Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP
FYI There are a BUNCH of open lab dates in RTP and SJC. No idea why but, there are open labs all of Oct through February. :) On Fri, Sep 27, 2013 at 11:45 AM, Martin Sloan martinsloa...@gmail.com wrote: Hey Alex, I hear ya. I went through all locations and checked availability and Tokyo is the closest for me, which is about an 18 hour flight. I've priced it all out. I'm on the fence a bit about traveling there but at this point I'm leaning toward not. On top of the additional expense for travel and time away from family, I'm paid by the hour so it would be at least 4 days unpaid for me. It starts to add up. Like everyone else I've invested a lot of money into this and I'm starting to get a little gun shy on putting up another couple thousand dollars for something that's not guaranteed. It could be money straight down the toilet. At a certain point, enough is enough. Good luck on your lab in RTP in Feb! Marty On Fri, Sep 27, 2013 at 12:17 PM, Alex Mendoza aa.mend...@icloud.comwrote: As Dave says, you can book at Tokyo or other location. I'm from Mexico and can book at RTP in february just one week ago. More pressure because will be my 2nd and last attempt. If you are so close to get your CCIE, look for a seat at other location even if you must pay for travel expenses. All my best for the last candidates. best regards Alex On Sep 27, 2013, at 10:57 AM, Martin Sloan martinsloa...@gmail.com wrote: I'm really disappointed as well. I just failed my second attempt on Wed and was worried about getting a 4th try in when I logged on to see no seats left for a 3rd! I figured it would get tight but this is nuts. I made a big improvement on my score from the first try and feel like the third time could have been the charm. Oh well. On Fri, Sep 27, 2013 at 11:35 AM, Dane Warner dwar...@epochuniversal.comwrote: There are no open dates in either San Jose or RTP anymore, period. ** ** Looks like if we want to take the Voice exam, which I’m sure Cisco doesn’t want us to do anymore, then it’s either Tokyo or we’re SOL. ** ** Very disappointing. ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *image001.png* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *OSL StudyList *Sent:* Friday, September 27, 2013 3:19 AM *To:* Josh Petro *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP ** ** Do you know what times the lab dates are released for those who have not paid? I thought it was at midnight SJC time but, I am not sure. ** ** On Fri, Sep 27, 2013 at 5:17 AM, Josh Petro josh.pe...@gmail.com wrote: If you mean Voice availability, then you are correct in that RTP is filled. San Jose had a few open spots in Jan Feb last week. I don't believe Collaboration dates are open yet for scheduling. Josh On Sep 27, 2013 5:58 AM, OSL StudyList collaboration.c...@gmail.com wrote: Is anyone having any luck scheduling exams at RTP or SJC? When I try to find an available date, I am seeing NOTHING available. ___ 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 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 ___ 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] unity connection - transfer option for users.
hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to truenbsp; and have update the users caller input option 9nbsp; transfer to extn to 9515111. nbsp;However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar ___ 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] unity connection - transfer option for users.
I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar Ganesha offers Company email website (FREE) at your own domain (FREE) - KNOW MORE ___ 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] unity connection - transfer option for users.
hi Martin, I have done that ( preserve original calling number ) in unitynbsp;nbsp; however it does not preserve the correct calling number for pstn callers .nbsp; Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan lt;martinsloa...@gmail.comgt; Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh lt;rameshdol...@rediffmail.comgt; Cc: ccie_voice@onlinestudylist.com lt;ccie_voice@onlinestudylist.comgt; Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number.nbsp; On Oct 2, 2013, at 9:02 AM, ramesh lt;rameshdol...@rediffmail.comgt; wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to truenbsp; and have update the users caller input option 9nbsp; transfer to extn to 9515111. nbsp;However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar Ganesha offers Company email amp; website (FREE) at your own domain (FREE) - KNOW MORE gt; ___ 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] unity connection - transfer option for users.
Hi Ramesh, Just to make sure we're on the same page, are you setting the CUCM service parameter: Display Original Calling Number on Transfer from Cisco Unity = True If you are setting this, can you explain in a little more detail the call flow and outcome? Thanks, Marty On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.com wrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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] unity connection - transfer option for users.
What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.com wrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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] unity connection - transfer option for users.
The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.comwrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
I wasn't sure RDNIS would matter here but figured I would throw it out there anyway (as it applies when redirecting TO CUC). It seems the unity service parameter mentioned earlier obviates the need to use RDNIS. With the option you proposed on creating a new RP/RL just for this requirement I would just set the digit manipulation/TON on the RL to whatever you see inbound from that specific PSTN ANI to HQ - unless the question told you what the expected outbound ANI/TON should be. Another option would be to compare the original PSTN number with the destination PSTN and set to local if same NPA, LD if different NPA, or international different country codes. If it comes in unknown/unknown then send it back out that way. On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote: The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.comwrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
Yeah, I wasn't sure on that one either and had to test it out. I can't recall what the exact requirement, if any, for calling party TON was on the 'practice test' that I had with a similar task but I'm thinking the only way to properly set the calling TON would be with Xforms on the port level since it could be any number on the PSTN phone, even the number you're trying to dial out to from VM. They'd have to be very specific Xforms though since it could potentially override the current dial-plan manipulations in RL and RP if general masks are used like 10 X's. On Wed, Oct 2, 2013 at 1:21 PM, Justin Carney justin.s.car...@gmail.comwrote: I wasn't sure RDNIS would matter here but figured I would throw it out there anyway (as it applies when redirecting TO CUC). It seems the unity service parameter mentioned earlier obviates the need to use RDNIS. With the option you proposed on creating a new RP/RL just for this requirement I would just set the digit manipulation/TON on the RL to whatever you see inbound from that specific PSTN ANI to HQ - unless the question told you what the expected outbound ANI/TON should be. Another option would be to compare the original PSTN number with the destination PSTN and set to local if same NPA, LD if different NPA, or international different country codes. If it comes in unknown/unknown then send it back out that way. On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote: The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.com wrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the
Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
I have a huge delay in my presence updates on my system. Im assuming that's from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote: Hi You could try a reboot of the CUPS server. Worked for me a couple of times... Cheers, Ovidiu On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not show On the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ___ 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 ___ 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] Presence - on hook and off hook status
Hi Josh, I do not believe it is related to vmware environment. I am assuming your CUPS is integrated with CUCM using SIP trunk. Can you enable SIP debug level to detail and run collect SIP debug logs ( on primary call processing engine i.e. Sub) and check SIP logs why there is delay in status.? Im my home lab I never had this issue it works almost instantly my CUPS client is installed on UCCX server. Regards, Ramcharan Arya CCIE # 28926 ( Voice/RS) On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.com wrote: I have a huge delay in my presence updates on my system. Im assuming that's from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote: Hi You could try a reboot of the CUPS server. Worked for me a couple of times... Cheers, Ovidiu On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not show On the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ___ 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 ___ 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] unity connection - transfer option for users.
Hi Marty, In order to preserve Original calling party TON you have to consider existing route pattern should not override so two possible ways to achieve this you can try to use a translation patter which evaluation prior to route pattern. Let us assume you prefix some additional character and create clng party x-formation pattern and DDI -predot what was prefix in TP and set appropriate plan and type. Use separate pt/css for clng party x-formation pattern. Another option is using application dial-rule can also use for this. Regards, Ramcharan Arya CCIE # 28926 (Voice/Routing Switching) On Wed, Oct 2, 2013 at 12:53 PM, Martin Sloan martinsloa...@gmail.comwrote: Yeah, I wasn't sure on that one either and had to test it out. I can't recall what the exact requirement, if any, for calling party TON was on the 'practice test' that I had with a similar task but I'm thinking the only way to properly set the calling TON would be with Xforms on the port level since it could be any number on the PSTN phone, even the number you're trying to dial out to from VM. They'd have to be very specific Xforms though since it could potentially override the current dial-plan manipulations in RL and RP if general masks are used like 10 X's. On Wed, Oct 2, 2013 at 1:21 PM, Justin Carney justin.s.car...@gmail.comwrote: I wasn't sure RDNIS would matter here but figured I would throw it out there anyway (as it applies when redirecting TO CUC). It seems the unity service parameter mentioned earlier obviates the need to use RDNIS. With the option you proposed on creating a new RP/RL just for this requirement I would just set the digit manipulation/TON on the RL to whatever you see inbound from that specific PSTN ANI to HQ - unless the question told you what the expected outbound ANI/TON should be. Another option would be to compare the original PSTN number with the destination PSTN and set to local if same NPA, LD if different NPA, or international different country codes. If it comes in unknown/unknown then send it back out that way. On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote: The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.com wrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and
Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
Thanks for the reply, Ramcharan. I just blasted my lab to start lab 3 of the OWLE, so I'll need the rest of this week to get it back to where I can test Presence status again. I'll let you know what I find out. Thanks again. Josh On Wed, Oct 2, 2013 at 8:59 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote: Hi Josh, I do not believe it is related to vmware environment. I am assuming your CUPS is integrated with CUCM using SIP trunk. Can you enable SIP debug level to detail and run collect SIP debug logs ( on primary call processing engine i.e. Sub) and check SIP logs why there is delay in status.? Im my home lab I never had this issue it works almost instantly my CUPS client is installed on UCCX server. Regards, Ramcharan Arya CCIE # 28926 ( Voice/RS) On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.com wrote: I have a huge delay in my presence updates on my system. Im assuming that's from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote: Hi You could try a reboot of the CUPS server. Worked for me a couple of times... Cheers, Ovidiu On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not show On the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ___ 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 ___ 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
