Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus on outgoing PRI calls
Many thank Justin. This is very helpful. Best regards Paul Onwude (Please excuse the brevity of this email as it was sent via a mobile device. Kindly excuse misspelled words or sentence structure) On 4 Nov, 2013, at 7:28 pm, Justin Carney justin.s.car...@gmail.com wrote: If you're adding the plus I cucm this is expected behavior that it will be lost at the ios gw. You're not missing anything on cucm, this is just how h323 works on ios routers...the only option is to add a plus in the ios gateway. IOS/h323 will (by design, unfortunately) discard a plus in dnis on the inbound call leg, so make sure you are adding it on the outbound dial or voice port. (You might be able to add plus on an inbound dial peer but I haven't tried and not sure if it would then get discarded.) For the lab I would recommend doing ALL h323 digit manipilation on the h323 gw. If you do it in cucm then later when you need to do srst you will need extra dialpeers and/or translations. For example, I use only 4 outbound pots dialpeers in h323 each with a translation profile that modifies ani (for each site, matching 2... 3... and 4...) and dnis (usually just type/plan). I send all the dialed digits from a phone to the h323 gw the let the gw make the ani and dnis match the pstn requirements for all call types including teho. The only time I modify digits in cucm is when no new rp is alowed for 911 at site b and that uses slrg - the ani is masked to 7digits in cucm, but the h323 still must set ani/dnis type and plan to unknown for dnis. Also, for teho from site a to site b pstn, I would strip the 91408 from the dialed digits and prefix 9 in the route list/rg for h323 so it matches my outbound local dialpeer. Hope this helps. -Justin On Nov 4, 2013 2:19 AM, Paul Onwude ponw...@gmail.com wrote: Hi All. Need expert opinion on this. I have a H323 gateway and i have setup called party transformations on CUCM to send “+” to the gateway. My issue is i don’t see the plus when i debug ids q931. When i do other digit manipulation like adding “#”, it show up on the gateway but not the “+” I know i can probably achieve this using translation on the GW but i can’t help but think there is something i am missing in CUCM. Any ideas?? Paul ___ 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] H323 gateway doesn't send the plus on outgoing PRI calls
If you're adding the plus I cucm this is expected behavior that it will be lost at the ios gw. You're not missing anything on cucm, this is just how h323 works on ios routers...the only option is to add a plus in the ios gateway. IOS/h323 will (by design, unfortunately) discard a plus in dnis on the inbound call leg, so make sure you are adding it on the outbound dial or voice port. (You might be able to add plus on an inbound dial peer but I haven't tried and not sure if it would then get discarded.) For the lab I would recommend doing ALL h323 digit manipilation on the h323 gw. If you do it in cucm then later when you need to do srst you will need extra dialpeers and/or translations. For example, I use only 4 outbound pots dialpeers in h323 each with a translation profile that modifies ani (for each site, matching 2... 3... and 4...) and dnis (usually just type/plan). I send all the dialed digits from a phone to the h323 gw the let the gw make the ani and dnis match the pstn requirements for all call types including teho. The only time I modify digits in cucm is when no new rp is alowed for 911 at site b and that uses slrg - the ani is masked to 7digits in cucm, but the h323 still must set ani/dnis type and plan to unknown for dnis. Also, for teho from site a to site b pstn, I would strip the 91408 from the dialed digits and prefix 9 in the route list/rg for h323 so it matches my outbound local dialpeer. Hope this helps. -Justin On Nov 4, 2013 2:19 AM, Paul Onwude ponw...@gmail.com wrote: *Hi All. * *Need expert opinion on this.* *I have a H323 gateway and i have setup called party transformations on CUCM to send “+” to the gateway. My issue is i don’t see the plus when i debug ids q931. When i do other digit manipulation like adding “#”, it show up on the gateway but not the “+”* *I know i can probably achieve this using translation on the GW but i can’t help but think there is something i am missing in CUCM.* *Any ideas??* *Paul* ___ 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] AAR Configuration
Hi Guys, i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and AAR-CSS on device pool it does not take effect rather i have to apply it each phone device and GW inorder for it to work.Is there any thing i am missing Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] AAR Configuration
The aar-group setting on device pool does NOT get pushed to all devices in the device pool, while the aar-css does. My strategy is to set aar-css at the dev pool and to manually set the aar-group on EVERY device (phone, gw, vm or cti port, etc) and EVERY line/dn. This take no thought while provisioning thing and when I get to an aar question the only thing to build is the rlist (maybe, if an existing doesnt match exactly) and route pattern. That said my strategy is slightly overprovisioned to save time. I did thorough testing and came up with the minimum config for aar: 1. The calling entity must have the AAR-CSS on the DEVICE (there is no aar css field on a dn, it only exists on a device/port/gw) 2. The calling entity must have the AAR-GROUP set on Either Device *OR* Line/DN 3. The called/target LINE/DN must have the AAR-GROUP. (this makes sense, as you call a dn and you don't care which device(s) have a line appearance for this dn.) if the called DN doesn't have the aar-group it will NOT work, regarless of whether the device where the dn is assigned has the aar-group In summary, my strategy pit the group every where and the css on devices and I don't have to memorize the minimum req in the lab - or more importantly I don't revisit config pahes just to setup aar. Hope this helps... -Justin On Nov 4, 2013 6:57 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Guys, i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and AAR-CSS on device pool it does not take effect rather i have to apply it each phone device and GW inorder for it to work.Is there any thing i am missing Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice Lab dates all gone?
Hi, Has anyone else been able to schedule a date? I don't see any dates passed Nov 10 at RTP or San Jose. Regards, Frank On Oct 30, 2013, at 8:40 PM, Patrick Henderson p.hender...@mac.commailto:p.hender...@mac.com wrote: Hi Bill, I read you mail an my heart missed a beat. I just scheduled my lab for Sunday Jan 5th. All the best on the 28th. And good luck with your exam Somphol. Ciao Pat On Oct 30, 2013, at 5:32 PM, Bill Tolentino btolent...@hotmail.commailto:btolent...@hotmail.com wrote: I installed Chrome and viola! I can see the dates now. I think they did do some re-arranging with the dates today, now allowing weekend testing. In the process, IE Firefox browsers got bugged somehow. In any case, I'm scheduled for Jan 28th now happy to get back to studies! Much thanks Somphol good luck on your exam! Take care! Bill Tolentino From: somp...@gmail.commailto:somp...@gmail.com Date: Thu, 31 Oct 2013 11:26:22 +1100 Subject: Re: [OSL | CCIE_Voice] Voice Lab dates all gone? To: btolent...@hotmail.commailto:btolent...@hotmail.com CC: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com It is also very good to see lab slots available on Saturday and Sunday at both RTP San Jose. Weekend lab seem to be for those two locations only at the moment. --Somphol. On Thu, Oct 31, 2013 at 8:43 AM, Somphol Boonjing somp...@gmail.commailto:somp...@gmail.com wrote: On Thu, Oct 31, 2013 at 8:21 AM, Bill Tolentino btolent...@hotmail.commailto:btolent...@hotmail.com wrote: Then at approximately 7:10am they were all gone. I have been re-checking all day and still no dates for any sites.? Hi Bill, I have just checked.There are still available slots in Bangalore. 17 in Nov/ 6 in Dec/ 36 in Jan. None in Sydney / Brussels / Beijing. San Jose, 1 in Nov, 1 in Dec and 1 in Jan. Tokyo - 8 in Nov, 5 in Jan. RTP - 5 in Nov. Because the back-end of the lab is in San Jose, I think Cisco Cert team can shuffle around to reduce slots in one location and make it appear to another, although not on a daily basis, the lab slots seeing today may drastically change. I was in Sydney last week, there are two equipment sets, but only one is being utilized. And based on that all the available slots are now booked in Sydney. I have also written to Cert Support team to give them the feedback. So far the response is that this is a known issue and too bad just try to book at other locations. My next attempt will be in Bangalore, I hope equipment and the network access speed is bearable. Regards, --Somphol. ___ 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.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] how-to: globalize/localize on missed call list vs display
Study group – below is a how-to/strategy response I sent to a friend earlier today who asked about the globalization/localization questions in the practice labs. I'm sharing here in case anyone else needs some pointers and has not yet asked the study list. I hope this helps someone…and if not that's fine, it already helped the person I wrote it for :-) *QUESTION*: CngPTP setup - globalize/localize on missed call list vs display. Should I globalize on the GW according to NPI/TON or do it manually using a CngPTP CSS? *ANSWER*: To solve those questions about modifying the display of a phone it helps me to break it into two steps/phases, at least conceptually. Once you practice it a few times you can configure it in one shot in a specific order, but don't try to do that until you nail the config down and do it a few times. *Phase 1*: IGNORE the phone display, focus on globalization - globalize the inbound call, make it show in the call list, and make it routable when you call outbound from the missed call list. 1. Look at the inbound q931 from the gw to see the PSTN's ANI, make note of the TON. Go to the gateway config in CUCM (either mgcp or h323), and set the prefix based on TON, for example if subscriber with 10 digits then set the sub prefix to +1 or whatever you need. Make this work first (i.e., make it show on the phone), then move on. 2. After you get a missed call with + in the call log (and for now + on the display), make it routable when redialing. Copy/create a RLIST (this is important for the display edits later, DON'T reuse an existing route list), and do your DDI/TON on the RLIST level to make the + call work – only worry about the specific source phone and destination number listed in the question. When this call works (ensuring the PSTN requirements are still met from the routing section), move on to the next phase. *Phase 2*: Now fix the localized display on the phone. 1. INBOUND – you have already modified inbound ANI on the gw (in CUCM) to be the +e164 number. Now create a new CngPTP (in a new PT or a placeholder xform-partition) to match this + number and manipulate to what the question states. The CSS that contains the PT for the CngPTP (either new CSS or a placeholder xform-css which contains form-partition) should now be applied at the target PHONE, remember to UNCHECK 'use dev pool CSS.' You should now have localized the inbound connected number and still have + in call log. 2. OUTBOUND – you should already be able to place the return call from call log, using RLIST for digit manip (part of Phase 1). The reason to do digit manip on RLIST is so that here you can ALSO do digit manip on the Route Pattern. The RP digit manip will be used for display on phone only because RLIST also does digit manip and RLIST will override the RP. (Side note, if you do DDI on RP and NOT on RLIST, then the DDI RP will take effect for the DNIS. In my labs I always do DDI RLIST except special situations like this or when I skip a RLIST and have the RP point directly to an ITSP trunk (and if the display number doesn't matter).) 1. Note – depending on which digits you are removing on the RLIST to make the call route (from Phase 1) the dot may be in the wrong place for your use pre dot strip here in the RP. In this case, move the dot to the right so you can use DDI for phone display to localize, but make sure to go edit the RLIST to prefix those extra digits that the gw needs to keep. 2. Example – you may have originally used +44.2077961234 and predot in RLIST to make the call route. Now for localization you may need to display 77961234 so you need +4420.77961234 in the RP and predot here. Since you moved the dot the DDI in the RLIST is broken, so now modify the RLIST based on the new dot position to prefix the 20. Now that it makes conceptual sense, practice a few times on different sites – make up your own questions/requirements once you get bored with the practice lab question. After a couple times, you'll see the 'big picture' when you read the question and you can go through the whole process and setup all the DDI on the first shot at the RL and RP (rather than doing the RL, then changing it after you realize the RP ddi needs to move the dot for phone display). Good Luck! ___ 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] AAR Configuration
Hi Justin, Thanks for your reply i will try that and wll update you on ths... Thanks On Mon, Nov 4, 2013 at 4:54 PM, Justin Carney justin.s.car...@gmail.comwrote: The aar-group setting on device pool does NOT get pushed to all devices in the device pool, while the aar-css does. My strategy is to set aar-css at the dev pool and to manually set the aar-group on EVERY device (phone, gw, vm or cti port, etc) and EVERY line/dn. This take no thought while provisioning thing and when I get to an aar question the only thing to build is the rlist (maybe, if an existing doesnt match exactly) and route pattern. That said my strategy is slightly overprovisioned to save time. I did thorough testing and came up with the minimum config for aar: 1. The calling entity must have the AAR-CSS on the DEVICE (there is no aar css field on a dn, it only exists on a device/port/gw) 2. The calling entity must have the AAR-GROUP set on Either Device *OR* Line/DN 3. The called/target LINE/DN must have the AAR-GROUP. (this makes sense, as you call a dn and you don't care which device(s) have a line appearance for this dn.) if the called DN doesn't have the aar-group it will NOT work, regarless of whether the device where the dn is assigned has the aar-group In summary, my strategy pit the group every where and the css on devices and I don't have to memorize the minimum req in the lab - or more importantly I don't revisit config pahes just to setup aar. Hope this helps... -Justin On Nov 4, 2013 6:57 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Guys, i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and AAR-CSS on device pool it does not take effect rather i have to apply it each phone device and GW inorder for it to work.Is there any thing i am missing Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
