connection via frame-relay. On Sat, Mar 26, 2011 at 4:52 PM, Shaun P wrote: > Hi Voicers: > > I have a home lab with 2811 as PSTN router and a 2522 as frame relay > router. Do I need to add a serial interface to my PSTN router and connect it > to the frame relay router? I see the IPexpert PSTN configuration has ospf in > it. > > What would be the minimum PVDM2 requirement to complete all the labs? > > Thank you in advance and keep labbing > > Shaun P > > ___ > 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] Remove prefix 9 from called number on phone screen
Hi Michael - Voice service voip no supplementary-service h225-notify cid-update Should restrict the h225 notify clid back to the CUCM phone hth Brian On Sun, Mar 27, 2011 at 7:07 PM, Michael Luo wrote: > I'm using CUCM 7.1.2 with H.323 gateway. > > In H323 GW, the dial-peer for local number is 9 with 7-digit. For example > "destination-pattern 9...". > > When user make a local call, the IP phone will display the called number > with prefix 9. For example "To: 95551212". > > If we're not allowed to change dial-peer on GW, what's the easiest way to > remove the prefix 9 from phone screen? > > I tried to use route pattern to discard the 9 then prefix 9 in route > group. But it's still showing up. Do I have to do called-party transform > pattern? I was trying to avoid that. > > Thanks! > Michael > > ___ > 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] Remove prefix 9 from called number on phone screen
You have to run the following on the gateway: 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 h225-notify cid-update* This tells the gateway not to send the updated info back to CUCM. You still need to do the display manipulation in the RP and the PSTN manipulation in the RL. On Sun, Mar 27, 2011 at 12:55 PM, Michael Luo wrote: > Here's what I found out from CCM traces: > > 1) When the call ring out. CCM instructs the phone display the called > number as "5551212" which is the desired behavior. > > 2) CCM receives a H.225 NOTIFY message from the GW with "called number = > 95551212". > > 3) CCM instructs the phone to update the screen as "To: 95551212". > > Any way to change this behavior? > > Thanks! > Michael > > On Sun, Mar 27, 2011 at 1:40 PM, Tamer Ismail wrote: > >> Hello, >> >> All manipulations done in route pattern will not make your requirements. >> >> You have to pre-dot in route pattern, then prefix in route list. >> >> >> >> Best regards, >> >> Tamer Ismail >> >> >> >> *From:* ccie_voice-boun...@onlinestudylist.com [mailto: >> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Michael Luo >> *Sent:* Sunday, March 27, 2011 8:08 PM >> *To:* ccie_voice@onlinestudylist.com >> *Subject:* [OSL | CCIE_Voice] Remove prefix 9 from called number on phone >> screen >> >> >> >> I'm using CUCM 7.1.2 with H.323 gateway. >> >> In H323 GW, the dial-peer for local number is 9 with 7-digit. For example >> "destination-pattern 9...". >> >> When user make a local call, the IP phone will display the called number >> with prefix 9. For example "To: 95551212". >> >> If we're not allowed to change dial-peer on GW, what's the easiest way to >> remove the prefix 9 from phone screen? >> >> I tried to use route pattern to discard the 9 then prefix 9 in route >> group. But it's still showing up. Do I have to do called-party transform >> pattern? I was trying to avoid that. >> >> Thanks! >> Michael >> > > > ___ > 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 can't get MGCP to register to lo0
My mistake Earl - I read the subject heading as generic registration issues, rather than specifically to l0. Thks for the tip! Brian On 27/03/2011, Hough, Earl wrote: > Brian, > > But, if it's an issue of using the FQDN vs. just the hostname as to how the > router resolves its name, wouldn't it be a case of it never registering, > rather than registering with the wrong binding address? > > Easiest way to see what the router expects to use for its domain name is to > issue "show ccm-manager" and use the value that shows up for "MGCP Domain > Name: x" > > > Earl Hough > CCIE #16508 (R&S/Security/Voice) > > > -Original Message- > From: Brian Mulgrew [mailto:btmulg...@gmail.com] > Sent: Sunday, March 27, 2011 2:28 PM > To: Hough, Earl; Randall Crumm; ccie_voice@onlinestudylist.com > Subject: Re: [OSL | CCIE_Voice] I can't get MGCP to register to lo0 > > Hi - Is DNS being used? May need the fqdn of the gateway. > > Hth > Brian > > On 27/03/2011, Hough, Earl wrote: >> Randal, >> >> It should work if you just do these 4 steps: >> >> On the gateway: >> >> >> 1) no mgcp >> >> 2) no mgcp bind control >> >> 3) mgcp bind control source-interface lo0 >> >> 4) mgcp >> >> This should alert UCM as to the change in source address and if you hit >> "Find" on your gateways page again, it should show the updated loopback >> address of the MGCP gateway. >> >> >> Earl Hough >> CCIE #16508 (R&S/Security/Voice) >> >> From: ccie_voice-boun...@onlinestudylist.com >> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm >> Sent: Sunday, March 27, 2011 12:54 PM >> To: ccie_voice@onlinestudylist.com >> Subject: [OSL | CCIE_Voice] I can't get MGCP to register to lo0 >> >> I've tried everything at least 4 times now. >> >> No mgcp bind control >> No mgcp bind media >> No isdn bind-l3 ccm >> Rebooting the rtr >> Deleteing the gw in CUCM >> >> CUCM always will not see the lo0 ip address. >> >> At this point anything is welcome... >> >> UGGG! >> >> >> Best Regards, >> >> Randall Crumm >> Voice and Video Architect >> Global Networking & Telecom >> [cid:image001.gif@01CBEC82.54BD0B10] >> >> [cid:image002.gif@01CBEC82.54BD0B10] >> >> 1007 Gibraltar Drive >> Milpitas, CA 95035 >> >> Direct +1 408.576.7344 >> VoIP .100.7344 >> >> >> Creating value that increases >> customer competitiveness >> >> Legal Disclaimer: The information contained in this message may be >> privileged and confidential. It is intended to be read only by the >> individual or entity to whom it is addressed or by their designee. If the >> reader of this message is not the intended recipient, you are on notice >> that >> any distribution of this message, in any form, is strictly prohibited. If >> you have received this message in error, please immediately notify the >> sender and delete or destroy any copy of this message >> _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ >> >> 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. >> >> > > -- > Sent from my mobile device > _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ > > 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. > > > -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen
Here's what I found out from CCM traces: 1) When the call ring out. CCM instructs the phone display the called number as "5551212" which is the desired behavior. 2) CCM receives a H.225 NOTIFY message from the GW with "called number = 95551212". 3) CCM instructs the phone to update the screen as "To: 95551212". Any way to change this behavior? Thanks! Michael On Sun, Mar 27, 2011 at 1:40 PM, Tamer Ismail wrote: > Hello, > > All manipulations done in route pattern will not make your requirements. > > You have to pre-dot in route pattern, then prefix in route list. > > > > Best regards, > > Tamer Ismail > > > > *From:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Michael Luo > *Sent:* Sunday, March 27, 2011 8:08 PM > *To:* ccie_voice@onlinestudylist.com > *Subject:* [OSL | CCIE_Voice] Remove prefix 9 from called number on phone > screen > > > > I'm using CUCM 7.1.2 with H.323 gateway. > > In H323 GW, the dial-peer for local number is 9 with 7-digit. For example > "destination-pattern 9...". > > When user make a local call, the IP phone will display the called number > with prefix 9. For example "To: 95551212". > > If we're not allowed to change dial-peer on GW, what's the easiest way to > remove the prefix 9 from phone screen? > > I tried to use route pattern to discard the 9 then prefix 9 in route > group. But it's still showing up. Do I have to do called-party transform > pattern? I was trying to avoid that. > > Thanks! > Michael > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen
Thanks for the reply. That's exactly what I did: 1) In route pattern, use Pre-Dot to remove prefix 9 2) In associated route list > route group, prefix 9 But it's still showing the prefix 9 on phone's screen. Thanks! Michael On Sun, Mar 27, 2011 at 1:40 PM, Tamer Ismail wrote: > Hello, > > All manipulations done in route pattern will not make your requirements. > > You have to pre-dot in route pattern, then prefix in route list. > > > > Best regards, > > Tamer Ismail > > > > *From:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Michael Luo > *Sent:* Sunday, March 27, 2011 8:08 PM > *To:* ccie_voice@onlinestudylist.com > *Subject:* [OSL | CCIE_Voice] Remove prefix 9 from called number on phone > screen > > > > I'm using CUCM 7.1.2 with H.323 gateway. > > In H323 GW, the dial-peer for local number is 9 with 7-digit. For example > "destination-pattern 9...". > > When user make a local call, the IP phone will display the called number > with prefix 9. For example "To: 95551212". > > If we're not allowed to change dial-peer on GW, what's the easiest way to > remove the prefix 9 from phone screen? > > I tried to use route pattern to discard the 9 then prefix 9 in route > group. But it's still showing up. Do I have to do called-party transform > pattern? I was trying to avoid that. > > Thanks! > Michael > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen
Hello, All manipulations done in route pattern will not make your requirements. You have to pre-dot in route pattern, then prefix in route list. Best regards, Tamer Ismail From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Luo Sent: Sunday, March 27, 2011 8:08 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen I'm using CUCM 7.1.2 with H.323 gateway. In H323 GW, the dial-peer for local number is 9 with 7-digit. For example "destination-pattern 9...". When user make a local call, the IP phone will display the called number with prefix 9. For example "To: 95551212". If we're not allowed to change dial-peer on GW, what's the easiest way to remove the prefix 9 from phone screen? I tried to use route pattern to discard the 9 then prefix 9 in route group. But it's still showing up. Do I have to do called-party transform pattern? I was trying to avoid that. Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I can't get MGCP to register to lo0
Brian, But, if it's an issue of using the FQDN vs. just the hostname as to how the router resolves its name, wouldn't it be a case of it never registering, rather than registering with the wrong binding address? Easiest way to see what the router expects to use for its domain name is to issue "show ccm-manager" and use the value that shows up for "MGCP Domain Name: x" Earl Hough CCIE #16508 (R&S/Security/Voice) -Original Message- From: Brian Mulgrew [mailto:btmulg...@gmail.com] Sent: Sunday, March 27, 2011 2:28 PM To: Hough, Earl; Randall Crumm; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] I can't get MGCP to register to lo0 Hi - Is DNS being used? May need the fqdn of the gateway. Hth Brian On 27/03/2011, Hough, Earl wrote: > Randal, > > It should work if you just do these 4 steps: > > On the gateway: > > > 1) no mgcp > > 2) no mgcp bind control > > 3) mgcp bind control source-interface lo0 > > 4) mgcp > > This should alert UCM as to the change in source address and if you hit > "Find" on your gateways page again, it should show the updated loopback > address of the MGCP gateway. > > > Earl Hough > CCIE #16508 (R&S/Security/Voice) > > From: ccie_voice-boun...@onlinestudylist.com > [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm > Sent: Sunday, March 27, 2011 12:54 PM > To: ccie_voice@onlinestudylist.com > Subject: [OSL | CCIE_Voice] I can't get MGCP to register to lo0 > > I've tried everything at least 4 times now. > > No mgcp bind control > No mgcp bind media > No isdn bind-l3 ccm > Rebooting the rtr > Deleteing the gw in CUCM > > CUCM always will not see the lo0 ip address. > > At this point anything is welcome... > > UGGG! > > > Best Regards, > > Randall Crumm > Voice and Video Architect > Global Networking & Telecom > [cid:image001.gif@01CBEC82.54BD0B10] > > [cid:image002.gif@01CBEC82.54BD0B10] > > 1007 Gibraltar Drive > Milpitas, CA 95035 > > Direct +1 408.576.7344 > VoIP .100.7344 > > > Creating value that increases > customer competitiveness > > Legal Disclaimer: The information contained in this message may be > privileged and confidential. It is intended to be read only by the > individual or entity to whom it is addressed or by their designee. If the > reader of this message is not the intended recipient, you are on notice that > any distribution of this message, in any form, is strictly prohibited. If > you have received this message in error, please immediately notify the > sender and delete or destroy any copy of this message > _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ > > 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. > > -- Sent from my mobile device _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ 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
Re: [OSL | CCIE_Voice] I can't get MGCP to register to lo0
Hi - Is DNS being used? May need the fqdn of the gateway. Hth Brian On 27/03/2011, Hough, Earl wrote: > Randal, > > It should work if you just do these 4 steps: > > On the gateway: > > > 1) no mgcp > > 2) no mgcp bind control > > 3) mgcp bind control source-interface lo0 > > 4) mgcp > > This should alert UCM as to the change in source address and if you hit > "Find" on your gateways page again, it should show the updated loopback > address of the MGCP gateway. > > > Earl Hough > CCIE #16508 (R&S/Security/Voice) > > From: ccie_voice-boun...@onlinestudylist.com > [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm > Sent: Sunday, March 27, 2011 12:54 PM > To: ccie_voice@onlinestudylist.com > Subject: [OSL | CCIE_Voice] I can't get MGCP to register to lo0 > > I've tried everything at least 4 times now. > > No mgcp bind control > No mgcp bind media > No isdn bind-l3 ccm > Rebooting the rtr > Deleteing the gw in CUCM > > CUCM always will not see the lo0 ip address. > > At this point anything is welcome... > > UGGG! > > > Best Regards, > > Randall Crumm > Voice and Video Architect > Global Networking & Telecom > [cid:image001.gif@01CBEC82.54BD0B10] > > [cid:image002.gif@01CBEC82.54BD0B10] > > 1007 Gibraltar Drive > Milpitas, CA 95035 > > Direct +1 408.576.7344 > VoIP .100.7344 > > > Creating value that increases > customer competitiveness > > Legal Disclaimer: The information contained in this message may be > privileged and confidential. It is intended to be read only by the > individual or entity to whom it is addressed or by their designee. If the > reader of this message is not the intended recipient, you are on notice that > any distribution of this message, in any form, is strictly prohibited. If > you have received this message in error, please immediately notify the > sender and delete or destroy any copy of this message > _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ > > 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. > > -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen
I'm using CUCM 7.1.2 with H.323 gateway. In H323 GW, the dial-peer for local number is 9 with 7-digit. For example "destination-pattern 9...". When user make a local call, the IP phone will display the called number with prefix 9. For example "To: 95551212". If we're not allowed to change dial-peer on GW, what's the easiest way to remove the prefix 9 from phone screen? I tried to use route pattern to discard the 9 then prefix 9 in route group. But it's still showing up. Do I have to do called-party transform pattern? I was trying to avoid that. Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Home lab PSTN
Dear SHaun IP expert has two models of numbering though volume two. So if you checked you will find that you can use one model only and arrange yourself with it. I am doing this in my home lab and doing all IPX labs. Regards, Wael Agina On Sun, Mar 27, 2011 at 7:19 PM, Shaun P wrote: > Thank you Wael for the information. > The IPExpert configuration has initial and final pstn configuration for > every lab in Volume 1. Do we need to load the pstn configuration on the pstn > router for each Volume 1 lab? > > Best, > > Shaun P > > > On Sat, Mar 26, 2011 at 8:37 PM, Wael Agina wrote: > >> Dear Shaun, >> >> I have 2811 as both PSTN and FR WAN RTR, so you can use single >> router. >> I think they connect the PSTN to FR just to have an IP Connectivity - so >> you can manage it remotly - and to simulate remote GK scenarios. >> >> You can use single 2811 as both and it will work fine. >> >> Regards, >> Wael Agina >> >> On Sun, Mar 27, 2011 at 1:52 AM, Shaun P wrote: >> >>> Hi Voicers: >>> >>> I have a home lab with 2811 as PSTN router and a 2522 as frame relay >>> router. Do I need to add a serial interface to my PSTN router and connect it >>> to the frame relay router? I see the IPexpert PSTN configuration has ospf in >>> it. >>> >>> What would be the minimum PVDM2 requirement to complete all the labs? >>> >>> Thank you in advance and keep labbing >>> >>> Shaun P >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >>> >> >> >> -- >> >> Thanks and Best Regards, >> Wael Agina >> > > -- Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I can't get MGCP to register to lo0
Randal, It should work if you just do these 4 steps: On the gateway: 1) no mgcp 2) no mgcp bind control 3) mgcp bind control source-interface lo0 4) mgcp This should alert UCM as to the change in source address and if you hit "Find" on your gateways page again, it should show the updated loopback address of the MGCP gateway. Earl Hough CCIE #16508 (R&S/Security/Voice) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Sunday, March 27, 2011 12:54 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] I can't get MGCP to register to lo0 I've tried everything at least 4 times now. No mgcp bind control No mgcp bind media No isdn bind-l3 ccm Rebooting the rtr Deleteing the gw in CUCM CUCM always will not see the lo0 ip address. At this point anything is welcome... UGGG! Best Regards, Randall Crumm Voice and Video Architect Global Networking & Telecom [cid:image001.gif@01CBEC82.54BD0B10] [cid:image002.gif@01CBEC82.54BD0B10] 1007 Gibraltar Drive Milpitas, CA 95035 Direct +1 408.576.7344 VoIP .100.7344 Creating value that increases customer competitiveness Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ 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
Re: [OSL | CCIE_Voice] Home lab PSTN
You will need to modify your ISDN gateway and FR router to match the interfaces/slots that are applicable to your hardware. This applies to the controllers, interfaces, and voice ports. On Sun, Mar 27, 2011 at 11:19 AM, Shaun P wrote: > Thank you Wael for the information. > The IPExpert configuration has initial and final pstn configuration for > every lab in Volume 1. Do we need to load the pstn configuration on the pstn > router for each Volume 1 lab? > > Best, > > Shaun P > > On Sat, Mar 26, 2011 at 8:37 PM, Wael Agina wrote: >> >> Dear Shaun, >> >> I have 2811 as both PSTN and FR WAN RTR, so you can use single >> router. >> I think they connect the PSTN to FR just to have an IP Connectivity - so >> you can manage it remotly - and to simulate remote GK scenarios. >> >> You can use single 2811 as both and it will work fine. >> >> Regards, >> Wael Agina >> >> On Sun, Mar 27, 2011 at 1:52 AM, Shaun P wrote: >>> >>> Hi Voicers: >>> >>> I have a home lab with 2811 as PSTN router and a 2522 as frame relay >>> router. Do I need to add a serial interface to my PSTN router and connect it >>> to the frame relay router? I see the IPexpert PSTN configuration has ospf in >>> it. >>> >>> What would be the minimum PVDM2 requirement to complete all the labs? >>> >>> Thank you in advance and keep labbing >>> >>> Shaun P >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >> >> >> >> -- >> >> 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
[OSL | CCIE_Voice] I can't get MGCP to register to lo0
I've tried everything at least 4 times now. No mgcp bind control No mgcp bind media No isdn bind-l3 ccm Rebooting the rtr Deleteing the gw in CUCM CUCM always will not see the lo0 ip address. At this point anything is welcome... UGGG! Best Regards, Randall Crumm Voice and Video Architect Global Networking & Telecom 1007 Gibraltar Drive Milpitas, CA 95035 Direct +1 408.576.7344 VoIP .100.7344 Creating value that increases customer competitiveness Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message <><>___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Local Gateway/Standard Local Route Group
Only one. I realized that if two sites then then both better be in the same dialing domain and preferably have both gateways using MGCP since h323 means pushing digit manipulation down to the ios translation rules and dial-peers which will require two different RL's or two different RP's On 27 March 2011 19:24, Roger Carpio wrote: > How many sites are supposed to use this RP? If only one site is meant to > use it; SLRG will have an extra step (assigning the RG to the DP). > > Regards, > Roger Carpio. > > On Sun, Mar 27, 2011 at 9:22 AM, Rogers Ochieng > wrote: > >> If i have a requirement to only use local gateway for a route pattern, is >> it advisable to use a specific route list pointing to the route group with >> the gateway or a route list pointing to a SLRG will do? >> >> >> Rogers >> >> ___ >> 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] Local Gateway/Standard Local Route Group
How many sites are supposed to use this RP? If only one site is meant to use it; SLRG will have an extra step (assigning the RG to the DP). Regards, Roger Carpio. On Sun, Mar 27, 2011 at 9:22 AM, Rogers Ochieng wrote: > If i have a requirement to only use local gateway for a route pattern, is > it advisable to use a specific route list pointing to the route group with > the gateway or a route list pointing to a SLRG will do? > > > Rogers > > ___ > 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] Home lab PSTN
Thank you Wael for the information. The IPExpert configuration has initial and final pstn configuration for every lab in Volume 1. Do we need to load the pstn configuration on the pstn router for each Volume 1 lab? Best, Shaun P On Sat, Mar 26, 2011 at 8:37 PM, Wael Agina wrote: > Dear Shaun, > > I have 2811 as both PSTN and FR WAN RTR, so you can use single > router. > I think they connect the PSTN to FR just to have an IP Connectivity - so > you can manage it remotly - and to simulate remote GK scenarios. > > You can use single 2811 as both and it will work fine. > > Regards, > Wael Agina > > On Sun, Mar 27, 2011 at 1:52 AM, Shaun P wrote: > >> Hi Voicers: >> >> I have a home lab with 2811 as PSTN router and a 2522 as frame relay >> router. Do I need to add a serial interface to my PSTN router and connect it >> to the frame relay router? I see the IPexpert PSTN configuration has ospf in >> it. >> >> What would be the minimum PVDM2 requirement to complete all the labs? >> >> Thank you in advance and keep labbing >> >> Shaun P >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> > > > -- > > Thanks and Best Regards, > Wael Agina > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 61, Issue 151
Do you want your phones to associate to the Subscriber in preference of the Publisher? I think I achieved that in a UC implementation some time back. But I did not rely on TFTP options over DHCP etc.. It's a simple trick in CUCM itself.. Sadaseeven Saminaden (Davi) Network Engineer GSM: (230) 951 0502 Email: sadaseev...@emenetworks.com EME Networks Ltd|95, Victoria Avenue, Quatre Bornes, Mauritius| Tel: (230) 467 9007|Fax: (230) 467 9002 | | Email: emenetwo...@intnet.mu| "Excellence, Quality and Commitment" MAURITIUS ● SEYCHELLES ● MADAGASCAR Please consider the environment before printing this email Disclaimer: This communication is intended for the use of the recipient to which it is addressed, and may contain confidential, personal, and/or privileged information. Please contact us immediately if you are not the intended recipient of this communication, and do not copy, distribute, or take action relying on it. Any communication received in error, or subsequent reply, should be deleted or destroyed -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: 26 March, 2011 20:00 To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 61, Issue 151 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. RES: TFTP option 150 and 66 (Marcelo Alexandria) 2. Re: TFTP option 150 and 66 (Julien Krieger) 3. Re: TFTP option 150 and 66 (Miron Kobelski) -- Message: 1 Date: Sat, 26 Mar 2011 00:38:27 -0300 From: "Marcelo Alexandria" To: "'Shrini'" , "'Julien Krieger'" Cc: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RES: TFTP option 150 and 66 Message-ID: <003601cbeb67$471aab00$d5500100$@com.br> Content-Type: text/plain; charset="iso-8859-1" I think you can enter , but in the phone , if the first TFTP fails ..you don?t have a backup tftp. Need a simple test. J But in the lab you can use the option 150. Marcelo #27021 Voice De: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Em nome de Shrini Enviada em: sexta-feira, 25 de mar?o de 2011 23:51 Para: Julien Krieger Cc: ccie_voice@onlinestudylist.com Assunto: Re: [OSL | CCIE_Voice] TFTP option 150 and 66 That is true but I can enter multiple IP addresses with option 66. what it does ? that is my question. On 3/25/2011 6:27 PM, Julien Krieger wrote: To me, option 66 is not supposed to support a secondary TFTP server... whereas option 150 does. 2011/3/22 Shrini I want to use SUB as primary and PUB as secondary TFTP. option 150 ip option 66 ip Which is appropriate. On Cisco router 12.4T(22) both are supported. What is the difference ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Nenhum v?rus encontrado nessa mensagem recebida. Verificado por AVG - www.avgbrasil.com.br Vers?o: 9.0.894 / Banco de dados de v?rus: 271.1.1/3529 - Data de Lan?amento: 03/25/11 13:54:00 -- next part -- An HTML attachment was scrubbed... URL: -- Message: 2 Date: Sat, 26 Mar 2011 09:24:30 +0100 From: Julien Krieger To: Marcelo Alexandria Cc: ccie_voice@onlinestudylist.com, Shrini Subject: Re: [OSL | CCIE_Voice] TFTP option 150 and 66 Message-ID: Content-Type: text/plain; charset="windows-1252" Yep, Marcely is right. 2011/3/26 Marcelo Alexandria > I think you can enter , but in the phone , if the first TFTP fails ..you > don?t have a backup tftp. > > Need a simple test. > > > > J But in the lab you can use the option 150. > > > > Marcelo > > #27021 Voice > > > > *De:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *Em nome de *Shrini > *Enviada em:* sexta-feira, 25 de mar?o de 2011 23:51 > *Para:* Julien Krieger > *Cc:* ccie_voice@onlinestudylist.com > *Assunto:* Re: [OSL | CCIE_Voice] TFTP option 150 and 66 > > > > That is true but I can enter multiple IP addresses with option 66. what it > does ? that is my question. > > On 3/25/2011 6:27 PM, Julien Krieger wrote: > > To me, option 66 is not supposed to support a secondary TFTP server... > whereas option 150 does. > > 2011/3/22 Shrini > > I want to use SUB as primary and PUB as secondary TF
Re: [OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incoming traffic
With CoS mapping to a queue and doing share/shape on the trunk port to router On 27 March 2011 18:43, 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
[OSL | CCIE_Voice] Any way of getting 7961 using locally stored missed calls numbers?
Guys, Anybody knows if it's possible to get 7961/65 to use locally stored (not globalized) missed calls number? Is there any setting/parameter which can change directory behavior to one similar to 7940/60 phones? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Fwd: 3750 bandwidth guarantee for incoming traffic
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
[OSL | CCIE_Voice] Local Gateway/Standard Local Route Group
If i have a requirement to only use local gateway for a route pattern, is it advisable to use a specific route list pointing to the route group with the gateway or a route list pointing to a SLRG will do? Rogers ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization
Hi Roger, I checked the replication on a cli on both, PUB/SUB: admin:show perf query class "Number of Replicates Created and State of Replication" ==>query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount -> Number of Replicates Created = 412 ReplicateCount -> Replicate_State= 2 I know the config is correct, because if I stop Call manager service on the Sub, or remove it from the Call Manager group, everything works fine... So it has to be good. I don't want to loose much time with this, as I'm with proctorlabs online rack rental, so I refer to finish the lab. at the end if I finish it much earlier, then I might do it, and reset the replication between sub/pub. Thanks for anwering Roger! Cheers, On Sun, Mar 27, 2011 at 3:45 PM, Roger Carpio wrote: > Hello George, > > Very rarely I've seen replication status 2 and weird behaviors happen. Have > you checked replication anywhere else? CLI / Unified Reporting? > > If you've done this many times and you're sure the configuration has been > done correctly; drop this replication and reset it "cluster wide". I hope it > helps. > > Regards, > Roger Carpio. > > On Sun, Mar 27, 2011 at 8:28 AM, George Goglidze wrote: > >> Hi all, >> >> I was wondering if anyone has seen this before. >> >> There is a requirement to show in Missed/Received directory numbers in >> E164 format, but in alerting/connected state it should be localized for >> user. >> Subscriber to 7digit, national to 10digit, international to 011 >> >> I have configured correct prefixes on the incoming gateways, so >> Missed/Received directory is fine! It shows correct values with the +sign >> E164 format. >> >> And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani, >> which are in their corresponding css-norm-hq-ani, css-norm-br1-ani, >> css-norm-br2-ani >> and I've applied them to the corresponding Device Pools, in "Calling Party >> Transformation CSS". >> >> But when the call comes in, I still get in alerting/connected state E164 >> format. >> >> All my devices phones and gateways are registered to Sub first, and in >> failed to Publisher. >> >> If I change this and I register all the devices to the Publisher, instead >> of a subscriber, changing it in Callmanager group configuration, then it all >> works as expected. >> Therefore I thought maybe Sub is not replicating correctly, but I've >> checked and the replication Status is 2! which means it's replicating fine. >> I have even restarted the Subscriber but it didn't fix the issue. >> >> I am out of ideas, and I've done it many times in prior Labs, and it has >> always worked fine! >> I've searched through CCM Service Parameters, and couldn't find anything >> that could affect this behaviour. >> >> If anyone has seen this, please let me know... >> >> Regards, >> >> ___ >> 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] WB II, Lab 10, Calling Party Localization
Hello George, Very rarely I've seen replication status 2 and weird behaviors happen. Have you checked replication anywhere else? CLI / Unified Reporting? If you've done this many times and you're sure the configuration has been done correctly; drop this replication and reset it "cluster wide". I hope it helps. Regards, Roger Carpio. On Sun, Mar 27, 2011 at 8:28 AM, George Goglidze wrote: > Hi all, > > I was wondering if anyone has seen this before. > > There is a requirement to show in Missed/Received directory numbers in E164 > format, but in alerting/connected state it should be localized for user. > Subscriber to 7digit, national to 10digit, international to 011 > > I have configured correct prefixes on the incoming gateways, so > Missed/Received directory is fine! It shows correct values with the +sign > E164 format. > > And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani, > which are in their corresponding css-norm-hq-ani, css-norm-br1-ani, > css-norm-br2-ani > and I've applied them to the corresponding Device Pools, in "Calling Party > Transformation CSS". > > But when the call comes in, I still get in alerting/connected state E164 > format. > > All my devices phones and gateways are registered to Sub first, and in > failed to Publisher. > > If I change this and I register all the devices to the Publisher, instead > of a subscriber, changing it in Callmanager group configuration, then it all > works as expected. > Therefore I thought maybe Sub is not replicating correctly, but I've > checked and the replication Status is 2! which means it's replicating fine. > I have even restarted the Subscriber but it didn't fix the issue. > > I am out of ideas, and I've done it many times in prior Labs, and it has > always worked fine! > I've searched through CCM Service Parameters, and couldn't find anything > that could affect this behaviour. > > If anyone has seen this, please let me know... > > Regards, > > ___ > 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
[OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization
Hi all, I was wondering if anyone has seen this before. There is a requirement to show in Missed/Received directory numbers in E164 format, but in alerting/connected state it should be localized for user. Subscriber to 7digit, national to 10digit, international to 011 I have configured correct prefixes on the incoming gateways, so Missed/Received directory is fine! It shows correct values with the +sign E164 format. And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani, which are in their corresponding css-norm-hq-ani, css-norm-br1-ani, css-norm-br2-ani and I've applied them to the corresponding Device Pools, in "Calling Party Transformation CSS". But when the call comes in, I still get in alerting/connected state E164 format. All my devices phones and gateways are registered to Sub first, and in failed to Publisher. If I change this and I register all the devices to the Publisher, instead of a subscriber, changing it in Callmanager group configuration, then it all works as expected. Therefore I thought maybe Sub is not replicating correctly, but I've checked and the replication Status is 2! which means it's replicating fine. I have even restarted the Subscriber but it didn't fix the issue. I am out of ideas, and I've done it many times in prior Labs, and it has always worked fine! I've searched through CCM Service Parameters, and couldn't find anything that could affect this behaviour. If anyone has seen this, please let me know... Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA and partition
You MUST have a css on the MVA. if other extensions are int the none partition, it probably doesn't matter what is in the CSS, but you have to have one assigned. Adam Compton On Sat, Mar 26, 2011 at 11:04 PM, CCIE for Me wrote: > Hi All, > > Has anyone else gotten MVA to allow you to call internal extensions if > those extensions are in the partition and the calling search space > you have on the remote destination is blank? As a time saver for > troubleshooting issues I have routinely and successfully been using the > "none" partition for all of my DNs, but it seems that if I have to use MVA, > I cannot make internal calls through MVA, I will always get "your call > cannot be completed as dialed". I have ran this through the ringer, meaning > I have tried different permutations of CSS's on the line and remote > destination and it never works. If I set my whole lab up to use PT-INTERNAL > on all extensions it works fine. > > thoughts? Possible bug? > > thanks for your time > > > ___ > 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
[OSL | CCIE_Voice] Multicast MoH to CUBE
Hi All, When trying to practice some MoH lab today, I notice that whenever HQ/BR1 phone make a call to BR2 Phone via H323 gateway, which BR2 setup as CME (telephony-service), when HQ/BR1 phone press hold, the MoH will not work (BR2 phone hear silence). But when BR2 phone press Hold, HQ/BR1 phone can hear MoH. Same for when HQ/BR1 tried to send call out to PSTN via the H323 gateway, the PSTN hear no MoH. BR2 Phone call PSTN the MoH is ok though. My HQ is set to send Unicast MoH, while BR1 is set to send Multicast MoH. MoH servers have different DP and Region to all site is g711. I did tried to turn the BR2 into a H323 gateway only (no CUBE) and configured the following commands: ccm-manager music-on-hold, moh music-on-hold.au multicast moh 239.1.1.5 port 16384 Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via the H323 gateway, also when HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working fine for both direction. Is this normal that Multicast MoH or Unicast MoH is not supporting CUBE in this case? also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine? Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Home lab PSTN
For the lab, I believe you will not have access to the PSTN so you can just use ethernet for access or even console. You might like to see debug on PSTN but again I do not believe you will be tested on anything at the PSTN during the lab so once configured, you should be able to leave it alone. Running your Frame relay on the 2522 is a good use of old equipment, good thinking and a great way to save money. PVDM modules I use are the PVDM2-8 and I bought several of them. I know it becomes a hassle to move them around but I have learned a lot about how they work because of this. However, in the lab you will have a standardized platform I recently got a router with a couple of PVDM2's in it so if you need them give me a ping and we can go from there. Bill On Sat, Mar 26, 2011 at 5:52 PM, Shaun P wrote: > Hi Voicers: > > I have a home lab with 2811 as PSTN router and a 2522 as frame relay > router. Do I need to add a serial interface to my PSTN router and connect it > to the frame relay router? I see the IPexpert PSTN configuration has ospf in > it. > > What would be the minimum PVDM2 requirement to complete all the labs? > > Thank you in advance and keep labbing > > Shaun P > > ___ > 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] MVA using sip trunk
hi there I had the same problem SIp trunk MVA is support on CUCM 7.13 . If is for the CCIE lab you need to configure it as H323. regards. On 27 March 2011 07:24, Erwan Erwan wrote: > hi all, > > I tried to config MVA in HQ site, > > Scenario > - > I dial in from PSTN to 5999 (MVA #) > > it gave me prompt for pin and press "1" to dial any outside number (911) > > 1. If I use SIP trunk It will gave me error saying (below is the complete > "debug ccsip message" > > * "Sent: > SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' "* > > 2. However if I use, H323 GW , call out by MVA to outside (ie :911 ) work > fine > > Question > - > - wondering any parameter in SIP\ trunk I need to check ? > > tks in adv > > > > *Complete debug * > *===* > *PSTN-WAN# > Mar 27 06:07:26.026: //-1//SIP/Msg/ccsipDisplayMsg: > Sent: > SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' > Via: SIP/2.0/TCP 10.10.210.10:5060;branch=z9hG4bK1861aabffa > From: >;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-20651851 > To: ;tag=1343344-12F6 > Call-ID: > **7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10*<7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10> > *CSeq: 101 INVITE > Reason: Q.850;cause=100 > Content-Length: 0* > ** > ** > ** > ** > > > ___ > 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] MVA using sip trunk
which version of CUCM are you using? because MVA on SIP is supported starting 7.1.3 I think... On Sun, Mar 27, 2011 at 7:24 AM, Erwan Erwan wrote: > hi all, > > I tried to config MVA in HQ site, > > Scenario > - > I dial in from PSTN to 5999 (MVA #) > > it gave me prompt for pin and press "1" to dial any outside number (911) > > 1. If I use SIP trunk It will gave me error saying (below is the complete > "debug ccsip message" > > * "Sent: > SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' "* > > 2. However if I use, H323 GW , call out by MVA to outside (ie :911 ) work > fine > > Question > - > - wondering any parameter in SIP\ trunk I need to check ? > > tks in adv > > > > *Complete debug * > *===* > *PSTN-WAN# > Mar 27 06:07:26.026: //-1//SIP/Msg/ccsipDisplayMsg: > Sent: > SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' > Via: SIP/2.0/TCP 10.10.210.10:5060;branch=z9hG4bK1861aabffa > From: >;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-20651851 > To: ;tag=1343344-12F6 > Call-ID: > **7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10*<7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10> > *CSeq: 101 INVITE > Reason: Q.850;cause=100 > Content-Length: 0* > ** > ** > ** > ** > > > ___ > 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
