Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Sorry for revisiting this old thread. The Calling Party Transformation at the Device Pool level would come in handy for this particular need. In the document starting 7.1.2, this is stated explicitly, http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscallpn.html#wp1325305 . *Cisco Unity/Cisco Unity Connection* Because no calling party transformation options exist in Cisco Unified Communications Manager Administration for voice-messaging ports, make sure that you configure the calling party number transformations in the device pool that is associated with the voice-messaging ports. ... Table 7-8 Configuring the Calling Party Transformation CSS to Localize the Calling Party Number Also mentioned Use Device Pool Calling Party Transformation CSS as a method to Localize the Calling Party Number. ... ... The same document for 7.0.1 contained the table 7-8, but somehow doesn't have that explicit section on Cisco Unity/Cisco Unity Connection's calling party localization. (REF: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1276877) So, I am not so sure whether this is possible in CUCM 7.0.1/CUC 7.0.1. I don't have lab access to test this now, but would appreciate if anyone can help testing this. Note: I recall seeing some sort of Technotes outlining the strategy to perform Calling Party transformation for Call Manager 4.x or something that doesn't rely on Gateway's Calling Party Transformation.I can't locate it now, but if anyone could point me to the URL that would be great. Regards, --Somphol. On Sat, Mar 23, 2013 at 12:34 AM, Leslie Meade leslie.me...@lvs1.comwrote: Easy way of doing this is to copy the hunt pilot and give it another number.. set user caller ID and mask it to Then in the call-manager-fallback change the voicemail to the new hunt pilot and your done ** ** ** ** *Leslie Meade* .. * Mobile:778.228.4339* | *Main:* *604.676.5239* *Email:* leslie.me...@lvs1.com ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chadi H Hassoune (chassoun) *Sent:* Thursday, March 21, 2013 7:10 PM *To:* Pixar Perfect; Mark Thrash (marthras); Steve Keller *Cc:* CCIE Voice OSL *Subject:* Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension ** ** Calling Party Xform and assign it to the CUC Device Pool works fine for me. ** ** HTH ** ** *From: *Pixar Perfect pixarperf...@live.com *Date: *Wednesday, March 20, 2013 11:43 PM *To: *Mark Thrash (marthras) marth...@cisco.com, Steve Keller skeller...@gmail.com *Cc: *CCIE Voice OSL ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension ** ** the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. ** ** here is another way of doing it ... ** ** Voicemail Pilot for CUC is 2200 ** ** call-manager-fallback voicemail 2777 --- siteB specific ** ** translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. ** ** ** ** ** ** there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) ** ** ** ** ** ** ** ** -- From: marth...@cisco.com To: skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Easy way of doing this is to copy the hunt pilot and give it another number.. set user caller ID and mask it to Then in the call-manager-fallback change the voicemail to the new hunt pilot and your done Leslie Meade .. Mobile:778.228.4339 | Main: 604.676.5239 Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chadi H Hassoune (chassoun) Sent: Thursday, March 21, 2013 7:10 PM To: Pixar Perfect; Mark Thrash (marthras); Steve Keller Cc: CCIE Voice OSL Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension Calling Party Xform and assign it to the CUC Device Pool works fine for me. HTH From: Pixar Perfect pixarperf...@live.commailto:pixarperf...@live.com Date: Wednesday, March 20, 2013 11:43 PM To: Mark Thrash (marthras) marth...@cisco.commailto:marth...@cisco.com, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com Cc: CCIE Voice OSL ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. here is another way of doing it ... Voicemail Pilot for CUC is 2200 call-manager-fallback voicemail 2777 --- siteB specific translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) From: marth...@cisco.commailto:marth...@cisco.com To: skeller...@gmail.commailto:skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.commailto:b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Thanks to all for all of your input on this question. Based on my testing of the suggestions provided we are essentially always stripping down the ANI to 4 digits before routing to CUC, that is really our only option since I have stated that Alternate Extension is not allowed in my question. With this technique , if the ANI is read back to the message recipient it will be chopped down to the last 4 digits. I suppose this is okay as long as there is no requirement to play Senders ANI before the message ( which is off by default ) . I dont see a way around this drawback. But the on the hunt pilot calling party transform mask seems to do the job. If I were to see a question like this on the lab exam, i hope the grading gods will be on my side. I do not think you can have it both ways, either CUC gets 4 digit ANI , finds the mailbox and prompts to sign in - and the ANI read back to the message recipient is 4 digits, or CUC gets 10 digits and you must use the Alternate Extension field to find the mailbox. On Thu, Mar 21, 2013 at 12:43 AM, Pixar Perfect pixarperf...@live.comwrote: the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. here is another way of doing it ... Voicemail Pilot for CUC is 2200 call-manager-fallback voicemail 2777 --- siteB specific translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) -- From: marth...@cisco.com To: skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.comwrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Calling Party Xform and assign it to the CUC Device Pool works fine for me. HTH From: Pixar Perfect pixarperf...@live.commailto:pixarperf...@live.com Date: Wednesday, March 20, 2013 11:43 PM To: Mark Thrash (marthras) marth...@cisco.commailto:marth...@cisco.com, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com Cc: CCIE Voice OSL ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. here is another way of doing it ... Voicemail Pilot for CUC is 2200 call-manager-fallback voicemail 2777 --- siteB specific translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) From: marth...@cisco.commailto:marth...@cisco.com To: skeller...@gmail.commailto:skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.commailto:b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Alternatively, you could also create a translation rule in a partition accessible only by the inbound gateway that translates the calling number to 4 digits before sending it to voicemail. The hunt pilot calling transform mask will work, but you could have issues if you have any caller input requirements to route back out to the PSTN from UCON. Derek Wyss CCIE#38238(Voice) On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
Thanks Derek On the surface it seems like that would chop down my ANI to 4 digits for any call into site A not just calls to vm. Also in my case site A is MGCP controlled so I that is not an option for me... On Mar 20, 2013 5:16 PM, Derek Wyss wys...@gmail.com wrote: Alternatively, you could also create a translation rule in a partition accessible only by the inbound gateway that translates the calling number to 4 digits before sending it to voicemail. The hunt pilot calling transform mask will work, but you could have issues if you have any caller input requirements to route back out to the PSTN from UCON. Derek Wyss CCIE#38238(Voice) On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller skeller...@gmail.comwrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
Translation rule on CUCM. MGCP or H323 should not matter. You would match on the called number which is the voicemail pilot number then manipulate the calling number and send it on its way. It would not affect standard calls into Site A as it would not match the rule. Derek On Wed, Mar 20, 2013 at 4:51 PM, Steve Keller skeller...@gmail.com wrote: Thanks Derek On the surface it seems like that would chop down my ANI to 4 digits for any call into site A not just calls to vm. Also in my case site A is MGCP controlled so I that is not an option for me... On Mar 20, 2013 5:16 PM, Derek Wyss wys...@gmail.com wrote: Alternatively, you could also create a translation rule in a partition accessible only by the inbound gateway that translates the calling number to 4 digits before sending it to voicemail. The hunt pilot calling transform mask will work, but you could have issues if you have any caller input requirements to route back out to the PSTN from UCON. Derek Wyss CCIE#38238(Voice) On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller skeller...@gmail.comwrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
As far as i remember the requirement under call routing doesn't specify the called party type National for long distance calls out SiteB gateway AND the question clearly said that YOU ARE ALLOWED TO SEND 4 DIGITS in Calling Number field to the Telco for these calls ... so why not use this dialpeer ..it still meets the requirement ... let me know if anyone has seen any different . dial-peer voice 2600 potsdescription voicemail-pilotdestination-pattern 2600no digit-stripport 0/0/0:23prefix 1408202 Mar 20 22:55:57.807: ISDN Se0/1/0:23 Q931: TX - SETUP pd = 8 callref = 0x0084 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Progress Ind i = 0x8183 - Origination address is non-ISDN Display i = '+19723033002' Calling Party Number i = 0x0080, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '14082022600' Plan:Unknown, Type:Unknown Date: Wed, 20 Mar 2013 17:01:48 -0500 From: wys...@gmail.com To: skeller...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension Translation rule on CUCM. MGCP or H323 should not matter. You would match on the called number which is the voicemail pilot number then manipulate the calling number and send it on its way. It would not affect standard calls into Site A as it would not match the rule. Derek On Wed, Mar 20, 2013 at 4:51 PM, Steve Keller skeller...@gmail.com wrote: Thanks Derek On the surface it seems like that would chop down my ANI to 4 digits for any call into site A not just calls to vm. Also in my case site A is MGCP controlled so I that is not an option for me... On Mar 20, 2013 5:16 PM, Derek Wyss wys...@gmail.com wrote: Alternatively, you could also create a translation rule in a partition accessible only by the inbound gateway that translates the calling number to 4 digits before sending it to voicemail. The hunt pilot calling transform mask will work, but you could have issues if you have any caller input requirements to route back out to the PSTN from UCON. Derek Wyss CCIE#38238(Voice) On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 potsdescription voicemail-pilotdestination-pattern 2600 no digit-stripport 0/0/0:23prefix 1408202If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002.In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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] SRST to voicemail without Alternate Extension
What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.commailto:b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002tel:9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002tel:9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002tel:9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ 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
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. here is another way of doing it ... Voicemail Pilot for CUC is 2200 call-manager-fallbackvoicemail 2777 --- siteB specific translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) From: marth...@cisco.com To: skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit
