Re: [OSL | CCIE_Voice] MVA partial match issue
I see, I have make sure my RDP CSS include the 4 digit ext. but still failed. On Sun, Apr 7, 2013 at 2:44 PM, William Bell b...@ucguerrilla.com wrote: Not quite. The RDP CSS is used by the MVA process in CUCM to make the final call routing decision. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 7, 2013, at 3:46 PM, donny f wrote: yes i had specified it under service param, so far i only restart the MVA service in UCM/. I think this no need RDP css, as i only test MVA. When i press 4 ext , debug voip dialpeer show it hits the MVA number 5999. Here is how I understand , pls correct if this is not right. - when press 1 to call 4 digit, dial-peer voip in IOS router will match 5999 to CallManager VMA 5999 (under Media Resources). - after successfully in UCM MVA, it is up to CallManager VMA process to dial 4 digit (and no need CSS here) Tks d On Sun, Apr 7, 2013 at 12:40 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote: It uses the RDP's css while snr uses the re-routing css. Did you also specify the MVA number in the service parameters? A peer had mentioned to me that the service may need to get restarted as well haven't tested it yet though. Regards, Hugo On Apr 6, 2013, at 8:38 PM, donny f f.faraday...@gmail.com wrote: hi Bill and others, I had put the MVA under Media Resources, however when i dial 4 digit ext, it said: the number you dial can't be reached. Questions: - when we use MVA to call 4 digit, are they use IOS dial-peer or RD css to call this 4 digit local ext - my partial match never work , i use 7 digit as match. any idea what missed? tks On Wed, Mar 27, 2013 at 5:46 AM, William Bell b...@ucguerrilla.com b...@ucguerrilla.com wrote: I have ran into a similar problem. In my case I would get a fast busy after entering the extension number followed by #. The issue was I neglected to provision Mobile Voice Access under Media Resources. On Tuesday, March 26, 2013, Barrera, Hugo wrote: Regarding MVA during my first attempt (real lab) I had it working except for when I dialed in and tried to call another 4-digit ext like SAPH1 or SBPH2 any ideas why that didn’t work? ** ** *Regards,*** *Hugo* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Justin Carney *Sent:* Monday, March 25, 2013 1:51 AM *To:* donny f *Cc:* ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com); michael.se...@compucom.com; networksanitytoinsan...@gmail.com *Subject:* Re: [OSL | CCIE_Voice] MVA partial match issue ** ** You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote:*** * Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
hi all, does anyone know the quickest trick on provision phone and DN in exam? i ever read ,there is super copy. tks d On Sat, Apr 6, 2013 at 12:50 AM, Peter Simmons pe...@grayrigg.com wrote: Bill, Your feedback has been awesome - much appreciated by all here, I'm certain - it has definitely refined and clarified my own vague thinking in this area considerably - thankyou! Could you expand on the phone provisioning method you use? I'm taking far too long on this during practice (and obviously when I've been supporting the profitability of the Cisco catering team in Brussels previously...) and it is an area I've been looking to completely transform for my next attempt. Is there a documentation link or example you could point me at on this? I'd like to see what difference it would make to my times by trying this (radically) different approach! regards Peter Peter Simmons On 4/3/2013 5:59 PM, William Bell wrote: I am not familiar with Marko's approach for on-screen window placement. I actually don't have a specific strategy in this area. I do create a notepad file for the following: basic.txt : basic infrastructure notes and notes on phone/user configs sw.txt : switch configs hq.txt : HQ gateway/router configs sb.txt : Site B gateway/router configs sc.txt : Site C gateway/router configs rp.txt : Route plan configs (when I get to that point) I have the above .txt files open all of the time. I only keep basic.txt up on the screen. I keep the others minimized. I restore them as needed. During the course of the exam I will create other notepad files temporarily. Most notably: 1. When I create partitions. I have a naming convention that is basically uniform across sites. So, I lay out the HQ versions in notepad. Paste in CUCM. Then do a search/replace for HQ/SB. Repeat for Site C. Kill the notepad 2. When I provision phones. I use a series of SQL commands from the CLI to provision phones. I type them out in notepad and paste from there. Then I kill the notepad. 3. Troubleshooting questions. Because I don't want to deal with VNC's sluggish nature, I'll do my TS work in notepad on the candidate PC and then copy/paste to the VNC desktop. I think that's it. As far as window orientation. I keep basic.txt in the top right corner of the screen. If I need hq.txt/sb.txt/etc. then I restore to bottom right. I'll keep (or try to keep) console sessions in the middle and IE sessions near the left. But I haven't really thought about it that much. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 3, 2013, at 12:32 PM, Ramcharan Arya wrote: Hi Bill, Thank you very much for nice writeup on strategy. This is really helpful for CCIE vocie lab aspirants. Do you have any strategy how many notepad sessions to keep open simultaneously. How to arrange SecureCRT sessions screen, online lab webpage, and notepad on 32 screen. I am still practice same method which I learn during RS bootcamp with Marko.If you have any better approach please share. Regards, Ramcharan Arya CCIE # 28926 (RS) On Wed, Apr 3, 2013 at 10:57 AM, William Bell b...@ucguerrilla.comwrote: I have had this as a draft for a few days. Just too busy to finish it until now. So, some of my thoughts are redundant to what others have said. Hopefully that isn't a bad thing. Timing is definitely a critical aspect of the exam. I know I have areas where I am slower than I should be. I suspect most people do. Most of my comments herein are based on my self-study practice labs. I have taken the lab a couple of times but most of the tinkering I have done with my method is during self-study. When I sit for the real lab, I don't tinker. I go with whatever method I have been practicing. So, that is suggestion #1: Don't tinker on lab day, stick to your guns and don't 2nd guess your method. Going back to the OP, I believe you should look at the bright side. Your statement ...I seemed to keep moving forward... is key. The fact you were able to avoid a stall is important. I believe controlling this exam is about rhythm and finding what config approach helps you establish a sustainable and consistent rhythm. Speed on any individual task is critical but rhythm is king in my opinion. Like others (most?), I follow the device-based approach. It has been around since pre 3.0 blueprint (contrary to popular opinion) and is a proven strategy. However, I have found that you will need to customize that approach to suit your needs. For me, it is about managing the transitions. Again, I believe focusing on establishing and maintaining a rhythm is absolutely key. Smoothing the transitions and/or stacking tasks that help ease transitions is important. Also, you won't maintain the same rhythm throughout the exam. Some tasks you will bang out (or should) very fast. Others, you will need to pay close attention to what you are doing. So,
Re: [OSL | CCIE_Voice] No MWI Light SC calls SA or B thru GK
Hi Hugo, If I remember correctly, This is an issue with IPexpert setup. Since Both SA and SB users are imported to CUC, MWI will work but SC user is not imported to CUC and hence MWI will not work. It sounds weird but to confirm this, delete SA user from CUC and try to leave a VM for SB user. I might be wrong but I read some old emails about this issue from the OSL's archive. Regards,Mohamed Gazzaz From: hugo.barr...@nexusis.com To: ccie_voice@onlinestudylist.com Date: Mon, 8 Apr 2013 05:03:14 + Subject: [OSL | CCIE_Voice] No MWI Light SC calls SA or B thru GK Hi Guy’s, I have been experiencing a weird issue on my last couple of remote lab sessions…When SA and SB call each other and forward to CUC to leave a msg the MWI lights up just fine. However when SC calls site A or B, thru gatekeeper, and forwards to VM to leave a msg NO MWI light. My session ended and I didn’t really get a chance to troubleshoot it, wondering if anyone has seen this before? Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME - GK - UCM bandwidth
Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards,Mohamed Gazzaz Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
Will check and let you know. BTW you mentioned a known bug can you please post a link of it? Thanks. On Mon, Apr 8, 2013 at 7:41 PM, Mohamed Gazzaz mgaz...@hotmail.com wrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
I don't have a the link to it but I read about it in this OSL. Date: Mon, 8 Apr 2013 19:47:35 +0545 Subject: Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth From: bring...@gmail.com To: mgaz...@hotmail.com CC: ccie_voice@onlinestudylist.com Will check and let you know. BTW you mentioned a known bug can you please post a link of it? Thanks. On Mon, Apr 8, 2013 at 7:41 PM, Mohamed Gazzaz mgaz...@hotmail.com wrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards,Mohamed Gazzaz Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA partial match issue
Hi Donny, Partial/Complete match is NOT related to 4 digit dial. I thought you were asking about separate issue with matching remote destination. Let me ask you this, you're not trying to call the extension on the phone for which you have Remote Destination/Profile configured, are you? Sergey On Mon, Apr 8, 2013 at 1:38 AM, donny f f.faraday...@gmail.com wrote: hi sergey, yes my dial-peer for MVA that match the service parameter is there and it actually trigger it when i press 1 and 4 digit ext. Also CSS for GW and RDP seem include the 4 digit partition. When you said this : -- I know there are some bugs with partial match in early versions of CUCM 7.X, the workaround is to use complete match. Can you explain bit, how this complete/partial match related to failed 4 digit dial ? So far I always think if you can call MVA from PSTN and they prompt to enter PIN, then means partial/complete match is ok here Tks d On Sun, Apr 7, 2013 at 8:59 AM, Sergey Heyphets ser...@heyphets.comwrote: Hi Donny, When you dial 4 digit extension from the MVA, the IOS sends call to the MVA number defined under Media Resources, so you must have a dial-peer that matches that number and sends the call to the CUCM. The extension you've dialed is transfered in the Redirected number IE inside the SETUP message sent to the MVA number defined under media resources. Once the call gets to CUCM, it extracts the extension you've dialed from the Redirected Number IE and uses either Gateway CSS or RDP+Line CSS (depending on Service Parameters) to place the call to extension. So, if your call to extension doesn't work, you need to check that you have dial-peer that matches MVA number defined in Media Resources, the Service Params to see which CSS you use for MVA calls and then make sure that whatever CSS you use can reach that extension. I know there are some bugs with partial match in early versions of CUCM 7.X, the workaround is to use complete match. Sergey ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://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] CME - GK - UCM bandwidth
Here is bug id CSCsl74701 Bug Details ARQ requests 1280 when no regions are defined to use g711 Regards, Ramcharan Arya CCIE # 28926 (RS) On Mon, Apr 8, 2013 at 9:02 AM, Suresh Bhandari bring...@gmail.com wrote: Will check and let you know. BTW you mentioned a known bug can you please post a link of it? Thanks. On Mon, Apr 8, 2013 at 7:41 PM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.com wrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
The workaround is to change the service parameter for default intra-region codec to g729. You will then obviously need to update your regions for site a b c to use g711 rather than 'default' which is now g729. This bug mentioned above is where a gk send a call to cucm and it doesn't look at the region's setting (where you define a gk region and set to g729 for intra region) but instead only looks at the service parameter. On Apr 8, 2013 11:26 AM, Suresh Bhandari bring...@gmail.com wrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
The brq parameter does does apply here (ie, won't fix the issue) since a brq is sent after a call is already connected and is requesting a *change* in bandwidth. The initial call setup is done with an arq that contains the initial bandwidth request. If you debug this issue end to end, you will see site 3 router send arq with bandwidth 16 (shown as 160), and gk will send acf to site 3 with be 16. The issue is from the go to cucm, I don't recall if the go arq is asking cucm for 16 or 128, but I know the acf from cucm to go is 128. No where in this call setup is a brq used, and the call is setup at 128k and uses g711. If you have time on your hands, it would certainly be a good exercise to try both service params and debug each...or just save yourself the trouble and use the intra region SVC param :-) On Apr 8, 2013 12:26 PM, Suresh Bhandari bring...@gmail.com wrote: I confirm that I was hitting the bug! Here are my results: 1. When I changed the SP BRQ Enabled to True I got the bandwidth while ringing state 128K and in connected state 16K. 2. When I changed the Intraregion codec to G729, I got the bandwidth - while ringing and connected - to be 16K. So, it really depends upon the question we face, whether to enable BRQ or set Intraregion codec. So, thank you guys. On Mon, Apr 8, 2013 at 8:46 PM, Suresh Bhandari bring...@gmail.comwrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
...looks like I didn't read you email correctly the first time and you already tried both methods :-) My recommendation (may or may not be the way the lab is graded) is that you should use intra region param. Reason is the grading script might not connect the call or it will only look at debugs for arq. In this case you lose points in the brq method. On the other hand, if a call is setup and the codec is check from the phones, then either method works. I can't think of a reason where you would want to have a diff bw when setting up the call than when connected, and why take the chance of losing points using brq? That said, it wouldn't hurt to use both params, as another question may require you to turn on brq. I would just recommend against *only* enabling brq to answer this question. On Apr 8, 2013 12:56 PM, Justin Carney justin.s.car...@gmail.com wrote: The brq parameter does does apply here (ie, won't fix the issue) since a brq is sent after a call is already connected and is requesting a *change* in bandwidth. The initial call setup is done with an arq that contains the initial bandwidth request. If you debug this issue end to end, you will see site 3 router send arq with bandwidth 16 (shown as 160), and gk will send acf to site 3 with be 16. The issue is from the go to cucm, I don't recall if the go arq is asking cucm for 16 or 128, but I know the acf from cucm to go is 128. No where in this call setup is a brq used, and the call is setup at 128k and uses g711. If you have time on your hands, it would certainly be a good exercise to try both service params and debug each...or just save yourself the trouble and use the intra region SVC param :-) On Apr 8, 2013 12:26 PM, Suresh Bhandari bring...@gmail.com wrote: I confirm that I was hitting the bug! Here are my results: 1. When I changed the SP BRQ Enabled to True I got the bandwidth while ringing state 128K and in connected state 16K. 2. When I changed the Intraregion codec to G729, I got the bandwidth - while ringing and connected - to be 16K. So, it really depends upon the question we face, whether to enable BRQ or set Intraregion codec. So, thank you guys. On Mon, Apr 8, 2013 at 8:46 PM, Suresh Bhandari bring...@gmail.comwrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
Justin, I hope you read my earlier mail. I had my results when I used BRQ and when I used Intraregion codec. Like they say, my (and hopefully others' including you too) answer depends on what is asked. Thanks. On Mon, Apr 8, 2013 at 10:41 PM, Justin Carney justin.s.car...@gmail.comwrote: The brq parameter does does apply here (ie, won't fix the issue) since a brq is sent after a call is already connected and is requesting a *change* in bandwidth. The initial call setup is done with an arq that contains the initial bandwidth request. If you debug this issue end to end, you will see site 3 router send arq with bandwidth 16 (shown as 160), and gk will send acf to site 3 with be 16. The issue is from the go to cucm, I don't recall if the go arq is asking cucm for 16 or 128, but I know the acf from cucm to go is 128. No where in this call setup is a brq used, and the call is setup at 128k and uses g711. If you have time on your hands, it would certainly be a good exercise to try both service params and debug each...or just save yourself the trouble and use the intra region SVC param :-) On Apr 8, 2013 12:26 PM, Suresh Bhandari bring...@gmail.com wrote: I confirm that I was hitting the bug! Here are my results: 1. When I changed the SP BRQ Enabled to True I got the bandwidth while ringing state 128K and in connected state 16K. 2. When I changed the Intraregion codec to G729, I got the bandwidth - while ringing and connected - to be 16K. So, it really depends upon the question we face, whether to enable BRQ or set Intraregion codec. So, thank you guys. On Mon, Apr 8, 2013 at 8:46 PM, Suresh Bhandari bring...@gmail.comwrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
And this last mail I received when I just pressed send. Anyways, thanks for sharing your views. On Mon, Apr 8, 2013 at 10:47 PM, Justin Carney justin.s.car...@gmail.comwrote: ...looks like I didn't read you email correctly the first time and you already tried both methods :-) My recommendation (may or may not be the way the lab is graded) is that you should use intra region param. Reason is the grading script might not connect the call or it will only look at debugs for arq. In this case you lose points in the brq method. On the other hand, if a call is setup and the codec is check from the phones, then either method works. I can't think of a reason where you would want to have a diff bw when setting up the call than when connected, and why take the chance of losing points using brq? That said, it wouldn't hurt to use both params, as another question may require you to turn on brq. I would just recommend against *only* enabling brq to answer this question. On Apr 8, 2013 12:56 PM, Justin Carney justin.s.car...@gmail.com wrote: The brq parameter does does apply here (ie, won't fix the issue) since a brq is sent after a call is already connected and is requesting a *change* in bandwidth. The initial call setup is done with an arq that contains the initial bandwidth request. If you debug this issue end to end, you will see site 3 router send arq with bandwidth 16 (shown as 160), and gk will send acf to site 3 with be 16. The issue is from the go to cucm, I don't recall if the go arq is asking cucm for 16 or 128, but I know the acf from cucm to go is 128. No where in this call setup is a brq used, and the call is setup at 128k and uses g711. If you have time on your hands, it would certainly be a good exercise to try both service params and debug each...or just save yourself the trouble and use the intra region SVC param :-) On Apr 8, 2013 12:26 PM, Suresh Bhandari bring...@gmail.com wrote: I confirm that I was hitting the bug! Here are my results: 1. When I changed the SP BRQ Enabled to True I got the bandwidth while ringing state 128K and in connected state 16K. 2. When I changed the Intraregion codec to G729, I got the bandwidth - while ringing and connected - to be 16K. So, it really depends upon the question we face, whether to enable BRQ or set Intraregion codec. So, thank you guys. On Mon, Apr 8, 2013 at 8:46 PM, Suresh Bhandari bring...@gmail.comwrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
Re: [OSL | CCIE_Voice] No MWI Light SC calls SA or B thru GK
Bill,
Good to know I was concerned it might have been GK or something else. I will
try this next time I lab.
Regards,
Hugo
From: William Bell [mailto:b...@ucguerrilla.com]
Sent: Sunday, April 07, 2013 11:59 PM
To: Barrera, Hugo
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] No MWI Light SC calls SA or B thru GK
I believe you may be running into a known issue with the proctor lab setup.
Something about the SMTP domain being changed in CUC.
From my notes:
- connect cli on CUC
- run the following to identify old and new domains
run cuc dbquery unitydirdb select * from tbl_alias
- Run this procedure to fix
run cuc dbquery unitydirdb EXECUTE PROCEDURE
CDP_SmtpAddressMigrate('new.comhttp://new.com/','old.comhttp://old.com/')
-Bill
--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla
On Apr 8, 2013, at 1:03 AM, Barrera, Hugo wrote:
Hi Guy's,
I have been experiencing a weird issue on my last couple of remote lab
sessions...When SA and SB call each other and forward to CUC to leave a msg the
MWI lights up just fine. However when SC calls site A or B, thru gatekeeper,
and forwards to VM to leave a msg NO MWI light.
My session ended and I didn't really get a chance to troubleshoot it, wondering
if anyone has seen this before?
Hugo
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.comhttp://www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlacement.com
[OSL | CCIE_Voice] incoming call from PSTN
Hi All,when they asked to make sure VM pilot (+1408200) can be called directly from PSTN.do i have to make a call from PSTN and dial +1408200? , but pstn phone not support to dial '+'.is that mean dial 1408200 from pstn phone or some tricks behind here? thanks advance-ikizoo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] incoming call from PSTN
When you have the strip digits at gateway level set to 4, and as long as 2220 is a DID then from wherever it is called, it will go to the VM. Regarding plus dialing of VM DID from PSTN, it will be/is registered to CME (h.323), so I don't expect it will send a plus. My two cents. On Tue, Apr 9, 2013 at 7:51 AM, ikizoo hello ikiz...@hotmail.com wrote: Hi All, when they asked to make sure VM pilot (+1408200) can be called directly from PSTN. do i have to make a call from PSTN and dial +1408200? , but pstn phone not support to dial '+'. is that mean dial 1408200 from pstn phone or some tricks behind here? thanks advance -ikizoo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
